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r19137 | raarts | 2009-11-19 12:03:32 +0100 (Thu, 19 Nov 2009) | 3 lines

Replace operatorext() with getoperatorext() that also works when the
call falls back from a queue.

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r19091 | nico | 2009-11-13 12:31:11 +0100 (Fri, 13 Nov 2009) | 2 lines

Check  lockfile function outcome and exit if empty!

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r19088 | nico | 2009-11-13 12:08:55 +0100 (Fri, 13 Nov 2009) | 3 lines

Create php function for lockfile. info: lockfile(_filename_, _seconds_)
checks if a lockfile hasn't been created since _seconds_ ago and then creates a lockfile

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r19046 | raarts | 2009-11-10 17:14:29 +0100 (Tue, 10 Nov 2009) | 2 lines

Allow very large recording to be emailed (as opposed to running out of memory)..

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r19033 | raarts | 2009-11-10 12:09:18 +0100 (Tue, 10 Nov 2009) | 3 lines

When nobody is logged in, the phone number is all zeroes.
The length should be the current extension length though. Fixed.

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r19023 | raarts | 2009-11-09 22:01:17 +0100 (Mon, 09 Nov 2009) | 2 lines

Give the outbound dialplan preference above the national dialplan.

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r19004 | raarts | 2009-11-06 12:38:45 +0100 (Fri, 06 Nov 2009) | 4 lines

Do not send qualify messages to software faxes. This lessens
the amount of IAX messages sent, especially for large amounts
of software faxes.

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r18971 | nico | 2009-11-04 15:02:49 +0100 (Wed, 04 Nov 2009) | 5 lines

Do not supply the company name in the SIP From header any more. 
This is used by some SIP providers for the callerid number, even though
this goes against the SIP standard. So this commit fixes Xeloq not
showing CID...

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r18970 | raarts | 2009-11-04 13:01:54 +0100 (Wed, 04 Nov 2009) | 3 lines

When you dial someone in your own area AND prefix it with your areacode
it should be marked as a local call, not a national one.

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r18930 | raarts | 2009-11-02 14:27:09 +0100 (Mon, 02 Nov 2009) | 2 lines

Add support for Polycom 321/331/560 to the provisioning files.

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r18929 | raarts | 2009-11-02 13:55:41 +0100 (Mon, 02 Nov 2009) | 2 lines

Prevent 'syntax error, unexpected TOK_GT' in the dialplan when companyid is undefined.

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r18923 | raarts | 2009-10-30 17:16:30 +0100 (Fri, 30 Oct 2009) | 4 lines

When a fastdial number expanded to a local number that starts with
31, the convert-number routine thought it was an international number,
and removed the 31. Fixed.

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r18919 | raarts | 2009-10-30 15:15:10 +0100 (Fri, 30 Oct 2009) | 2 lines

Fix Polycom and SNOM provisioning through NAT not working.

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r18914 | raarts | 2009-10-30 14:24:48 +0100 (Fri, 30 Oct 2009) | 2 lines

Fix operator display on blind transfer and subsequent fallback between super companies.

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r18912 | nico | 2009-10-29 17:10:27 +0100 (Thu, 29 Oct 2009) | 2 lines

Fix cross-company fallback in the dialplan..

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r18907 | raarts | 2009-10-29 15:18:44 +0100 (Thu, 29 Oct 2009) | 2 lines

Removed superfluous 'break' statement.

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r18906 | raarts | 2009-10-29 15:18:06 +0100 (Thu, 29 Oct 2009) | 3 lines

If the short-circuit settings changes, recreate configuration for all companies.
Only applies to hosted environments with multiple SIP trunks to SIP providers.

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r18886 | raarts | 2009-10-28 12:42:47 +0100 (Wed, 28 Oct 2009) | 2 lines

Fix cross-company *36 where the companies have different extension lengths.

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r18880 | raarts | 2009-10-28 11:50:39 +0100 (Wed, 28 Oct 2009) | 2 lines

Improved sending the costclass to astiumd.

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r18878 | nico | 2009-10-28 10:50:52 +0100 (Wed, 28 Oct 2009) | 4 lines

Calling cross company: company is seperated with : (which is a sb seperator) so the name is omiited in CALLERID

Changed to (company) name !

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r18864 | nico | 2009-10-26 14:38:06 +0100 (Mon, 26 Oct 2009) | 2 lines

Update on extlen for Calvi reports.

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r18863 | nico | 2009-10-26 14:05:25 +0100 (Mon, 26 Oct 2009) | 2 lines

Add Calvi README.

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r18816 | nico | 2009-10-21 16:09:08 +0200 (Wed, 21 Oct 2009) | 2 lines

updates on Calvi reports

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r18801 | raarts | 2009-10-21 11:37:16 +0200 (Wed, 21 Oct 2009) | 3 lines

Fixed astcfgd stop when it couldn't find the contact id in an astiumcdr record.
This is not fatal error, so just log a warning and continue;

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r18774 | raarts | 2009-10-19 11:20:46 +0200 (Mon, 19 Oct 2009) | 2 lines

Send the RDNIS as RDNIS and not as Dial in the USEREVENT, due to change in astiumd.

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r18772 | raarts | 2009-10-17 21:29:39 +0200 (Sat, 17 Oct 2009) | 3 lines

Fixes sending mail for a recorded call that is incoming on a trunk,
and forwarded to an external number. 

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r18759 | raarts | 2009-10-16 14:59:41 +0200 (Fri, 16 Oct 2009) | 4 lines

Which music on hold plays used to differ depending on who's calling
and who would press the Hold button. This is fixed now, everybody should
hear the official company music on hold style.

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r18700 | raarts | 2009-10-13 13:50:57 +0200 (Tue, 13 Oct 2009) | 2 lines

Added missing quotes around an expression.

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r18695 | raarts | 2009-10-13 11:50:55 +0200 (Tue, 13 Oct 2009) | 2 lines

Re-add macro-set-company-soundpath for use by plugins.

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r18685 | raarts | 2009-10-12 20:23:47 +0200 (Mon, 12 Oct 2009) | 2 lines

Also send CMC to astiumd on outgoing calls.

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r18648 | raarts | 2009-10-09 14:12:42 +0200 (Fri, 09 Oct 2009) | 2 lines

Real fix for unowned incoming calls that are transferred out again.

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r18642 | raarts | 2009-10-08 22:30:30 +0200 (Thu, 08 Oct 2009) | 6 lines

[129194]: [SpeakUp #40361] Astium 2.0: Doorschakelingen krijgen callerid van andere company.
Zie achteraan dit ticket. 
'Unowned' doorschakelingen zoals bv op een queue extensie werken niet 
meer omdat de variable outplan niet gezet is. (omdat take-call-responsability niet 
meer aangeroepen wordt). Fixed.

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r18618 | raarts | 2009-10-07 16:45:45 +0200 (Wed, 07 Oct 2009) | 3 lines

Add *216/*226/*236/*246 to set forwarding to your mobile phone.
Still needs a good soundprompt because currently we only have the robot speaking.

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r18609 | raarts | 2009-10-07 12:29:54 +0200 (Wed, 07 Oct 2009) | 3 lines

When the switchboard retrieved a call from the parking queue, by pressing F1 for example,
the call in the parking queue would be hung up, if the operator phone was offhook. Fixed.

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r18606 | raarts | 2009-10-07 11:01:28 +0200 (Wed, 07 Oct 2009) | 2 lines

Fix small syntax error in dialplan.

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r18598 | raarts | 2009-10-06 15:01:20 +0200 (Tue, 06 Oct 2009) | 2 lines

Remove superfluous line which gave a warning in the asterisk log.

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r18588 | ivar | 2009-10-05 09:36:40 +0200 (Mon, 05 Oct 2009) | 1 line

increase timeout for xml push to aastra phones
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r18583 | raarts | 2009-10-04 16:01:37 +0200 (Sun, 04 Oct 2009) | 3 lines

Try to find out the customerid on outgoing calls. Neede for our paid
support plugin, and what's behind all that.

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r18580 | raarts | 2009-10-04 00:58:02 +0200 (Sun, 04 Oct 2009) | 2 lines

Create hook for plugins to hook into astcfgd.

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r18574 | raarts | 2009-10-03 20:28:45 +0200 (Sat, 03 Oct 2009) | 2 lines

Add syslog macro. For plugins to use.

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r18560 | raarts | 2009-10-02 14:07:54 +0200 (Fri, 02 Oct 2009) | 2 lines

Honour short-circuit advanced setting in the dialplan.

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r18559 | raarts | 2009-10-02 14:03:45 +0200 (Fri, 02 Oct 2009) | 2 lines

Oops, forgot a '-'.

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r18550 | ivar | 2009-10-02 10:22:07 +0200 (Fri, 02 Oct 2009) | 4 lines

Bugfix: in case that both source and answer party are mailed the recorded call,
create new mail object for second mail.
Otherwise the source party is included as destination for the mail to the answer party, as well.
Also the recording is attached twice.
------------------------------------------------------------------------
r18540 | raarts | 2009-10-01 12:20:52 +0200 (Thu, 01 Oct 2009) | 5 lines

When the root user logs in, he will now get a performance edge on the rest
of the system, so when it is overloaded, he'll be able to fix things faster.
Normal users on the other hand swing the other way, because they should
not be able to steal the CPU or Disk away from telephony processes.

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r18539 | raarts | 2009-10-01 12:18:32 +0200 (Thu, 01 Oct 2009) | 1 line

fix setting from-extension when forwarding calls
------------------------------------------------------------------------
r18538 | raarts | 2009-10-01 12:16:15 +0200 (Thu, 01 Oct 2009) | 4 lines

Fix for super companies: when you assign an extension from another company,
ensure that the backend understand the it should search for that extension
in that other companies dialplan. Just making Joop happy ;-)

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r18494 | raarts | 2009-09-24 16:13:43 +0200 (Thu, 24 Sep 2009) | 2 lines

Fix taking call ownership. In some cases the callerid was wrong on incoming and forwarded calls. Fixed.

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r18487 | raarts | 2009-09-23 20:06:34 +0200 (Wed, 23 Sep 2009) | 2 lines

Problem caused by rev 17745: phones could not dial out on a trunk in another country. Fixed.

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r18483 | raarts | 2009-09-22 18:42:38 +0200 (Tue, 22 Sep 2009) | 2 lines

Oops, fixed typo.

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r18482 | raarts | 2009-09-22 18:34:13 +0200 (Tue, 22 Sep 2009) | 2 lines

Commit the proper directories for sound prompts

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r18477 | ivar | 2009-09-22 10:59:34 +0200 (Tue, 22 Sep 2009) | 2 lines

(NeoNova 127941) If astcfgd can't get a lock because a lockfile exists,
and that lockfile is older than 2 minutes, remove it because it's probably a leftover from a crash.
------------------------------------------------------------------------
r18475 | raarts | 2009-09-22 08:42:07 +0200 (Tue, 22 Sep 2009) | 9 lines

Randomize the qualify values a bit, in an attempt to prevent 'OPTIONS storms'.
Note: asterisk 'pings' a phone normally every 60 seconds, but unreachable or
lagged phones are tried eery 10 seconds. Changing this requires recompiling.
Not that I see a reason the change this by the way...
This tries to address an issue where - if hundreds of phones are used - 
it seems to be more common for phones to become lagged or unreachable.
It still is not proven that asterisk causes this in any way, but on the
other hand, this patch doesn't do any harm.

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r18463 | raarts | 2009-09-18 16:14:45 +0200 (Fri, 18 Sep 2009) | 2 lines

Add Language support for Grandstream phones.

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r18462 | raarts | 2009-09-18 15:04:05 +0200 (Fri, 18 Sep 2009) | 2 lines

Support for Grandstream GXP-280 in the backend.

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r18398 | raarts | 2009-09-10 14:15:36 +0200 (Thu, 10 Sep 2009) | 2 lines

Remove the guess-zaptel script. Should be replaced by a much smarter PHP script.

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r18397 | raarts | 2009-09-10 14:13:19 +0200 (Thu, 10 Sep 2009) | 3 lines

[122096]: VLAN configuratie WebGUI issue.
Phone configs were not regenerated when VLAN settings changed. Fixed.

------------------------------------------------------------------------
r18396 | ivar | 2009-09-10 12:25:41 +0200 (Thu, 10 Sep 2009) | 1 line

(NeoNova 122121) Convert number (removing +) after checking for fastdial number
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r18374 | raarts | 2009-09-09 12:04:33 +0200 (Wed, 09 Sep 2009) | 2 lines

Add guess-zaptel script so we can scan for PCI cards at installation time, and add them automatically to the database.

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r18361 | raarts | 2009-09-08 21:59:35 +0200 (Tue, 08 Sep 2009) | 2 lines

astcfgd also depends on fedtival and sox.

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r18360 | raarts | 2009-09-08 21:46:23 +0200 (Tue, 08 Sep 2009) | 2 lines

No warnings during rpm installation: don't create plcmspip home dirs, redirect passwd output to /dev/null.

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r18355 | raarts | 2009-09-08 18:05:54 +0200 (Tue, 08 Sep 2009) | 2 lines

Fixes for 'yum install astium'.

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r18352 | raarts | 2009-09-08 15:52:40 +0200 (Tue, 08 Sep 2009) | 2 lines

Template file moved to another directory.

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r18317 | raarts | 2009-09-08 10:58:01 +0200 (Tue, 08 Sep 2009) | 2 lines

Move creation of /var/run/corefiles to astium-setup.

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r18310 | raarts | 2009-09-08 09:53:58 +0200 (Tue, 08 Sep 2009) | 3 lines

When a SNOM phone is paged, we need to wait 1 second first, to make sure
the phone has been free for 1 second, otherwise paging does not work reliably.

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r18305 | raarts | 2009-09-07 17:07:36 +0200 (Mon, 07 Sep 2009) | 2 lines

Add two missing dependancies to astcfgd (while making 'yum install astium' possible).

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r18290 | raarts | 2009-09-07 10:15:58 +0200 (Mon, 07 Sep 2009) | 2 lines

Fixed typo.

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r18288 | raarts | 2009-09-07 10:03:13 +0200 (Mon, 07 Sep 2009) | 9 lines

[124577]: DBGA : Switchboard doorverbonden calls op Handsfree // M0908 1733
This was a case where a call was:

  1. sent to the switchbaord, then 
  2. accepted by pressing spacebar, and then 
  3. ON THE PHONE ITSELF blind-transferred to a phone OF THE SAME BRAND, 

then that target phone was put into auto-answer mode. Fixed.

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r18286 | raarts | 2009-09-06 21:32:29 +0200 (Sun, 06 Sep 2009) | 2 lines

Improvements to the ha scripts.

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r18282 | raarts | 2009-09-04 17:12:11 +0200 (Fri, 04 Sep 2009) | 2 lines

When astcfgd encounters errors during configuration because of locking, continue, don't abort.

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r18281 | nnrelman | 2009-09-04 16:31:20 +0200 (Fri, 04 Sep 2009) | 1 line

added some files to the spec file.
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r18280 | raarts | 2009-09-04 16:16:04 +0200 (Fri, 04 Sep 2009) | 2 lines

Add the new HA scripts to astcfgd.

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r18276 | raarts | 2009-09-04 15:07:04 +0200 (Fri, 04 Sep 2009) | 7 lines

Incoming calls on SIP trunks sometimes come in on the wrong SIP trunk. That's why we drop them into the general [inbound] context.
This has the disadvantage that calls to internal extensions are not recognized. There is no way around that at the moment.

But IAX trunks don't have this problem. So that's why we drop those into the [company-inbound] context, where internal extensions
are recognized as well. Ergo: if you have to connect multiple Astiums together and want to call internal extensions: this *only*
works on IAX trunks.

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r18273 | raarts | 2009-09-04 12:09:18 +0200 (Fri, 04 Sep 2009) | 2 lines

(Scratching my own itch here..) For IAX software client users, show the fact that you are paging them by prefixing PAGE: to the CallerID Name. The IAX client now should auto-answer. At least Loudhush on the Mac supports this.

------------------------------------------------------------------------
r18259 | raarts | 2009-09-03 21:17:53 +0200 (Thu, 03 Sep 2009) | 4 lines

Remove forced translation to national format. NOTE: this update may impact incoming calls.
If incoming calls get "This number is disconnected or out of service", check the number translation
on incoming trunks, and set it to whatever type of number you have in the inbound number table..

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r18253 | raarts | 2009-09-03 16:02:39 +0200 (Thu, 03 Sep 2009) | 2 lines

Ensure that *36 calls can be recorded as well, so switchboard-managed calls can be recorded.

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r18244 | raarts | 2009-09-03 09:48:17 +0200 (Thu, 03 Sep 2009) | 4 lines

Prevent the dialplan from running around in circles if the data concerning the
extension could not be found (for example in case of a database inconsistency).
Now give a 'Extension does not exist' voice message.

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r18242 | raarts | 2009-09-02 17:33:10 +0200 (Wed, 02 Sep 2009) | 3 lines

Add mac address to SIP configurration. Used by the reversed-engineered remote-reboot-through-SIP-NOTIFY feature for Grandstream phones.
See: https://issues.asterisk.org/view.php?id=7667.

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r18227 | ivar | 2009-09-02 12:26:45 +0200 (Wed, 02 Sep 2009) | 1 line

New field: allowinempty for queues. Replaces 'joinempty' and 'leavewhenempty'
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r18219 | raarts | 2009-09-01 16:56:22 +0200 (Tue, 01 Sep 2009) | 2 lines

Also handle the case when a voicemail is being deleted from the GUI without listeningg to it.

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r18213 | raarts | 2009-09-01 14:25:36 +0200 (Tue, 01 Sep 2009) | 2 lines

Fixing Web voicemail, backend stuff.

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r18204 | ivar | 2009-09-01 11:08:04 +0200 (Tue, 01 Sep 2009) | 1 line

Bugfix: when mailing recorded calls, and file is not found, stop trying to mail it anyway
------------------------------------------------------------------------
r18180 | raarts | 2009-08-28 10:09:05 +0200 (Fri, 28 Aug 2009) | 4 lines

Sync copying logfiles by asrtcfgd with asterisk reload. Hopefully 
this fixes the long-standing reload problem where everything
seems to lock up and asterisk needs to be restarted.

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r18176 | raarts | 2009-08-28 09:44:00 +0200 (Fri, 28 Aug 2009) | 2 lines

Never auto-logout software faxes.

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r18162 | ivar | 2009-08-27 10:07:12 +0200 (Thu, 27 Aug 2009) | 4 lines

Bugfix: after upgrade to 2.0, astcfgd would always crash.
This was because the file containing the astiumid would be empty.
Now astcfgd will deal with that by putting the default value of '2' in the astiumid file,
and will keep running.
------------------------------------------------------------------------
r18158 | raarts | 2009-08-26 15:50:35 +0200 (Wed, 26 Aug 2009) | 2 lines

Add trunkid to the CALLDATA message, so even funny SIP trunks will be recognized as a trunk by astiumd.

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r18157 | raarts | 2009-08-26 15:25:27 +0200 (Wed, 26 Aug 2009) | 2 lines

Fix *9000 and friends not working in the IVR.

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r18137 | raarts | 2009-08-25 17:27:05 +0200 (Tue, 25 Aug 2009) | 2 lines

Only try to find which userid the channel belongs to, if you don't know it yet...

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r18128 | raarts | 2009-08-24 17:42:17 +0200 (Mon, 24 Aug 2009) | 4 lines

When PBX can terminate a call, in the end it tries the configured receptionist.
This was erroneously logged as 'Presses '0' for receptionist while in voice mail box xxx'.
Fix, it now just logs: 'Forward to receptionist'.

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r18126 | raarts | 2009-08-24 16:43:44 +0200 (Mon, 24 Aug 2009) | 2 lines

Ensure that logged-in users cannot disrupt the system too much

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r18124 | raarts | 2009-08-24 14:25:47 +0200 (Mon, 24 Aug 2009) | 2 lines

Some fixes after testing Grandstream GXV-2020 provisioning.

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r18123 | raarts | 2009-08-24 13:48:30 +0200 (Mon, 24 Aug 2009) | 2 lines

Fixed strange character in tftp.php.

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r18122 | raarts | 2009-08-24 13:08:58 +0200 (Mon, 24 Aug 2009) | 3 lines

Added support for Grandstream GXP-2020 in the backend. Also fixed some omissions
in other Grandstream phonetype support.

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r18112 | raarts | 2009-08-24 11:05:21 +0200 (Mon, 24 Aug 2009) | 2 lines

Support for Aastra 6730i/6731i in the backend.

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r18107 | raarts | 2009-08-21 16:42:43 +0200 (Fri, 21 Aug 2009) | 2 lines

Fix bug with wiping CALLERID number where outprefixes were not used.

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r18104 | raarts | 2009-08-21 15:35:57 +0200 (Fri, 21 Aug 2009) | 2 lines

Rearrange operator code in the dialplan a bit, so the calldetail screen shows the correct data when CallForward to operator is active.

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r18083 | raarts | 2009-08-20 16:33:56 +0200 (Thu, 20 Aug 2009) | 2 lines

From now on, we delete all CDR and recording data older than two years.

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r18082 | raarts | 2009-08-20 16:28:24 +0200 (Thu, 20 Aug 2009) | 2 lines

Drop year-old records from astiumcdr instead of cdr which does not exist any more.

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r18046 | raarts | 2009-08-18 15:33:25 +0200 (Tue, 18 Aug 2009) | 2 lines

Fix hardcoded neonova text in IVR fix.

------------------------------------------------------------------------
r18042 | ivar | 2009-08-18 13:07:36 +0200 (Tue, 18 Aug 2009) | 1 line

(NeoNova 121643) Bugfix: time schedule on an extensions wouldn't work if call was coming from an ivr menu. Fixed.
------------------------------------------------------------------------
r18037 | ivar | 2009-08-18 11:39:16 +0200 (Tue, 18 Aug 2009) | 1 line

Bugfix: dialing alarm numbers when logged out didn't work anymore. Fixed.
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r18031 | nico | 2009-08-18 10:05:50 +0200 (Tue, 18 Aug 2009) | 2 lines

Merge in 2.0 18024 from 1.6: Dialplan changes to tell astiumd the reason a call has ended.

------------------------------------------------------------------------
r18007 | raarts | 2009-08-16 21:19:55 +0200 (Sun, 16 Aug 2009) | 2 lines

Fixed two spelling mistakes.

------------------------------------------------------------------------
r18003 | raarts | 2009-08-15 11:59:57 +0200 (Sat, 15 Aug 2009) | 3 lines

Fall back to ntp-pool.neonova.nl if none of the
ntp_server_* astglobals are filled in.

------------------------------------------------------------------------
r18002 | raarts | 2009-08-15 11:17:35 +0200 (Sat, 15 Aug 2009) | 6 lines

in 1.6 revision 15977 was made:
When generating ntp configuration,
* only generate /etc/ntpd.conf - not all other files,
* take ntp server from database instead of hardcoded ntp-pool.neonova.nl
This wasn't ported forward to 2.0, so do that now..

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r17992 | raarts | 2009-08-13 16:21:02 +0200 (Thu, 13 Aug 2009) | 3 lines

Second part of fixing the showing of external callerid when 
calliing companies that are on the same Astium.

------------------------------------------------------------------------
r17991 | raarts | 2009-08-13 15:56:57 +0200 (Thu, 13 Aug 2009) | 4 lines

On incoming calls, only set the company shortname and id if not already set.
Partial fix for a bug where internal calls betweeen companies didn't show
the correct CallerID.

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r17966 | raarts | 2009-08-12 17:28:07 +0200 (Wed, 12 Aug 2009) | 2 lines

Also allow recording midcall by using *3 for queue calls.

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r17960 | raarts | 2009-08-12 15:00:59 +0200 (Wed, 12 Aug 2009) | 5 lines

Removed a block of code Ivar thought should stay.
From now on: reroute external calls internally if we should handle them anyway.
This *does* have the disadvantage that they are treated as internal calls,
whatever that means,

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r17940 | raarts | 2009-08-11 15:47:49 +0200 (Tue, 11 Aug 2009) | 2 lines

Grandstream GXP 2000/2020 also support Auto-answer/Paging. Enabled.

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r17922 | raarts | 2009-08-10 12:11:50 +0200 (Mon, 10 Aug 2009) | 3 lines

Extend mixmonitor to mix correctly even when *3 is pressed
multiple times during one conversation.

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r17920 | raarts | 2009-08-10 11:59:09 +0200 (Mon, 10 Aug 2009) | 2 lines

Allow parties to start/stop recording by pressing *3.

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r17917 | raarts | 2009-08-10 11:36:45 +0200 (Mon, 10 Aug 2009) | 3 lines

When dialing an invalid extension, fallback if you can. This helps
operator and employees who do lind transfers to invalid numbers.

------------------------------------------------------------------------
r17916 | ivar | 2009-08-10 11:35:36 +0200 (Mon, 10 Aug 2009) | 2 lines

Bugfix: don't try to offer 'callback dialing 5' feature for calls coming through a queue.
Because that feature will actually answer the call, stopping the Queue application from trying to offer the call to other agents in the queue.
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r17909 | raarts | 2009-08-07 16:11:12 +0200 (Fri, 07 Aug 2009) | 2 lines

Fix 'Error in extension logic' warning.

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r17905 | raarts | 2009-08-07 12:22:31 +0200 (Fri, 07 Aug 2009) | 2 lines

Allow overriding stdastium macros from a plugin.

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r17897 | ivar | 2009-08-05 14:20:40 +0200 (Wed, 05 Aug 2009) | 2 lines

New field: allow_direct_out for trunk,
to indicate that incoming calls over that trunk are allowed to go outbound, if the called number is not in the inbound numberplan
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r17891 | raarts | 2009-08-05 11:48:55 +0200 (Wed, 05 Aug 2009) | 2 lines

Allow for plugins to modify the inbound dialplan context as well.

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r17890 | raarts | 2009-08-05 11:48:06 +0200 (Wed, 05 Aug 2009) | 5 lines

This was a bugreport without corresponding CMS task made by GM telematisca.
When an incoming call had callerid suppressed, the Astium could not change the
CallerID any more, so we could for example not display on the agent phone 
via which queue this call entered. Fixed.

------------------------------------------------------------------------
r17888 | ivar | 2009-08-05 11:18:37 +0200 (Wed, 05 Aug 2009) | 3 lines

On sip trunks, don't set companyid variable.
This is because incoming calls over sip trunks can seem to come in at any random trunk which registers on the same ip address. Therefore we don't know for which company the incoming call is.
(NeoNova 122914)
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r17874 | ivar | 2009-08-04 12:09:51 +0200 (Tue, 04 Aug 2009) | 1 line

Bugfix: on snom phones, retreive button for voicemail was broken
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r17830 | ivar | 2009-07-30 12:46:13 +0200 (Thu, 30 Jul 2009) | 2 lines

Bugfix: when installing a new Astium, astcfgd won't run because there's no astiumid file present.
Fix this bug by creating the file if it doesn't exist.
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r17791 | ivar | 2009-07-27 12:06:19 +0200 (Mon, 27 Jul 2009) | 3 lines

Add two new scripts:
one to show current astium version and database version,
one to show current number of queued activate records.
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r17790 | nnrelman | 2009-07-27 11:57:59 +0200 (Mon, 27 Jul 2009) | 1 line

added /etc/profile.d/astcfgd.sh
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r17769 | ivar | 2009-07-22 11:33:33 +0200 (Wed, 22 Jul 2009) | 1 line

Bugfix: DealIT had strange problem, that when using Polycom 500 phone, and talking to someone on the phone, when another call would come in (callwaiting = on), then user of the Polycom wouldn't hear the person they were talking with anymore. This problem would occur even before the new call coming in was answered. Fixed by letting Polycom 300 and Polycom 500 retreive another sip.cfg file.
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r17757 | ivar | 2009-07-17 09:39:18 +0200 (Fri, 17 Jul 2009) | 3 lines

Bugfix: lately astcfgd would crash on Astiums, saying:
Invalid SQL: update astium set zapchanged = 'yes' where id = ''
Fixed.
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r17755 | ivar | 2009-07-16 12:08:57 +0200 (Thu, 16 Jul 2009) | 1 line

New phonetype: Polycom 450
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r17754 | ivar | 2009-07-16 12:07:15 +0200 (Thu, 16 Jul 2009) | 1 line

New phonetype: Polycom 450
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r17748 | ivar | 2009-07-16 11:09:10 +0200 (Thu, 16 Jul 2009) | 3 lines

Make it possible for incoming calls to go outbound again.
Done for a situation where client as a gsm solution connected to the Astium with a sip trunk.
Incoming calls over the sip trunk should be allowed to go out over isdn.
------------------------------------------------------------------------
r17745 | ivar | 2009-07-14 12:48:23 +0200 (Tue, 14 Jul 2009) | 4 lines

2 bugfixes;
1. Astium trying to internationalise called phonenumbers before sending them out over a trunk causes lots of problems.
Make sure this only happens when the called number has sufficient length.
2. For sip trunks belonging to a company, the company variable wouldn't be set.
------------------------------------------------------------------------
r17741 | ivar | 2009-07-13 13:17:57 +0200 (Mon, 13 Jul 2009) | 1 line

Bugfix: only mail recording if the involved phoneline is configured to do so
------------------------------------------------------------------------
r17740 | ivar | 2009-07-13 12:56:54 +0200 (Mon, 13 Jul 2009) | 1 line

Add required include file for astcfgd
------------------------------------------------------------------------
r17739 | ivar | 2009-07-13 12:45:26 +0200 (Mon, 13 Jul 2009) | 1 line

(NeoNova 121947) Mailing of recorded calls was broken. Implemented again.
------------------------------------------------------------------------
r17724 | ivar | 2009-07-06 14:20:27 +0200 (Mon, 06 Jul 2009) | 3 lines

(NeoNova 121872) Problem: If a SNOM phone is set to Do Not Disturb, the Astium follows callforward settings of 'logged out' instead of 'busy'.
We don't really want this. But we do want other cases of Congestion to be handled as 'logged out', for example when a wifi sip client is taken out of the office, and its SIP registration ends.
Solution is to distinguish on HANGUPCAUSE, if that is 19 then we know it's DND and it's treated as Busy.
------------------------------------------------------------------------
r17722 | ivar | 2009-07-06 12:45:50 +0200 (Mon, 06 Jul 2009) | 5 lines

(NeoNova 121872) Previously, if a phone which was dialed could not be reached / wasn't registered at asterisk, this was considered to be 'logged out' for callforward settings.
That was at some point changed such that in that case the phone would be considered 'busy'.
This change breaks current configurations on Astiums where callforward when logged out was used.
So this commit reverts the change.
The reason why the change was made will be solved in a different way, by changing SNOM provisioning such that when it is set to Do Not Disturb, it signals 'busy' instead of 'congestion'.
------------------------------------------------------------------------
r17719 | ivar | 2009-07-06 11:05:50 +0200 (Mon, 06 Jul 2009) | 3 lines

(NeoNova 121966) Bug: when dialing out to an external number,
hash transfer would work for that external phone, but not for the caller.
Fixed.
------------------------------------------------------------------------
r17693 | raarts | 2009-07-02 10:40:15 +0200 (Thu, 02 Jul 2009) | 4 lines

Fix incoming SIP calls from unknown IP addresses. G.729 was preferred codec, which
doesn't work on normal Astiums. Also had to set certain dialplan variables.
Fix incoming SIP calls sometimes coming in the wrong company.

------------------------------------------------------------------------
r17676 | nico | 2009-06-30 13:41:50 +0200 (Tue, 30 Jun 2009) | 5 lines

merge from 1.6 r17579
recording of conferencing now controlled by astiumd
from asterisk the meetmejoin event is send to astium where a session is started for the conference.
until the last participant has left all records including astiumcdr are written

------------------------------------------------------------------------
r17648 | raarts | 2009-06-29 15:21:43 +0200 (Mon, 29 Jun 2009) | 2 lines

Fix forwarding on queues/ivr etc in case an out_prefix is used.

------------------------------------------------------------------------
r17637 | ivar | 2009-06-29 11:33:44 +0200 (Mon, 29 Jun 2009) | 1 line

Fix forwarding external calls to outbound numbers
------------------------------------------------------------------------
r17636 | ivar | 2009-06-29 11:05:27 +0200 (Mon, 29 Jun 2009) | 3 lines

Bug: forwarding configured on phone itself didn't work, when call came from external.
Problem was that 'outplan' variabel was not set.
Fixed by calling 'take-call-responsability'
------------------------------------------------------------------------
r17633 | raarts | 2009-06-28 09:57:43 +0200 (Sun, 28 Jun 2009) | 5 lines

On each outgoing call set the class dialplan variable, and send it to astiumd.
We need this to get the outgoing costclass into the astiumcdr record. 
This in turn is needed for reporting, because I need to fix
[121494]: Rapportage bugs, items 4 and 5.

------------------------------------------------------------------------
r17632 | raarts | 2009-06-28 09:23:46 +0200 (Sun, 28 Jun 2009) | 3 lines

Redirect all emergancy calls to the company-alarm context, so
the callerid will ALWAYS show the proper values for that.

------------------------------------------------------------------------
r17609 | raarts | 2009-06-25 16:10:33 +0200 (Thu, 25 Jun 2009) | 2 lines

Fix *36 for extension numbers.

------------------------------------------------------------------------
r17606 | raarts | 2009-06-25 14:01:25 +0200 (Thu, 25 Jun 2009) | 3 lines

If after a blind transfer the callee is busy, fall back to the person
that last talked the customer, and show it one the phone screen. 

------------------------------------------------------------------------
r17565 | raarts | 2009-06-22 17:30:47 +0200 (Mon, 22 Jun 2009) | 2 lines

Fix Blind/hash transfer WITH out_prefix.

------------------------------------------------------------------------
r17563 | raarts | 2009-06-22 16:26:29 +0200 (Mon, 22 Jun 2009) | 1 line

Don't prepend an outprefix on the callerid of internal calls
------------------------------------------------------------------------
r17562 | raarts | 2009-06-22 16:14:15 +0200 (Mon, 22 Jun 2009) | 1 line

remove extraneous code
------------------------------------------------------------------------
r17560 | raarts | 2009-06-22 16:11:28 +0200 (Mon, 22 Jun 2009) | 1 line

make *36 work in forwarding settings
------------------------------------------------------------------------
r17558 | raarts | 2009-06-22 15:33:05 +0200 (Mon, 22 Jun 2009) | 1 line

Fix not being able to forward external calls to outbound trunks
------------------------------------------------------------------------
r17554 | ivar | 2009-06-22 14:54:52 +0200 (Mon, 22 Jun 2009) | 1 line

Don't use seperate variable name for dialback prefix, just use out_prefix
------------------------------------------------------------------------
r17553 | ivar | 2009-06-22 14:53:13 +0200 (Mon, 22 Jun 2009) | 1 line

Don't use seperate variable name for dialback prefix, just use out_prefix
------------------------------------------------------------------------
r17550 | raarts | 2009-06-22 13:58:49 +0200 (Mon, 22 Jun 2009) | 1 line

unified all out_prefix number handling into one context for clarity
------------------------------------------------------------------------
r17547 | ivar | 2009-06-22 12:08:33 +0200 (Mon, 22 Jun 2009) | 1 line

Fix transfer to external numbers
------------------------------------------------------------------------
r17532 | ivar | 2009-06-19 14:01:22 +0200 (Fri, 19 Jun 2009) | 1 line

Bugfix: set outplan variable also when hunting
------------------------------------------------------------------------
r17520 | ivar | 2009-06-19 11:10:23 +0200 (Fri, 19 Jun 2009) | 1 line

Slightly increase timeout value for call to curl script which pushes new info to phone after login/logout
------------------------------------------------------------------------
r17496 | ivar | 2009-06-17 12:19:40 +0200 (Wed, 17 Jun 2009) | 2 lines

(NeoNova 121394) Bugfix: dialplan would loop when two collegues would
enter each others extension in 'hunt' ('zoek een collega') function.
------------------------------------------------------------------------
r17492 | ivar | 2009-06-17 10:33:18 +0200 (Wed, 17 Jun 2009) | 6 lines

(NeoNova 121395) Bugfix.
Example: inbound phonenumbers has two extensions, phoneline1 and phoneline2.
phoneline1 is set to 'refuse' when busy.
Then the incoming call should go to phoneline2
But that didn't happen, instead caller would get busy tone.

------------------------------------------------------------------------
r17485 | raarts | 2009-06-16 17:09:26 +0200 (Tue, 16 Jun 2009) | 3 lines

Move macro owncountry around, so we can make a workaround for
connecting internal Astiums together wrt number translation.

------------------------------------------------------------------------
r17478 | raarts | 2009-06-15 17:05:53 +0200 (Mon, 15 Jun 2009) | 3 lines

Fix the problem where you could not dial local numbers
starting with your country code.

------------------------------------------------------------------------
r17458 | raarts | 2009-06-12 11:53:42 +0200 (Fri, 12 Jun 2009) | 2 lines

Improve logging of phone rebooting/check-sync.

------------------------------------------------------------------------
r17456 | raarts | 2009-06-12 10:48:55 +0200 (Fri, 12 Jun 2009) | 2 lines

rebootphone.pl logged to syslog that its name was check-sync.pl. Corrected.

------------------------------------------------------------------------
r17452 | raarts | 2009-06-11 16:19:47 +0200 (Thu, 11 Jun 2009) | 2 lines

Fix loop in dialplan when anonymous click-2-dial is used.

------------------------------------------------------------------------
r17433 | nico | 2009-06-10 14:23:45 +0200 (Wed, 10 Jun 2009) | 2 lines

Merge from 1.6 -r 17432 trunc name in event 500 and fdname in reports

------------------------------------------------------------------------
r17424 | raarts | 2009-06-10 11:03:40 +0200 (Wed, 10 Jun 2009) | 2 lines

Renamed variable to contain an underscore instead of a dash.

------------------------------------------------------------------------
r17421 | nico | 2009-06-09 16:48:41 +0200 (Tue, 09 Jun 2009) | 2 lines

merged from 1.6 cms117492 r 17415-17416

------------------------------------------------------------------------
r17399 | raarts | 2009-06-09 11:21:53 +0200 (Tue, 09 Jun 2009) | 8 lines

On click-to-dial, the callerid/name of the first channel to the phone
are set to the destinartion number+name, so that will display nicely
on the phone screen.
As soon as the user picks up, the channel is send through the asterisk
dialplan (into the [phone] context). But now the callerid+name should
be set to the user's as if he/she picked up the phone and dialed himself.
Fix this.

------------------------------------------------------------------------
r17390 | ivar | 2009-06-09 09:45:47 +0200 (Tue, 09 Jun 2009) | 1 line

Previous commit made change it shouldn't have. Undo
------------------------------------------------------------------------
r17378 | raarts | 2009-06-08 15:38:14 +0200 (Mon, 08 Jun 2009) | 2 lines

Prevent dialplan loop

------------------------------------------------------------------------
r17377 | ivar | 2009-06-08 15:23:12 +0200 (Mon, 08 Jun 2009) | 1 line

Bugfix: outbound callrouting for number with same length as extension length was broken. Fixed.
------------------------------------------------------------------------
r17365 | raarts | 2009-06-08 11:42:01 +0200 (Mon, 08 Jun 2009) | 3 lines

Send expanded Dial string to astiumd, so it will
see any prepended areacodes.

------------------------------------------------------------------------
r17332 | ivar | 2009-06-02 11:57:50 +0200 (Tue, 02 Jun 2009) | 1 line

Support outgoing prefix for hunting ('find colleague' feature)
------------------------------------------------------------------------
r17324 | raarts | 2009-05-29 14:14:30 +0200 (Fri, 29 May 2009) | 2 lines

Fixed auto-answer after blind transfer by the switchboard.

------------------------------------------------------------------------
r17303 | raarts | 2009-05-28 16:46:34 +0200 (Thu, 28 May 2009) | 2 lines

Fix setting the correct zapchanged value when musiconhold changes.

------------------------------------------------------------------------
r17289 | ivar | 2009-05-28 13:10:52 +0200 (Thu, 28 May 2009) | 1 line

Add calvi script to cron, include file which has date in filename for calvi output
------------------------------------------------------------------------
r17268 | nico | 2009-05-27 16:08:01 +0200 (Wed, 27 May 2009) | 2 lines

Merge calvi and mixmonitor report from 1.6 into 2.0.

------------------------------------------------------------------------
r17254 | ivar | 2009-05-27 10:49:03 +0200 (Wed, 27 May 2009) | 2 lines

Bugfix: it seems that out_prefix wasn't retreived from database,
and also code tried to get result from query before query was done.
------------------------------------------------------------------------
r17219 | ivar | 2009-05-25 10:59:22 +0200 (Mon, 25 May 2009) | 1 line

Bugfix: when 'out' variable is set (to ***), prepend that to called number.
------------------------------------------------------------------------
r17182 | raarts | 2009-05-16 15:36:48 +0200 (Sat, 16 May 2009) | 4 lines

Committed the press-5-to-callback-when-colleague-is-busy 
dialplan mods. Sorry, hadn't been committed before.
Now it should work...

------------------------------------------------------------------------
r17175 | ivar | 2009-05-15 10:39:46 +0200 (Fri, 15 May 2009) | 3 lines

Changes to facilitate outgoing call prefix further.
Strip prefix from forwarded calls.
Strip prefix from fastdial numbers
------------------------------------------------------------------------
r17174 | raarts | 2009-05-14 22:52:21 +0200 (Thu, 14 May 2009) | 9 lines

Merged from 1.6:

------------------------------------------------------------------------
r17121 | raarts | 2009-05-12 11:57:06 +0200 (Tue, 12 May 2009) | 3 lines

Since recording is now handled by astiumd, remove all
recording stuff from the dialplan, and sip/iax/zapata.conf.


------------------------------------------------------------------------
r17173 | raarts | 2009-05-14 22:49:19 +0200 (Thu, 14 May 2009) | 19 lines

Merged from 1.6:

------------------------------------------------------------------------
r17050 | raarts | 2009-05-06 18:06:00 +0200 (Wed, 06 May 2009) | 13 lines

Fix 2 bugs in owncountry macro: 

1. If you are in the same country as the trunk, and you dial a
   number in your own country but in international format,
   than convert  it to national format.
   On the other hand, on SIP and IAX trunks, ALWAYS use ITU format
   numbers, even within your own country.

2. If you are in a different country from the trunk you dial out
   on, but you dial a number in your own country, convert that
   to international format. Previously the conversion was wrong
   and left out the country prefix. Fixed that.


------------------------------------------------------------------------
r17172 | raarts | 2009-05-14 22:48:37 +0200 (Thu, 14 May 2009) | 8 lines

Merged from 1.6:

------------------------------------------------------------------------
r17022 | nico | 2009-05-01 15:13:58 +0200 (Fri, 01 May 2009) | 2 lines

Add calvi report script.


------------------------------------------------------------------------
r17165 | ivar | 2009-05-14 20:17:17 +0200 (Thu, 14 May 2009) | 1 line

Bugfix: implement outgoing prefix for analog phones as well
------------------------------------------------------------------------
r17164 | ivar | 2009-05-14 19:00:26 +0200 (Thu, 14 May 2009) | 1 line

Bugfix: if *36 was not allowed, it would say 'not allowed' but then work anyway
------------------------------------------------------------------------
r17153 | ivar | 2009-05-13 12:53:49 +0200 (Wed, 13 May 2009) | 1 line

Prevent 'contacting redirector' message on aastra
------------------------------------------------------------------------
r17146 | ivar | 2009-05-13 10:19:38 +0200 (Wed, 13 May 2009) | 1 line

More support for Aastra 51i
------------------------------------------------------------------------
r17098 | raarts | 2009-05-10 22:52:35 +0200 (Sun, 10 May 2009) | 2 lines

Fix voicemail processing.

------------------------------------------------------------------------
r17057 | raarts | 2009-05-07 14:45:39 +0200 (Thu, 07 May 2009) | 2 lines

Support for Polycom 6000/7000 in the backend.

------------------------------------------------------------------------
r17056 | raarts | 2009-05-07 14:34:19 +0200 (Thu, 07 May 2009) | 2 lines

Support for WebGUI voicemail in astcfgd.

------------------------------------------------------------------------
r17029 | raarts | 2009-05-01 17:26:35 +0200 (Fri, 01 May 2009) | 2 lines

Moved some functions from vm_externnotify into functions.php.

------------------------------------------------------------------------
r17026 | raarts | 2009-05-01 16:38:38 +0200 (Fri, 01 May 2009) | 3 lines

Add extra channelvariable for callback-when-busy-and-press-5,
so we can display the callerdi properly on Originated callback calls.

------------------------------------------------------------------------
r17025 | raarts | 2009-05-01 16:37:10 +0200 (Fri, 01 May 2009) | 4 lines

Added the externnotify script. This ensures that whenever a voicemail
is left/removed/listened to, the viocemail table in the astium database 
is updated to reflect the actual state of the voicemail files on disk.

------------------------------------------------------------------------
r17000 | raarts | 2009-04-29 09:36:08 +0200 (Wed, 29 Apr 2009) | 2 lines

Fix query error in enabling firewall script.

------------------------------------------------------------------------
r16986 | raarts | 2009-04-28 12:22:10 +0200 (Tue, 28 Apr 2009) | 3 lines

Add phoneline id to the sip.conf/iax.conf and zapata.conf config files
when someone is logged in to a phone.

------------------------------------------------------------------------
r16973 | raarts | 2009-04-28 10:22:01 +0200 (Tue, 28 Apr 2009) | 2 lines

Re-implement the fix for high CPU load.

------------------------------------------------------------------------
r16966 | raarts | 2009-04-27 17:06:12 +0200 (Mon, 27 Apr 2009) | 3 lines

When editing system locations, rebuild configs for all companies.
Dialing across companies works again, including invalid extensions.

------------------------------------------------------------------------
r16964 | raarts | 2009-04-27 15:46:39 +0200 (Mon, 27 Apr 2009) | 5 lines

Re-applied some missing fixes/features from 1.6 by hand. They were:
- Pickup Groups
- Only generate when zap)_channels are active
and some more I don't remember right now...

------------------------------------------------------------------------
r16928 | ivar | 2009-04-23 10:38:35 +0200 (Thu, 23 Apr 2009) | 4 lines

Improve handling of prepending dialback prefix.
Sometimes a Dial is called before the previous Dial was completed.
In that case the dialback prefix should not be prepended again.
So keep the fact that the prefix was prepended in a variable, which we check before we want to prepend again.
------------------------------------------------------------------------
r16926 | ivar | 2009-04-23 09:52:31 +0200 (Thu, 23 Apr 2009) | 2 lines

If a non-existing extension is dialed, tell the person who is dialing that it is an invalid extension,
instead of trying to call the invalid extension on a trunk
------------------------------------------------------------------------
r16921 | ivar | 2009-04-22 11:50:07 +0200 (Wed, 22 Apr 2009) | 7 lines

re-introduce checking permissions before paging a phone.
Rules:
* if the callers' permission 'allow page' is set to 'yes', he/she can page everyone,
* if the callee's permission 'switchboard operator' is set to 'yes', everybody can page him/her.

So it's not neccessary for both permisssions to be set to 'yes' for paging to work.
Maybe in the future we need another option which says that for a certain person, nobody can page that person.
------------------------------------------------------------------------
r16914 | ivar | 2009-04-22 11:00:51 +0200 (Wed, 22 Apr 2009) | 1 line

re-introduce 'break through callforwards' feature which was gone when we copied the 1.6 stdastium.inc file to 2.0 again
------------------------------------------------------------------------
r16907 | raarts | 2009-04-22 09:56:46 +0200 (Wed, 22 Apr 2009) | 3 lines

Copied the stdastium.inc and extensions.conf from 1.6 into 2.0
This is the basis we will work on from now.

------------------------------------------------------------------------
r16871 | ivar | 2009-04-20 11:07:54 +0200 (Mon, 20 Apr 2009) | 1 line

Merged changes from 1.9 to 2.0
------------------------------------------------------------------------
r16747 | ivar | 2009-04-10 11:45:25 +0200 (Fri, 10 Apr 2009) | 1 line

show in display if call falls back to operator
------------------------------------------------------------------------
r16628 | ivar | 2009-03-24 09:56:42 +0100 (Tue, 24 Mar 2009) | 1 line

add missing '>'
------------------------------------------------------------------------
r16572 | ivar | 2009-03-18 13:59:55 +0100 (Wed, 18 Mar 2009) | 1 line

set svnmerge init to 16162 1.9
------------------------------------------------------------------------
r16544 | raarts | 2009-03-16 21:10:58 +0100 (Mon, 16 Mar 2009) | 2 lines

When an astium is added, let astcfgd set the astium md5sum.

------------------------------------------------------------------------
r16343 | nnrelman | 2009-02-27 13:26:51 +0100 (Fri, 27 Feb 2009) | 1 line

Added some backup options
------------------------------------------------------------------------
r16133 | nnrelman | 2009-02-05 11:09:06 +0100 (Thu, 05 Feb 2009) | 1 line

Process xml just as php files.
------------------------------------------------------------------------
r15960 | nnrelman | 2009-01-20 10:37:25 +0100 (Tue, 20 Jan 2009) | 1 line

Added some comments. When in debugging mode, script wil show memory usage.
------------------------------------------------------------------------
r15952 | ivar | 2009-01-16 14:01:55 +0100 (Fri, 16 Jan 2009) | 1 line

Have astcfgd use new class.provision.inc for generating aastra configuration files
------------------------------------------------------------------------
r15920 | raarts | 2009-01-09 16:35:02 +0100 (Fri, 09 Jan 2009) | 3 lines

Make sure astcfgd looks at the astium ID while
creating its configuration.

------------------------------------------------------------------------
r15913 | ivar | 2009-01-09 14:38:01 +0100 (Fri, 09 Jan 2009) | 2 lines

Another change in choice of directory name.
aastra configuration files to be fetched through HTTP can now be found in web/html/ps/aastra/cfg
------------------------------------------------------------------------
r15912 | ivar | 2009-01-09 14:27:25 +0100 (Fri, 09 Jan 2009) | 1 line

place aastra configuration files in web path, in aastra/configuration directory
------------------------------------------------------------------------
r15911 | ivar | 2009-01-09 14:16:00 +0100 (Fri, 09 Jan 2009) | 3 lines

Aastra's will be provisioned through HTTP instead of TFTP.
aastra.cfg which is generic for all phones will still be retreived with TFTP,
that file contains the instruction to get the rest (including firmware, and aastra.cfg again) with HTTP.
------------------------------------------------------------------------
r15910 | raarts | 2009-01-09 14:04:43 +0100 (Fri, 09 Jan 2009) | 2 lines

Removed debug info

------------------------------------------------------------------------
r15909 | raarts | 2009-01-09 13:32:30 +0100 (Fri, 09 Jan 2009) | 5 lines

Let astcfgd read the id of the astium it is running on,
so it can determine what kind of config files to generate
for the PCI cards, trunks and firewall rules.
This is located in /usr/local/astium/etc/astiumid.

------------------------------------------------------------------------
r15852 | raarts | 2008-12-30 21:08:47 +0100 (Tue, 30 Dec 2008) | 3 lines

Ensure that extensions.conf.local exists, otherwise
asterisk 1.6 barfs, and won't start.

------------------------------------------------------------------------
r15851 | raarts | 2008-12-30 17:30:44 +0100 (Tue, 30 Dec 2008) | 2 lines

Add some dmmy files in the plugins directory, or asterisk 1.6 would not load.

------------------------------------------------------------------------
r15838 | raarts | 2008-12-30 11:03:12 +0100 (Tue, 30 Dec 2008) | 2 lines

Fix the asterisk logrotate script we provide with astcfgd.

------------------------------------------------------------------------
r15822 | raarts | 2008-12-29 12:35:34 +0100 (Mon, 29 Dec 2008) | 2 lines

Bumped astcfgd VERSION to 2.0

------------------------------------------------------------------------
r15813 | raarts | 2008-12-28 17:21:50 +0100 (Sun, 28 Dec 2008) | 2 lines

Merge recent changes in 1.9 into 2.0 (astcfgd).

------------------------------------------------------------------------
r15810 | raarts | 2008-12-28 17:07:44 +0100 (Sun, 28 Dec 2008) | 2 lines

Merged recent changes in 1.6 into astcfgd 2.0.

------------------------------------------------------------------------
r15684 | nnrelman | 2008-12-09 14:07:21 +0100 (Tue, 09 Dec 2008) | 1 line

Added some extra options for the backup.
------------------------------------------------------------------------
r15680 | nnrelman | 2008-12-09 12:48:53 +0100 (Tue, 09 Dec 2008) | 1 line

added some more options to the backup config file.
------------------------------------------------------------------------
r15657 | nnrelman | 2008-12-08 15:06:47 +0100 (Mon, 08 Dec 2008) | 1 line

Added the astium serial to the backup config file.
------------------------------------------------------------------------
r15645 | raarts | 2008-12-08 12:01:59 +0100 (Mon, 08 Dec 2008) | 2 lines

Changes in astcfgd for new backup procedure.

------------------------------------------------------------------------
r15526 | nnrelman | 2008-11-20 22:09:55 +0100 (Thu, 20 Nov 2008) | 1 line

start 2.0 from 1.9
------------------------------------------------------------------------
r15366 | ivar | 2008-10-30 15:31:52 +0100 (Thu, 30 Oct 2008) | 2 lines

Spy function disabled.

------------------------------------------------------------------------
r15362 | ivar | 2008-10-30 13:34:25 +0100 (Thu, 30 Oct 2008) | 2 lines

Disable paging by default for IP Business Manager 1.0

------------------------------------------------------------------------
r15216 | ivar | 2008-10-09 11:10:35 +0200 (Thu, 09 Oct 2008) | 1 line

Merged changes from 1.6 to 1.9
------------------------------------------------------------------------
r15208 | ivar | 2008-10-09 10:36:22 +0200 (Thu, 09 Oct 2008) | 1 line

more useful 'svnmerge init'?
------------------------------------------------------------------------
r15207 | ivar | 2008-10-09 10:32:50 +0200 (Thu, 09 Oct 2008) | 1 line

'svnmerge init' done. Was that useful?
------------------------------------------------------------------------
r14868 | ivar | 2008-08-21 13:51:54 +0200 (Thu, 21 Aug 2008) | 1 line

incorporate permissions on new bypass callforward feature
------------------------------------------------------------------------
r14832 | ivar | 2008-08-18 16:00:04 +0200 (Mon, 18 Aug 2008) | 2 lines

New feature: dial *60<extension> to ignore callforward settings of that extension.
To be made: extra permission for each person, which indicates that the person is allowed to use this new feature.
------------------------------------------------------------------------
r14825 | ivar | 2008-08-18 13:37:27 +0200 (Mon, 18 Aug 2008) | 1 line

disable voicemail on/off/status
------------------------------------------------------------------------
r14776 | ivar | 2008-08-13 12:35:26 +0200 (Wed, 13 Aug 2008) | 1 line

Enable new phonebook on aastra phones
------------------------------------------------------------------------
r14673 | ivar | 2008-08-04 13:37:19 +0200 (Mon, 04 Aug 2008) | 1 line

make aastra.cfg more generic again
------------------------------------------------------------------------
r14641 | ivar | 2008-07-31 10:38:18 +0200 (Thu, 31 Jul 2008) | 1 line

Disable spy- and page feature for city of Amsterdam (1.9 only)
------------------------------------------------------------------------
r14640 | ivar | 2008-07-31 10:35:11 +0200 (Thu, 31 Jul 2008) | 1 line

aastra.cfg specific for city of Amsterdam (1.9 only)
------------------------------------------------------------------------
r14477 | ivar | 2008-07-16 11:07:11 +0200 (Wed, 16 Jul 2008) | 18 lines

svnmerge, conflict:

<<<<<<< .working
        $out .= sprintf("exten => %s,n,Set(redir=)\n", $extension);
        $out .= sprintf("exten => %s,n,PGfget(redir=activeschedule(%s))\n", $extension, $phoneline);
        $out .= sprintf("exten => %s,n,GotoIf(\$[\"foo\${redir}\" != \"foo\"]?%s-extensions,\${redir},1)\n",
                        $extension, $shortname);
        $out .= sprintf("exten => %s,n,Macro(extension,%s,%s,\${%s-%s},\${DIAL_TIMEOUT},) ; %s %s\n",
                        $extension, $shortname, $extension, $shortname, $extension, $function, $name);
======= 
        $out .= sprintf("exten => %s,n,Set(ext_company=%s)\n", $extension, $shortname);
        $out .= sprintf("exten => %s,n,Set(ext_number=%s)\n", $extension, $extension);
        $out .= sprintf("exten => %s,n,Set(ext_device=\${%s-%s})\n", $extension, $shortname, $extension);
        $out .= sprintf("exten => %s,n,Gosub(extension,s,1) ; %s %s\n", $extension, $function, $name);
>>>>>>> .merge-right.r14476



------------------------------------------------------------------------
r14473 | ivar | 2008-07-16 10:49:17 +0200 (Wed, 16 Jul 2008) | 1 line

svnmerge init
------------------------------------------------------------------------
r14175 | ivar | 2008-05-21 15:22:11 +0200 (Wed, 21 May 2008) | 1 line

New feature: time schedules available on every extension
------------------------------------------------------------------------
r14158 | nnrelman | 2008-05-20 14:17:34 +0200 (Tue, 20 May 2008) | 1 line

updated version number from 1.6 to 1.9.
------------------------------------------------------------------------
r14157 | nnrelman | 2008-05-20 14:00:14 +0200 (Tue, 20 May 2008) | 1 line

creating astcfgd version 1.9
------------------------------------------------------------------------
r14152 | raarts | 2008-05-20 09:36:05 +0200 (Tue, 20 May 2008) | 2 lines

G.729 support voor Polycom phones.

------------------------------------------------------------------------
r14089 | ivar | 2008-04-22 12:39:58 +0200 (Tue, 22 Apr 2008) | 1 line

Bugfix requested by Patrick
------------------------------------------------------------------------
r14087 | ivar | 2008-04-22 12:27:40 +0200 (Tue, 22 Apr 2008) | 1 line

Bugfix by Patrick
------------------------------------------------------------------------
r14082 | ivar | 2008-04-22 10:54:10 +0200 (Tue, 22 Apr 2008) | 1 line

Revert to old-style address book because new one causes complaints
------------------------------------------------------------------------
r14063 | ivar | 2008-04-21 10:27:49 +0200 (Mon, 21 Apr 2008) | 1 line

Bugfix: always join contact in query. Otherwise agents with queue pause aren't always on queue pause
------------------------------------------------------------------------
r14056 | raarts | 2008-04-18 21:26:48 +0200 (Fri, 18 Apr 2008) | 3 lines

Ensure that astcfgd skips trunks that are set to 'disabled',
so no config will be generated for them.

------------------------------------------------------------------------
r14036 | raarts | 2008-04-16 17:36:17 +0200 (Wed, 16 Apr 2008) | 2 lines

Fix cisco 790x regardless of using capitals in the max addres when donwload configfile.

------------------------------------------------------------------------
r13974 | raarts | 2008-04-12 18:00:50 +0200 (Sat, 12 Apr 2008) | 2 lines

Fix some bugs where the CDR was not filled correctly with phonenumbers.

------------------------------------------------------------------------
r13972 | raarts | 2008-04-11 18:10:59 +0200 (Fri, 11 Apr 2008) | 3 lines

In the ask caller menu, options that were checked in the
GUI were not announced, but could still be chosen. Fixed.

------------------------------------------------------------------------
r13953 | ivar | 2008-04-10 12:48:08 +0200 (Thu, 10 Apr 2008) | 1 line

Update: when choosing 'telefoonboek' for aastra softkey, always use dynamic phonebook
------------------------------------------------------------------------
r13943 | raarts | 2008-04-10 11:56:32 +0200 (Thu, 10 Apr 2008) | 2 lines

Regenerate pending zap changes at boot time, since we are booting anyway..

------------------------------------------------------------------------
r13925 | ivar | 2008-04-10 09:42:37 +0200 (Thu, 10 Apr 2008) | 1 line

bugfix: function was called incorrectly
------------------------------------------------------------------------
r13920 | raarts | 2008-04-08 14:04:55 +0200 (Tue, 08 Apr 2008) | 2 lines

Remove pedantic setting

------------------------------------------------------------------------
r13916 | ivar | 2008-04-07 15:55:18 +0200 (Mon, 07 Apr 2008) | 2 lines

Change: if language is dutch, choose language pack lang_nl.txt for aastra phone.
Language pack is not committed yet (but it is available)
------------------------------------------------------------------------
r13911 | raarts | 2008-04-07 11:41:36 +0200 (Mon, 07 Apr 2008) | 2 lines

More ask caller fixes

------------------------------------------------------------------------
r13908 | raarts | 2008-04-07 10:06:59 +0200 (Mon, 07 Apr 2008) | 2 lines

More fixes for ask caller. 

------------------------------------------------------------------------
r13906 | raarts | 2008-04-07 00:09:42 +0200 (Mon, 07 Apr 2008) | 3 lines

Ask caller only worked for the unconditional column. Not 
for loggedoff, busy or no answer. Fixed.

------------------------------------------------------------------------
r13895 | raarts | 2008-04-06 09:53:08 +0200 (Sun, 06 Apr 2008) | 3 lines

System never asked 'for operator press 1' even when that
was selected in the forwarding options. Fixed.

------------------------------------------------------------------------
r13868 | raarts | 2008-04-02 08:48:14 +0200 (Wed, 02 Apr 2008) | 3 lines

Let iaxmodem processes register themselves more often to the 
asterisk. We have problems with iaxmodems losing registration.

------------------------------------------------------------------------
r13851 | raarts | 2008-03-31 11:15:15 +0200 (Mon, 31 Mar 2008) | 3 lines

Only internationalize in the outbound dialplan when the flag
in the company table is set to yes.

------------------------------------------------------------------------
r13826 | raarts | 2008-03-26 16:13:05 +0100 (Wed, 26 Mar 2008) | 2 lines

Also record SOAP-initiated calls (click2dial and callmenow).

------------------------------------------------------------------------
r13794 | raarts | 2008-03-21 17:01:35 +0100 (Fri, 21 Mar 2008) | 2 lines

Fix dialplan loop.

------------------------------------------------------------------------
r13793 | raarts | 2008-03-21 16:13:45 +0100 (Fri, 21 Mar 2008) | 2 lines

ResponseTimeout and DigitTimeout are deprecated.

------------------------------------------------------------------------
r13789 | ivar | 2008-03-21 15:23:09 +0100 (Fri, 21 Mar 2008) | 2 lines

Place vlan configuration in aastra.cfg.
This is done for now to be sure, as at Activite vlan configuration is tested with aastra.cfg and not with phone specific configuration.
------------------------------------------------------------------------
r13785 | raarts | 2008-03-20 21:12:32 +0100 (Thu, 20 Mar 2008) | 2 lines

Corrected event code.

------------------------------------------------------------------------
r13772 | raarts | 2008-03-17 23:02:16 +0100 (Mon, 17 Mar 2008) | 2 lines

Also add countryprefix and nationalprefix to zap channels.

------------------------------------------------------------------------
r13771 | raarts | 2008-03-17 22:49:39 +0100 (Mon, 17 Mar 2008) | 2 lines

Set some variables we need on incoming Zap channels as well.

------------------------------------------------------------------------
r13769 | raarts | 2008-03-17 14:59:53 +0100 (Mon, 17 Mar 2008) | 2 lines

Don't forget to generate DHCP for company #1. It may have buildings/locations.

------------------------------------------------------------------------
r13760 | raarts | 2008-03-15 14:07:43 +0100 (Sat, 15 Mar 2008) | 3 lines

Incoming SIP calls by domain should go to the company-pre-inbound
context, not the company-inbound one.

------------------------------------------------------------------------
r13757 | raarts | 2008-03-15 12:17:54 +0100 (Sat, 15 Mar 2008) | 2 lines

Fix problems with mixing macros and gosubs.

------------------------------------------------------------------------
r13743 | ivar | 2008-03-13 10:13:36 +0100 (Thu, 13 Mar 2008) | 1 line

Turn DISA into application which can be dialed with *9001
------------------------------------------------------------------------
r13734 | ivar | 2008-03-12 12:44:02 +0100 (Wed, 12 Mar 2008) | 1 line

Also include international version of mobile number in disa dialplan
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r13733 | ivar | 2008-03-12 12:16:30 +0100 (Wed, 12 Mar 2008) | 1 line

Only generate lines in asterisk dialplan for mobile number if disa.disa_cidauth=yes
------------------------------------------------------------------------
r13727 | ivar | 2008-03-12 11:25:06 +0100 (Wed, 12 Mar 2008) | 1 line

Enable disa without pincode when callerid is someone's mobile number
------------------------------------------------------------------------
r13726 | raarts | 2008-03-12 11:17:45 +0100 (Wed, 12 Mar 2008) | 2 lines

Add missing newline after OpFlags for Cisco 7909/12 phones.

------------------------------------------------------------------------
r13686 | ivar | 2008-03-11 10:23:24 +0100 (Tue, 11 Mar 2008) | 1 line

Bugfix: second line of aastra phone diplay is now used for extension. Erase this if users logs out
------------------------------------------------------------------------
r13685 | raarts | 2008-03-10 18:00:45 +0100 (Mon, 10 Mar 2008) | 2 lines

Fix missing label in trunk handling.

------------------------------------------------------------------------
r13683 | ivar | 2008-03-10 17:57:39 +0100 (Mon, 10 Mar 2008) | 1 line

Use ** as directed call pickup instead of *8
------------------------------------------------------------------------
r13679 | ivar | 2008-03-10 15:40:04 +0100 (Mon, 10 Mar 2008) | 1 line

directed call pickup prefix should be *8?
------------------------------------------------------------------------
r13669 | raarts | 2008-03-10 11:29:28 +0100 (Mon, 10 Mar 2008) | 3 lines

Small fixes in forwarding.
Personal IVR menu, only needs prompts now.

------------------------------------------------------------------------
r13666 | raarts | 2008-03-09 13:52:54 +0100 (Sun, 09 Mar 2008) | 4 lines

Okay, hunting seems to work ok now. Both for outgoing and incoming calls.
Note that on incoming calls the recepionist is now always tried if a call
cannot be answered somewhere. 

------------------------------------------------------------------------
r13665 | raarts | 2008-03-09 12:02:28 +0100 (Sun, 09 Mar 2008) | 2 lines

Changes for the new call forwarding in the dialplan.

------------------------------------------------------------------------
r13664 | raarts | 2008-03-09 11:28:42 +0100 (Sun, 09 Mar 2008) | 4 lines

When someone dials a number on the PBX that is not handled by
the dialplan (= is not entered into the incoming number list), now
have a voice say: the number you dialed is not in use).

------------------------------------------------------------------------
r13663 | raarts | 2008-03-09 11:13:03 +0100 (Sun, 09 Mar 2008) | 6 lines

Removed not existing context feature-play-selected-voicemail.
Previously decisions about incoming calls were made both
in inbound.conf and in extensions.conf (in the company-inbound context).
Decided that inbound.conf should only contain Goto statements to
the company-pre-inbound dialplan, and moved it there.

------------------------------------------------------------------------
r13651 | raarts | 2008-03-06 09:11:40 +0100 (Thu, 06 Mar 2008) | 2 lines

Add missing parenthesis.

------------------------------------------------------------------------
r13628 | raarts | 2008-03-04 12:40:24 +0100 (Tue, 04 Mar 2008) | 2 lines

Do vacuum every night. Doesn't do harm.

------------------------------------------------------------------------
r13626 | raarts | 2008-03-04 12:01:56 +0100 (Tue, 04 Mar 2008) | 2 lines

More music on hold fixes.

------------------------------------------------------------------------
r13623 | raarts | 2008-03-04 09:50:19 +0100 (Tue, 04 Mar 2008) | 3 lines

Fixed long-time omission: set company default musiconhold
for outbound calls too.

------------------------------------------------------------------------
r13554 | raarts | 2008-02-27 10:22:44 +0100 (Wed, 27 Feb 2008) | 4 lines

Set proper defaults for incoming sip and IAX trunks for user and
department. On incoming calls that are transfered outwards, we sometimes
need these variables to exist.

------------------------------------------------------------------------
r13551 | ivar | 2008-02-26 16:03:46 +0100 (Tue, 26 Feb 2008) | 1 line

Bugfix: in voicemail menu, when announcing new message, stop saying 'ontvangen om 2 uur uur 36' (2 times 'uur')
------------------------------------------------------------------------
r13546 | ivar | 2008-02-26 12:40:17 +0100 (Tue, 26 Feb 2008) | 1 line

When user changes voicemail password, call script which changes it in database
------------------------------------------------------------------------
r13498 | raarts | 2008-02-22 22:26:42 +0100 (Fri, 22 Feb 2008) | 2 lines

Fix for erroneously generating dupe channels in /etc/asterisk/zapata.conf

------------------------------------------------------------------------
r13497 | raarts | 2008-02-22 15:55:28 +0100 (Fri, 22 Feb 2008) | 3 lines

When all agents in a queue are logged off, or there are no agents,
handle the forwarding for this queue as a 'logged off' situation.

------------------------------------------------------------------------
r13495 | raarts | 2008-02-21 20:57:41 +0100 (Thu, 21 Feb 2008) | 2 lines

When inbound numbers contain wildcards, add the '_' prefix.

------------------------------------------------------------------------
r13494 | raarts | 2008-02-21 20:28:15 +0100 (Thu, 21 Feb 2008) | 2 lines

Fix first contact. netland was hard coded by accident.

------------------------------------------------------------------------
r13468 | raarts | 2008-02-20 12:09:43 +0100 (Wed, 20 Feb 2008) | 24 lines

Support for postponing activation of ISDN card changes until the
user clicks the activate now, or the activate at midnight button.
It works as follows: when any ISDN screen is changed, a record
is written to the activate log ('modify card_te110p 3' for example).
The astcfgd daemon sees these records and sets the astglobal 
'zapchanged' to 'yes', but does nothing else.

GUI screens can test for the zapchanged astglobal, and call
zap_changed_warning(). this displays a text like: 'you have changed
an ISDN card setting, press activate to activate all settings. Note
that all calls will be dropped'

When the user presses the button, the GUI:
1.  sets the astglobal 'zapchanged' to 'activating'. Any further GUI
    refreshes will show: 'ISDN card changes are currently being
    activated by the system'.
2.  adds a new activate record: 'modify zaptel 1'. 

When the astcfgd daemon sees the activate record, it recreates 
/etc/zaptel.conf, /etc/sysconfig/zaptel, and /etc/asterisk/zapata.conf
and restart zaptel and asterisk.
When this is ready, astcfgd will set the astglobal 'zapchanged' to 'no'.
This commit is for astcfgd.

------------------------------------------------------------------------
r13465 | raarts | 2008-02-20 10:01:25 +0100 (Wed, 20 Feb 2008) | 3 lines

Do not create the virtual postfix table any more. This is
outside the Astium responsability.

------------------------------------------------------------------------
r13462 | ivar | 2008-02-20 09:20:27 +0100 (Wed, 20 Feb 2008) | 1 line

Change: on Aastra phones, put name on line 1, extension on line 2 of phone display
------------------------------------------------------------------------
r13459 | raarts | 2008-02-19 19:26:41 +0100 (Tue, 19 Feb 2008) | 3 lines

Reverted earlier changes for ISDN CallerID Presentation.
They were not acceptable after all.

------------------------------------------------------------------------
r13454 | raarts | 2008-02-19 10:47:27 +0100 (Tue, 19 Feb 2008) | 2 lines

The ARG4 parameter of the extension macro is not used any more.

------------------------------------------------------------------------
r13453 | raarts | 2008-02-19 10:06:27 +0100 (Tue, 19 Feb 2008) | 3 lines

Fix blind transfers, try to always set the transfer context to the users' context
(which boils down to something like netland-nl-yyynnnnnn

------------------------------------------------------------------------
r13422 | raarts | 2008-02-17 22:42:33 +0100 (Sun, 17 Feb 2008) | 3 lines

Fixed the dialplan to not use numbered priorities any more
but 'n' and labels instead.

------------------------------------------------------------------------
r13421 | raarts | 2008-02-17 21:52:31 +0100 (Sun, 17 Feb 2008) | 2 lines

Remove setting CDR values, we do not use CDR any more.

------------------------------------------------------------------------
r13420 | raarts | 2008-02-17 21:50:47 +0100 (Sun, 17 Feb 2008) | 2 lines

Removed the SetVar command. Deprecated in 1.2 and removed in 1.4

------------------------------------------------------------------------
r13417 | raarts | 2008-02-17 17:08:41 +0100 (Sun, 17 Feb 2008) | 2 lines

Removed Macro eventlog. Not used any more.

------------------------------------------------------------------------
r13413 | ivar | 2008-02-15 14:23:38 +0100 (Fri, 15 Feb 2008) | 2 lines

Bugfix: when creating, deleting or modifying database users, make sure that we're dealing with lower case usernames.
This fix should prevent astcfgd crashes.
------------------------------------------------------------------------
r13395 | ivar | 2008-02-15 12:00:28 +0100 (Fri, 15 Feb 2008) | 1 line

To be sure, check if file exists before moving it
------------------------------------------------------------------------
r13392 | raarts | 2008-02-15 11:06:38 +0100 (Fri, 15 Feb 2008) | 3 lines

Fix ppp. Make sure 'module all' also reconfigs ppp settings.
Removed ms-ignore-domain option.

------------------------------------------------------------------------
r13389 | raarts | 2008-02-14 23:56:23 +0100 (Thu, 14 Feb 2008) | 2 lines

Also set default dialplans for trunks.

------------------------------------------------------------------------
r13383 | raarts | 2008-02-14 09:57:33 +0100 (Thu, 14 Feb 2008) | 2 lines

Fix missing quote in alarm callerid setting.

------------------------------------------------------------------------
r13382 | raarts | 2008-02-14 00:25:39 +0100 (Thu, 14 Feb 2008) | 2 lines

Support for setting callerid on 112 calls in the dialplan.

------------------------------------------------------------------------
r13362 | raarts | 2008-02-13 00:00:00 +0100 (Wed, 13 Feb 2008) | 2 lines

Support for Primary contacts in the dialplan (through using feature *9000)

------------------------------------------------------------------------
r13359 | raarts | 2008-02-12 19:53:25 +0100 (Tue, 12 Feb 2008) | 2 lines

Support hunt function in the forwarding settings.

------------------------------------------------------------------------
r13358 | raarts | 2008-02-12 17:47:27 +0100 (Tue, 12 Feb 2008) | 2 lines

Support greeting messages on incoming numbers.

------------------------------------------------------------------------
r13341 | ivar | 2008-02-12 09:58:48 +0100 (Tue, 12 Feb 2008) | 2 lines

1. don't generate 'dial plan' configuration for aastra phones,
2. only generate aastra xml configuration files larger than 0 bytes.
------------------------------------------------------------------------
r13331 | raarts | 2008-02-11 12:23:52 +0100 (Mon, 11 Feb 2008) | 2 lines

Fix spelling error. Should be `allowed_passed_screen` and not `allow_passed_screen`.

------------------------------------------------------------------------
r13329 | raarts | 2008-02-10 23:18:41 +0100 (Sun, 10 Feb 2008) | 2 lines

Add contacttype to AstExtInfo call. Only use hunt_number_ as forward number for now.

------------------------------------------------------------------------
r13312 | raarts | 2008-02-08 16:39:28 +0100 (Fri, 08 Feb 2008) | 2 lines

Ensure that even very strict telcos allow our callerid settings.

------------------------------------------------------------------------
r13309 | ivar | 2008-02-08 15:25:39 +0100 (Fri, 08 Feb 2008) | 1 line

New options for aastra 53i keys: save & delete
------------------------------------------------------------------------
r13303 | raarts | 2008-02-07 18:24:12 +0100 (Thu, 07 Feb 2008) | 2 lines

Fix typo in setting callerid.

------------------------------------------------------------------------
r13287 | raarts | 2008-02-06 16:00:06 +0100 (Wed, 06 Feb 2008) | 2 lines

Fix voicemail bug.

------------------------------------------------------------------------
r13260 | raarts | 2008-02-04 14:18:29 +0100 (Mon, 04 Feb 2008) | 5 lines

When a line is busy, don't Answer() but only do Busy().
This way the busy is passed through the calling channel
using out-of-band signalling, and that's the way it should
be done.

------------------------------------------------------------------------
r13203 | raarts | 2008-01-29 22:38:24 +0100 (Tue, 29 Jan 2008) | 2 lines

Make setting the external callerid more robust.

------------------------------------------------------------------------
r13199 | raarts | 2008-01-28 22:48:25 +0100 (Mon, 28 Jan 2008) | 3 lines

Honour the callee_xfer flags in the callrouting table in the
generating outbound Dial command.

------------------------------------------------------------------------
r13194 | ivar | 2008-01-28 11:48:04 +0100 (Mon, 28 Jan 2008) | 1 line

Bugfix: in Thomson config file, don't give same codec more than 1 priority
------------------------------------------------------------------------
r13192 | raarts | 2008-01-28 11:42:59 +0100 (Mon, 28 Jan 2008) | 2 lines

Fix dialplan syntax error.

------------------------------------------------------------------------
r13143 | ivar | 2008-01-22 14:57:00 +0100 (Tue, 22 Jan 2008) | 1 line

Allow for entering 10 digits for login/pin when switching user with *1*
------------------------------------------------------------------------
r13120 | raarts | 2008-01-21 14:40:56 +0100 (Mon, 21 Jan 2008) | 2 lines

Fix FTP URL for Aastra newer phones.

------------------------------------------------------------------------
r13114 | raarts | 2008-01-20 23:27:22 +0100 (Sun, 20 Jan 2008) | 3 lines

Don't wait 1 second when going to musiconhold. Takes too
long when the OSB initiates it.

------------------------------------------------------------------------
r13113 | raarts | 2008-01-20 13:45:51 +0100 (Sun, 20 Jan 2008) | 3 lines

When a trunk was chosen by extension, and it was unavailable, it would
not bounce to the next extension. Fixed.

------------------------------------------------------------------------
r13109 | raarts | 2008-01-18 22:18:18 +0100 (Fri, 18 Jan 2008) | 3 lines

Ensure that incoming calls are send to the proper diaplan context
when they are being forwarded.

------------------------------------------------------------------------
r13106 | raarts | 2008-01-18 18:13:42 +0100 (Fri, 18 Jan 2008) | 2 lines

Use bash as the shell

------------------------------------------------------------------------
r13103 | raarts | 2008-01-18 16:57:05 +0100 (Fri, 18 Jan 2008) | 2 lines

Stil must use revoke, because of users from pre-1.6

------------------------------------------------------------------------
r13102 | raarts | 2008-01-18 16:29:09 +0100 (Fri, 18 Jan 2008) | 2 lines

Send DDI as Dial in USERDATA Event.

------------------------------------------------------------------------
r13084 | raarts | 2008-01-17 18:12:17 +0100 (Thu, 17 Jan 2008) | 4 lines

when callers can't enter a queue, or are dropped because no
agents can be reached, go to the Forward-when-logged-off rule
of the queue extension.

------------------------------------------------------------------------
r13081 | raarts | 2008-01-17 17:28:05 +0100 (Thu, 17 Jan 2008) | 5 lines

Uneachable queue agents are now seen as 'logged out'. This ensures that
when a DSL is down, new callers are not entered into the queue, and forwarded
to the next extension. Also, when all agents become unreachable, all
callers are dropped out of the queue.

------------------------------------------------------------------------
r13080 | raarts | 2008-01-17 17:25:59 +0100 (Thu, 17 Jan 2008) | 2 lines

Generate national dialplans correctly

------------------------------------------------------------------------
r13058 | raarts | 2008-01-16 09:30:23 +0100 (Wed, 16 Jan 2008) | 2 lines

Generate The proper national dialplan for country-specific contexts.

------------------------------------------------------------------------
r13054 | raarts | 2008-01-15 17:15:04 +0100 (Tue, 15 Jan 2008) | 2 lines

Oeps, also add new astiumd.php script to the Makefile.am

------------------------------------------------------------------------
r13053 | raarts | 2008-01-15 16:10:25 +0100 (Tue, 15 Jan 2008) | 2 lines

Fixes in generatingen /etc/astiumd.conf.

------------------------------------------------------------------------
r13052 | raarts | 2008-01-15 16:03:06 +0100 (Tue, 15 Jan 2008) | 2 lines

Generate dialplans for all countries we're sitting in.

------------------------------------------------------------------------
r13050 | raarts | 2008-01-15 15:03:17 +0100 (Tue, 15 Jan 2008) | 2 lines

Fix Aastra vendorspecific DHCP options.

------------------------------------------------------------------------
r13047 | raarts | 2008-01-15 12:50:38 +0100 (Tue, 15 Jan 2008) | 3 lines

Don't create files for astmanproxy any more. Never used it anyway
Instead create config file for astiumd.

------------------------------------------------------------------------
r13039 | raarts | 2008-01-14 17:46:06 +0100 (Mon, 14 Jan 2008) | 3 lines

Okay, also put 0031 in front of mobile numbers and 0800/0900 numbers.
Leave international numbers alone.

------------------------------------------------------------------------
r13038 | raarts | 2008-01-14 17:32:17 +0100 (Mon, 14 Jan 2008) | 7 lines

Did something I have been planning to do for a long time:
- always send internationalized phone numbers.
- use different national dialplans for phones in different countries
- do not give each user its own dialplan, but differentiate on permissions.
  this greatly reduces the number of dialplans when you have lots of users,
  and also speeds up astcfgd processing a real lot.

------------------------------------------------------------------------
r12947 | raarts | 2008-01-07 12:26:17 +0100 (Mon, 07 Jan 2008) | 2 lines

You should quote usernames when adding users to postgresql.

------------------------------------------------------------------------
r12930 | raarts | 2008-01-03 14:29:08 +0100 (Thu, 03 Jan 2008) | 2 lines

Support CustomerID. Tell astiumd the id of the phoneline.

------------------------------------------------------------------------
r12929 | ivar | 2008-01-03 12:12:49 +0100 (Thu, 03 Jan 2008) | 1 line

(NeoNova 67330) Bugfix: when logging out phones at night, don't log out iax2 devices such as software faxes
------------------------------------------------------------------------
r12925 | ivar | 2008-01-03 11:25:17 +0100 (Thu, 03 Jan 2008) | 1 line

Bugfix: delete 'systemactions' array after actions have been done, otherwise at some point all phones will reboot if one person logs in or out
------------------------------------------------------------------------
r12922 | perry | 2008-01-03 10:32:31 +0100 (Thu, 03 Jan 2008) | 1 line

created function remove_pid_file
------------------------------------------------------------------------
r12905 | ivar | 2007-12-28 16:05:58 +0100 (Fri, 28 Dec 2007) | 1 line

direct call pickup is ** instead of *79
------------------------------------------------------------------------
r12902 | ivar | 2007-12-28 14:45:51 +0100 (Fri, 28 Dec 2007) | 1 line

enabled directed call pickup for aastra phones
------------------------------------------------------------------------
r12901 | nnrelman | 2007-12-28 14:01:58 +0100 (Fri, 28 Dec 2007) | 1 line

astcfgd requires now curl
------------------------------------------------------------------------
r12900 | ivar | 2007-12-28 13:36:05 +0100 (Fri, 28 Dec 2007) | 1 line

Updates to astcfgd, enhancing Aastra login/logout
------------------------------------------------------------------------
r12896 | ivar | 2007-12-27 14:09:25 +0100 (Thu, 27 Dec 2007) | 2 lines

Update to check-sync.pl:
push all xml files <mac>*.xml to the phone.
------------------------------------------------------------------------
r12895 | ivar | 2007-12-27 14:06:15 +0100 (Thu, 27 Dec 2007) | 2 lines

Update to tftp.php; generating aastra config files.
Cleaned up and make sure xml files are no more than 10000 bytes
------------------------------------------------------------------------
r12893 | raarts | 2007-12-27 10:48:18 +0100 (Thu, 27 Dec 2007) | 2 lines

Fold two UserEvents into one.

------------------------------------------------------------------------
r12892 | raarts | 2007-12-27 10:42:45 +0100 (Thu, 27 Dec 2007) | 2 lines

Give more info to astiumd when dialing an extension

------------------------------------------------------------------------
r12890 | ivar | 2007-12-27 09:56:58 +0100 (Thu, 27 Dec 2007) | 3 lines

Update to check-sync for Aastra phones.
Post configuration in xml format to phone's web server.
Use -b option to perform wget in background, so we don't have to wait 2 seconds after each wget command
------------------------------------------------------------------------
r12884 | ivar | 2007-12-24 12:01:08 +0100 (Mon, 24 Dec 2007) | 1 line

Update to astcfgd: create several xml files for aastra phones, because their size is limited to 10000 bytes. mac-expmods.xml will still exceed that amount
------------------------------------------------------------------------
r12881 | ivar | 2007-12-24 10:23:50 +0100 (Mon, 24 Dec 2007) | 1 line

Update to astcfgd: change the order in which modules' activate function is called
------------------------------------------------------------------------
r12879 | ivar | 2007-12-24 10:07:37 +0100 (Mon, 24 Dec 2007) | 1 line

Change in astcfgd: reboot / reset phones in activation stage at the end of processing, not in the action stage
------------------------------------------------------------------------
r12872 | raarts | 2007-12-21 16:36:45 +0100 (Fri, 21 Dec 2007) | 2 lines

Some small bugs are very hard to find. This time in cisco provisioning. Fixed.

------------------------------------------------------------------------
r12866 | raarts | 2007-12-21 15:05:29 +0100 (Fri, 21 Dec 2007) | 2 lines

Replace '+' on incoming SIP calls from SIP trunk.

------------------------------------------------------------------------
r12832 | ivar | 2007-12-19 15:04:23 +0100 (Wed, 19 Dec 2007) | 1 line

more changes to facilitate aastra phones login and logout without reboot
------------------------------------------------------------------------
r12807 | raarts | 2007-12-18 16:57:18 +0100 (Tue, 18 Dec 2007) | 2 lines

Change a UserEvent.

------------------------------------------------------------------------
r12804 | ivar | 2007-12-18 16:00:19 +0100 (Tue, 18 Dec 2007) | 1 line

Update: while generating configuration file for Aastra phone, also generate same info in xml format
------------------------------------------------------------------------
r12777 | raarts | 2007-12-14 22:53:06 +0100 (Fri, 14 Dec 2007) | 2 lines

Support using RDNIS as DDI (for 088 support for example)

------------------------------------------------------------------------
r12775 | raarts | 2007-12-14 17:22:07 +0100 (Fri, 14 Dec 2007) | 2 lines

Support phones behind NAT when asterisk itself is behind NAT.

------------------------------------------------------------------------
r12664 | raarts | 2007-12-05 15:52:53 +0100 (Wed, 05 Dec 2007) | 2 lines

Fixed call parking

------------------------------------------------------------------------
r12657 | raarts | 2007-12-05 12:43:03 +0100 (Wed, 05 Dec 2007) | 2 lines

Fix bug in string comparison

------------------------------------------------------------------------
r12646 | raarts | 2007-12-04 12:10:37 +0100 (Tue, 04 Dec 2007) | 2 lines

Add trunk DTMF to 1.6 astcfgd.

------------------------------------------------------------------------
r12636 | raarts | 2007-12-04 10:32:10 +0100 (Tue, 04 Dec 2007) | 3 lines

If a caller re-enters the same queu, don't prefix the
callerid again with the name of the queue

------------------------------------------------------------------------
r12635 | raarts | 2007-12-04 10:27:47 +0100 (Tue, 04 Dec 2007) | 3 lines

increment call-forward loopcounter from 5 to 100, Ask Joop Perlee
for re-enteringqueues from an IVR lots of times.

------------------------------------------------------------------------
r12608 | raarts | 2007-11-28 11:27:42 +0100 (Wed, 28 Nov 2007) | 3 lines

Eerste versie serieschakeling havenbedrijf.
Set callerpres to more acceptable values.

------------------------------------------------------------------------
r12593 | raarts | 2007-11-22 21:53:59 +0100 (Thu, 22 Nov 2007) | 2 lines

Remove Superfluous UserEvent.

------------------------------------------------------------------------
r12592 | ivar | 2007-11-22 10:59:55 +0100 (Thu, 22 Nov 2007) | 1 line

Implemented: 'last call return' function key for aastra
------------------------------------------------------------------------
r12589 | raarts | 2007-11-21 14:08:26 +0100 (Wed, 21 Nov 2007) | 2 lines

Fix to ensure that all companies are process in one go by astcfgd.

------------------------------------------------------------------------
r12585 | ivar | 2007-11-21 11:09:55 +0100 (Wed, 21 Nov 2007) | 1 line

Bugfix: Aastra 5xi phones require 'conf' as key type, instead of 'conference'
------------------------------------------------------------------------
r12584 | ivar | 2007-11-21 11:07:15 +0100 (Wed, 21 Nov 2007) | 1 line

Bugfix: Aastra 5xi phones require 'xfer' as key type, instead of 'transfer'
------------------------------------------------------------------------
r12582 | raarts | 2007-11-20 22:46:00 +0100 (Tue, 20 Nov 2007) | 2 lines

Fix timezone in voicemail.

------------------------------------------------------------------------
r12547 | raarts | 2007-11-17 00:23:32 +0100 (Sat, 17 Nov 2007) | 2 lines

Forgot a variable in the previous commit

------------------------------------------------------------------------
r12546 | raarts | 2007-11-16 23:59:00 +0100 (Fri, 16 Nov 2007) | 5 lines

Improved the chance that an incoming SIP call is actually 
processed. Sometimes incoming calls from fixed hosts are matched
on the [sip-out-n] entry, becuase it contains the same host.
In that case the entry needs to have a proper context, and variables.

------------------------------------------------------------------------
r12544 | raarts | 2007-11-16 22:50:08 +0100 (Fri, 16 Nov 2007) | 2 lines

This is dumb. SIP does not have auth=plaintext/md5/rsa. So removed it.

------------------------------------------------------------------------
r12543 | raarts | 2007-11-16 22:44:02 +0100 (Fri, 16 Nov 2007) | 3 lines

The 'port' command for SIP channels in the general section of
sip.conf was renamed to 'bindport' 

------------------------------------------------------------------------
r12540 | raarts | 2007-11-15 15:19:46 +0100 (Thu, 15 Nov 2007) | 2 lines

Call Waiting tone for Polycom and Aastra.

------------------------------------------------------------------------
r12528 | raarts | 2007-11-14 15:38:14 +0100 (Wed, 14 Nov 2007) | 2 lines

Let all polycom phone models download their own (short) firmware file.

------------------------------------------------------------------------
r12522 | raarts | 2007-11-14 10:42:29 +0100 (Wed, 14 Nov 2007) | 2 lines

Fix phonebuttons for Polycom 330/430/550/650

------------------------------------------------------------------------
r12505 | raarts | 2007-11-13 20:13:09 +0100 (Tue, 13 Nov 2007) | 3 lines

Tell end-of-life Polycom models (300/500) to always
use firmware version 2.1.2.

------------------------------------------------------------------------
r12504 | raarts | 2007-11-13 20:11:58 +0100 (Tue, 13 Nov 2007) | 2 lines

Loop if user enters wrong credentials on *1*

------------------------------------------------------------------------
r12489 | ivar | 2007-11-12 15:40:28 +0100 (Mon, 12 Nov 2007) | 1 line

Bugfix: add variable to prevent 'CSeq Number Out of order' error
------------------------------------------------------------------------
r12473 | ivar | 2007-11-09 10:00:44 +0100 (Fri, 09 Nov 2007) | 1 line

Bugfix: also generate nat variable for incoming sip trunk in sip.conf
------------------------------------------------------------------------
r12456 | raarts | 2007-11-07 10:31:17 +0100 (Wed, 07 Nov 2007) | 2 lines

Dialplan support for default contacts in the fastdial_company table.

------------------------------------------------------------------------
r12443 | ivar | 2007-11-06 10:57:50 +0100 (Tue, 06 Nov 2007) | 1 line

Bugfix: implement transfer and conference for function keys on aastra phones
------------------------------------------------------------------------
r12442 | raarts | 2007-11-06 10:38:04 +0100 (Tue, 06 Nov 2007) | 2 lines

Fix for Aastra lights not working.

------------------------------------------------------------------------
r12440 | raarts | 2007-11-05 17:56:44 +0100 (Mon, 05 Nov 2007) | 2 lines

When in_register = 'yes' adjust accrordingly.

------------------------------------------------------------------------
r12436 | ivar | 2007-11-05 13:16:50 +0100 (Mon, 05 Nov 2007) | 1 line

use new database field trunk.nat when generation sip.conf
------------------------------------------------------------------------
r12428 | raarts | 2007-11-04 22:23:33 +0100 (Sun, 04 Nov 2007) | 3 lines

Don't generate queue events from the dialplan any more, as
they are now generated from the Queue app itself.

------------------------------------------------------------------------
r12420 | raarts | 2007-11-04 16:46:31 +0100 (Sun, 04 Nov 2007) | 2 lines

Support for G.729 transcoder card.

------------------------------------------------------------------------
r12417 | raarts | 2007-11-04 16:31:13 +0100 (Sun, 04 Nov 2007) | 5 lines

Removed most code for asterisk's CDR handling, since we will be
generating our own CDR's from now on, since this is so much better.
Added UserEvents where necessary to let the dialplan send the
necessary info to astiumd.

------------------------------------------------------------------------
r12397 | raarts | 2007-11-01 12:38:13 +0100 (Thu, 01 Nov 2007) | 2 lines

Queue names changed, adapt.

------------------------------------------------------------------------
r12393 | raarts | 2007-11-01 09:26:30 +0100 (Thu, 01 Nov 2007) | 2 lines

Support call pickup on Aastra phones.

------------------------------------------------------------------------
r12391 | raarts | 2007-10-31 22:25:05 +0100 (Wed, 31 Oct 2007) | 2 lines

Add the CFWD UserEvents to the dialplan, so astiumd can log them into the callhist table.

------------------------------------------------------------------------
r12360 | raarts | 2007-10-29 08:31:16 +0100 (Mon, 29 Oct 2007) | 2 lines

Limit faxes to our software faes to max 4 hours/fax.

------------------------------------------------------------------------
r12359 | raarts | 2007-10-29 08:28:14 +0100 (Mon, 29 Oct 2007) | 3 lines

Use the new database queries for setting call forwarding from
the dialplan. The queries also log to the history table.

------------------------------------------------------------------------
r12357 | raarts | 2007-10-28 21:26:33 +0100 (Sun, 28 Oct 2007) | 6 lines

When dialing out on ISDN, start at the highest channel number and work
down instead of the other way around. If you start looking at channel
1 for a free channel, the chances for glare to happen are much smaller.
(glare is dialing out on a channel just when a call comes in. This
situation may lead to trouble sometimes in the chan_zap driver).

------------------------------------------------------------------------
r12349 | raarts | 2007-10-28 09:38:57 +0100 (Sun, 28 Oct 2007) | 6 lines

Generating the cdr by asterisk has been dropped, from now on
astiumd will take over that task. Astiumd will log to a new
table: astiumd_cdr, and will also handle billing. As a result
billing is not handled any more by astcfgd (using database
functions). 

------------------------------------------------------------------------
r12338 | raarts | 2007-10-26 23:41:15 +0200 (Fri, 26 Oct 2007) | 2 lines

Oops, I meant the queue id, not the company id.

------------------------------------------------------------------------
r12337 | raarts | 2007-10-26 23:40:06 +0200 (Fri, 26 Oct 2007) | 3 lines

Ensure that the internal name of a queue starts with the numberic queue id.
This way it is always unique AND enables astiumd to decode the id from the name.

------------------------------------------------------------------------
r12336 | raarts | 2007-10-26 17:26:53 +0200 (Fri, 26 Oct 2007) | 2 lines

When rebooting Aastra phones, only try that once.

------------------------------------------------------------------------
r12222 | raarts | 2007-10-14 16:57:34 +0200 (Sun, 14 Oct 2007) | 2 lines

Merged all changes in 1.4 into 1.6

------------------------------------------------------------------------
r10821 | perry | 2007-04-19 16:39:58 +0200 (Thu, 19 Apr 2007) | 1 line

increased version number from 1.4 to 1.6 : astcfgd
------------------------------------------------------------------------
r10788 | perry | 2007-04-19 12:38:20 +0200 (Thu, 19 Apr 2007) | 1 line

Creating branche astcfgd version 1.6
------------------------------------------------------------------------
r10780 | raarts | 2007-04-19 12:14:42 +0200 (Thu, 19 Apr 2007) | 2 lines

Added scripts to copy an Astium database to a slave machine.

------------------------------------------------------------------------
r10766 | raarts | 2007-04-18 16:37:29 +0200 (Wed, 18 Apr 2007) | 3 lines

Fix special characters in the festival application.
Fix missing aastra phonetypes.

------------------------------------------------------------------------
r10762 | raarts | 2007-04-18 14:18:05 +0200 (Wed, 18 Apr 2007) | 2 lines

Support for Aastra 53i, 55i, 57i in the backend end dialplan.

------------------------------------------------------------------------
r10730 | raarts | 2007-04-13 17:23:05 +0200 (Fri, 13 Apr 2007) | 3 lines

Add defaultip=x.x.x.x so the lights on (SNOM and other) phones 
will not be lit when the phone is totally off-line or unreachable.

------------------------------------------------------------------------
r10725 | raarts | 2007-04-13 12:16:03 +0200 (Fri, 13 Apr 2007) | 2 lines

For grandstream phones: P52=0 means _enable_ NAT, non-intuitively..

------------------------------------------------------------------------
r10724 | ivar | 2007-04-12 15:54:28 +0200 (Thu, 12 Apr 2007) | 1 line

(NeoNova 30668) Bugfix: be more flexible when creating conferences in extensions.conf
------------------------------------------------------------------------
r10720 | ivar | 2007-04-11 13:21:04 +0200 (Wed, 11 Apr 2007) | 1 line

Bugfix: only generate voicemail box for people
------------------------------------------------------------------------
r10715 | raarts | 2007-04-11 11:52:21 +0200 (Wed, 11 Apr 2007) | 2 lines

Do not enable reinvite on external SIP trunks.

------------------------------------------------------------------------
r10712 | raarts | 2007-04-10 16:05:04 +0200 (Tue, 10 Apr 2007) | 3 lines

Only allow '#' on analog phones and outgoing lines to prevent ALL
calls goign through the Astium.

------------------------------------------------------------------------
r10709 | raarts | 2007-04-09 13:39:08 +0200 (Mon, 09 Apr 2007) | 4 lines

Always direct incoming sip/iax trunks to the same location
as the PRI trunks, or various things will go wrong (such as
CDR records)

------------------------------------------------------------------------
r10708 | raarts | 2007-04-09 00:27:50 +0200 (Mon, 09 Apr 2007) | 2 lines

Disable echocancellation for incoming software fax calls.

------------------------------------------------------------------------
r10704 | raarts | 2007-04-06 17:16:51 +0200 (Fri, 06 Apr 2007) | 2 lines

Fix Polycom time.

------------------------------------------------------------------------
r10699 | ivar | 2007-04-06 14:26:33 +0200 (Fri, 06 Apr 2007) | 1 line

Bugfix: put personal phonebook on polycom phone
------------------------------------------------------------------------
r10692 | raarts | 2007-04-05 17:48:06 +0200 (Thu, 05 Apr 2007) | 2 lines

Support for Elmeg IP-290 in the backend.

------------------------------------------------------------------------
r10685 | raarts | 2007-04-05 16:50:33 +0200 (Thu, 05 Apr 2007) | 2 lines

Sometimes asterisk -rx outputs binary characters. Make grep ignore this fact.

------------------------------------------------------------------------
r10682 | raarts | 2007-04-04 22:49:34 +0200 (Wed, 04 Apr 2007) | 2 lines

Add support for new Polycom phones in the backend.

------------------------------------------------------------------------
r10676 | raarts | 2007-04-04 20:51:14 +0200 (Wed, 04 Apr 2007) | 4 lines

Do not remove people from the queue that are logged when honor forwarding
is enabled for those people. In that case they should still be part
of the queue.

------------------------------------------------------------------------
r10663 | raarts | 2007-04-03 23:15:08 +0200 (Tue, 03 Apr 2007) | 2 lines

Remove calling getringtonename() from the dialplan. It does not exist any more.

------------------------------------------------------------------------
r10662 | raarts | 2007-04-03 22:27:06 +0200 (Tue, 03 Apr 2007) | 3 lines

Ensure that Busy tones will not last more than 15 seconds.
After that the channel will be hung up.

------------------------------------------------------------------------
r10640 | raarts | 2007-04-01 00:28:34 +0200 (Sun, 01 Apr 2007) | 2 lines

GSM card support in the backend.

------------------------------------------------------------------------
r10623 | raarts | 2007-03-29 23:26:07 +0200 (Thu, 29 Mar 2007) | 2 lines

Update the ddi value in the cdr.

------------------------------------------------------------------------
r10579 | raarts | 2007-03-27 16:52:44 +0200 (Tue, 27 Mar 2007) | 2 lines

Use STUN server as provided from astglobals (System -> Advanced)

------------------------------------------------------------------------
r10561 | raarts | 2007-03-26 12:13:19 +0200 (Mon, 26 Mar 2007) | 2 lines

Only play congestion tones for 5 seconds, then hangup.:

------------------------------------------------------------------------
r10554 | raarts | 2007-03-24 22:24:15 +0100 (Sat, 24 Mar 2007) | 2 lines

Compensate for cases where we receive normalizeed callerid.

------------------------------------------------------------------------
r10530 | raarts | 2007-03-21 14:57:29 +0100 (Wed, 21 Mar 2007) | 2 lines

Allow the '#' to also work while in Queue calls.

------------------------------------------------------------------------
r10461 | raarts | 2007-03-16 13:59:14 +0100 (Fri, 16 Mar 2007) | 3 lines

Fix in determining hangup status.
Support for SNOM ringtones.

------------------------------------------------------------------------
r10426 | raarts | 2007-03-11 14:58:06 +0100 (Sun, 11 Mar 2007) | 2 lines

Do NOT use the h extension, or the destination will be 'h' in the CDR!

------------------------------------------------------------------------
r10401 | ivar | 2007-03-01 16:43:11 +0100 (Thu, 01 Mar 2007) | 1 line

Bugfix: prevent Astium from accepting mail itself
------------------------------------------------------------------------
r10392 | raarts | 2007-02-28 15:40:17 +0100 (Wed, 28 Feb 2007) | 2 lines

Fix error in voicemail line, forgot pager email address field.

------------------------------------------------------------------------
r10391 | raarts | 2007-02-28 15:38:23 +0100 (Wed, 28 Feb 2007) | 3 lines

Fix errors in postfix CIDR address. Also set myhostname from 
the hosstname, not from the domain

------------------------------------------------------------------------
r10366 | raarts | 2007-02-26 15:49:08 +0100 (Mon, 26 Feb 2007) | 2 lines

Restart iaxmodem when a faxmodem is added/deleted.

------------------------------------------------------------------------
r10361 | raarts | 2007-02-26 15:16:48 +0100 (Mon, 26 Feb 2007) | 3 lines

Also show the name of the inbound line if we also recognized.
have the name of the inbound caller...

------------------------------------------------------------------------
r10358 | raarts | 2007-02-26 15:05:47 +0100 (Mon, 26 Feb 2007) | 3 lines

When Name On Display == 'yes' in the inbound dial screen, display
the name of the inbound line on the phone display.

------------------------------------------------------------------------
r10334 | raarts | 2007-02-23 17:12:39 +0100 (Fri, 23 Feb 2007) | 2 lines

Also use only the firstname for the fax's tagline.

------------------------------------------------------------------------
r10333 | raarts | 2007-02-23 17:11:41 +0100 (Fri, 23 Feb 2007) | 2 lines

Use only the contact's first name for the cidname.

------------------------------------------------------------------------
r10273 | raarts | 2007-02-18 22:15:12 +0100 (Sun, 18 Feb 2007) | 2 lines

When retrieving userid also get the departmentid.

------------------------------------------------------------------------
r10270 | raarts | 2007-02-18 20:42:09 +0100 (Sun, 18 Feb 2007) | 5 lines

Don't ltrim phonenumbers any more. 
WARNING WARNING WARNING WARNING WARNING!!!!!!!!!!

This upgrade may kill the inbound diaplan!!! Check it.

------------------------------------------------------------------------
r10264 | raarts | 2007-02-18 16:31:47 +0100 (Sun, 18 Feb 2007) | 3 lines

Rely on the contact table for pptp users, and not on the
obsolete table pptpusers...

------------------------------------------------------------------------
r10251 | raarts | 2007-02-17 13:48:29 +0100 (Sat, 17 Feb 2007) | 2 lines

Fix PHP syntax error.

------------------------------------------------------------------------
r10227 | raarts | 2007-02-16 10:08:29 +0100 (Fri, 16 Feb 2007) | 2 lines

The 9991 extension now also pronounces the IP address of the phone.

------------------------------------------------------------------------
r10219 | raarts | 2007-02-15 11:17:11 +0100 (Thu, 15 Feb 2007) | 2 lines

Couldn't page XTEN-Lite. Also two NoCDR in a row is too much, so I removed one.

------------------------------------------------------------------------
r10218 | raarts | 2007-02-15 11:15:16 +0100 (Thu, 15 Feb 2007) | 2 lines

When paging, you don't want a CDR record for each paged phone, do you?

------------------------------------------------------------------------
r10201 | raarts | 2007-02-14 12:22:28 +0100 (Wed, 14 Feb 2007) | 2 lines

VLAN support for phone provisioning: Grandstream, Thomson, Aastra, Cisco790x, and Polycom.

------------------------------------------------------------------------
r10186 | raarts | 2007-02-13 23:30:53 +0100 (Tue, 13 Feb 2007) | 2 lines

Set CALL facility when dialing out, so the billing can bill this.

------------------------------------------------------------------------
r10174 | ivar | 2007-02-12 16:12:22 +0100 (Mon, 12 Feb 2007) | 1 line

Bugfix: ,s needed to be removed
------------------------------------------------------------------------
r10171 | ivar | 2007-02-12 15:19:05 +0100 (Mon, 12 Feb 2007) | 1 line

Bugfix: remove something that doesn't work
------------------------------------------------------------------------
r10162 | raarts | 2007-02-12 00:47:57 +0100 (Mon, 12 Feb 2007) | 2 lines

Now write used facilities into the new facilities CDR column.

------------------------------------------------------------------------
r10142 | raarts | 2007-02-08 21:53:13 +0100 (Thu, 08 Feb 2007) | 2 lines

Fix error in strftime specification.

------------------------------------------------------------------------
r10138 | raarts | 2007-02-08 11:40:30 +0100 (Thu, 08 Feb 2007) | 2 lines

Only put persons in a queue if they are logged in.

------------------------------------------------------------------------
r10137 | raarts | 2007-02-07 23:20:33 +0100 (Wed, 07 Feb 2007) | 2 lines

Now it's basically possible to set a maximum cost per minute for a call.

------------------------------------------------------------------------
r10119 | raarts | 2007-02-06 11:09:45 +0100 (Tue, 06 Feb 2007) | 3 lines

Removed setting plugin parameter values in dialplan.
Is now done in asterisk itself (app_pgdb.c)

------------------------------------------------------------------------
r10115 | raarts | 2007-02-05 22:51:05 +0100 (Mon, 05 Feb 2007) | 2 lines

Process the plugin parameters into Set() commands.

------------------------------------------------------------------------
r10104 | raarts | 2007-02-04 21:50:20 +0100 (Sun, 04 Feb 2007) | 3 lines

Put the dialed number in the recording filename on outgoing calls,
and the callerid on incoming calls.

------------------------------------------------------------------------
r10103 | raarts | 2007-02-04 20:46:59 +0100 (Sun, 04 Feb 2007) | 2 lines

Fixed missing '%'.

------------------------------------------------------------------------
r10102 | raarts | 2007-02-04 18:46:30 +0100 (Sun, 04 Feb 2007) | 4 lines

WAV filenames currently contain the ':' character. This is not appreciated
by Windows(tm). I changed it into YYYYMMDD-HHMMSS. This is also better
for the sort order in directories.

------------------------------------------------------------------------
r10099 | raarts | 2007-02-04 16:58:33 +0100 (Sun, 04 Feb 2007) | 3 lines

Add README or plugins.
Ensure the pugin directory exists..

------------------------------------------------------------------------
r10098 | raarts | 2007-02-02 23:33:00 +0100 (Fri, 02 Feb 2007) | 4 lines

The dialplan variables for user, department and company should
be defined as inheritable (by prefixing them with two underscores).
Same goes for DID. 

------------------------------------------------------------------------
r10097 | raarts | 2007-02-02 23:31:09 +0100 (Fri, 02 Feb 2007) | 3 lines

Identify a user when he calls in from one of his registered
caller id number (mobile, homephone, and altphone columns in contact table).

------------------------------------------------------------------------
r10094 | raarts | 2007-02-02 22:29:09 +0100 (Fri, 02 Feb 2007) | 5 lines

Normalize DID to national numbers. Sometimes a call comes in
on a trunk and it is prefixed with the country prefix.
Normalize these numbers to national numbers, so the number will 
match the one that is entered in the inbound numbers screen.

------------------------------------------------------------------------
r10089 | raarts | 2007-02-02 12:27:16 +0100 (Fri, 02 Feb 2007) | 2 lines

Could not connect incoming number to *31 and some other extensions. Fixed.

------------------------------------------------------------------------
r10075 | raarts | 2007-02-02 09:00:42 +0100 (Fri, 02 Feb 2007) | 2 lines

Watch maximum call length when prepay is enabled.

------------------------------------------------------------------------
r10074 | raarts | 2007-02-02 08:54:57 +0100 (Fri, 02 Feb 2007) | 3 lines

Use the prepay field in the backend. If set to yes, deduct
call costs from individuals' account.

------------------------------------------------------------------------
r10068 | raarts | 2007-02-02 01:26:07 +0100 (Fri, 02 Feb 2007) | 2 lines

Not only support billing, but now supports prepay as well..

------------------------------------------------------------------------
r10063 | raarts | 2007-02-01 20:18:24 +0100 (Thu, 01 Feb 2007) | 2 lines

Remove unnecessary label.

------------------------------------------------------------------------
r10062 | raarts | 2007-02-01 16:48:09 +0100 (Thu, 01 Feb 2007) | 2 lines

Clear voicemail and CallerID Name field for Grandstream when nobody is logged in.

------------------------------------------------------------------------
r10058 | ivar | 2007-02-01 15:24:32 +0100 (Thu, 01 Feb 2007) | 1 line

Change: take area code from building where company is
------------------------------------------------------------------------
r10044 | raarts | 2007-01-31 21:35:52 +0100 (Wed, 31 Jan 2007) | 2 lines

... and to the specfile..

------------------------------------------------------------------------
r10043 | raarts | 2007-01-31 21:35:05 +0100 (Wed, 31 Jan 2007) | 2 lines

Add the new astium.daily script to Makefile.am as well.

------------------------------------------------------------------------
r10042 | raarts | 2007-01-31 21:34:39 +0100 (Wed, 31 Jan 2007) | 4 lines

Reset queue statistics every night.
Replace several softlinks in /etc/cron.daily
with a link to astium.daily, which does all the work.

------------------------------------------------------------------------
r10032 | raarts | 2007-01-31 14:47:08 +0100 (Wed, 31 Jan 2007) | 2 lines

Fix setting relayhost from astglobals.

------------------------------------------------------------------------
r10031 | raarts | 2007-01-31 14:44:13 +0100 (Wed, 31 Jan 2007) | 2 lines

Changed refernce to trunk_zap into trunk, type = 'zap'

------------------------------------------------------------------------
r10028 | ivar | 2007-01-31 14:32:55 +0100 (Wed, 31 Jan 2007) | 1 line

Bugfix: taking company.areacode gives sql error and crashes astcfgd
------------------------------------------------------------------------
r10023 | ivar | 2007-01-31 13:47:39 +0100 (Wed, 31 Jan 2007) | 2 lines

Bugfix: don't get areacode because it's removed from company table.
There's another mention of getting company.areacode below but I'm not sure if I can remove that
------------------------------------------------------------------------
r10017 | raarts | 2007-01-31 09:47:02 +0100 (Wed, 31 Jan 2007) | 2 lines

You can now set the maximum number of concurrent incoming calls on a trunk.

------------------------------------------------------------------------
r10016 | raarts | 2007-01-31 09:27:39 +0100 (Wed, 31 Jan 2007) | 2 lines

Fix bug in forwarding in dialplan.

------------------------------------------------------------------------
r10015 | raarts | 2007-01-30 23:15:49 +0100 (Tue, 30 Jan 2007) | 2 lines

Set deparrtmentid to reasonable default

------------------------------------------------------------------------
r10013 | raarts | 2007-01-30 22:35:28 +0100 (Tue, 30 Jan 2007) | 2 lines

Fix small bug in 112 processing.

------------------------------------------------------------------------
r10012 | raarts | 2007-01-30 22:28:37 +0100 (Tue, 30 Jan 2007) | 2 lines

Fix blatant bug in logging statement

------------------------------------------------------------------------
r10005 | raarts | 2007-01-29 19:29:02 +0100 (Mon, 29 Jan 2007) | 2 lines

Make internal calls reflected correctly in the CDR.

------------------------------------------------------------------------
r10003 | raarts | 2007-01-29 19:08:10 +0100 (Mon, 29 Jan 2007) | 3 lines

Set default values for the src_* and dst_* columns
in the dialplan on incoming and outgoing calls.

------------------------------------------------------------------------
r10002 | raarts | 2007-01-29 18:58:54 +0100 (Mon, 29 Jan 2007) | 3 lines

Also put the areacode in the dialplan, so we can
use it to normalize phonenumbers..

------------------------------------------------------------------------
r9997 | raarts | 2007-01-29 17:06:21 +0100 (Mon, 29 Jan 2007) | 3 lines

Split contact, company, department and trunk into incoming
and outgoing (source and destination columns.

------------------------------------------------------------------------
r9987 | raarts | 2007-01-29 14:43:18 +0100 (Mon, 29 Jan 2007) | 3 lines

Turn our country international prefix (31 for Netherlands) into
a global variable in the dialplan.

------------------------------------------------------------------------
r9975 | ivar | 2007-01-29 10:50:54 +0100 (Mon, 29 Jan 2007) | 1 line

support enabling/disabling dhcp
------------------------------------------------------------------------
r9966 | raarts | 2007-01-28 17:49:31 +0100 (Sun, 28 Jan 2007) | 5 lines

Ensure that calls to phoneline numbers that are actually trunks will
be billed correctly, by putting something that ends with -outbound
in the dcontext field of the CDR. That's hhow we know this is an
outgoing call.

------------------------------------------------------------------------
r9965 | raarts | 2007-01-27 18:00:28 +0100 (Sat, 27 Jan 2007) | 4 lines

Attached trunkid variables to incoming trunk channels.
Added new Set commands, so the values will be written into the dialplan.
Changed the deprecated SetVar() into the new Set() function.

------------------------------------------------------------------------
r9963 | raarts | 2007-01-27 17:57:17 +0100 (Sat, 27 Jan 2007) | 5 lines

Write values for contact, department, company and trunk to the CDR
in the dialplan. The CDR is extended with these new columns, and
the modified asterisk cdr_pgsql.c will save the values in these
columns.

------------------------------------------------------------------------
r9958 | raarts | 2007-01-26 20:41:34 +0100 (Fri, 26 Jan 2007) | 2 lines

Don't call macro dial-answered. Not needed for the moment.

------------------------------------------------------------------------
r9953 | raarts | 2007-01-26 14:56:11 +0100 (Fri, 26 Jan 2007) | 2 lines

Add TDM2400P to unload list

------------------------------------------------------------------------
r9946 | raarts | 2007-01-26 09:55:33 +0100 (Fri, 26 Jan 2007) | 2 lines

Properly handle calls the are blocked because of invalid permissions.

------------------------------------------------------------------------
r9942 | raarts | 2007-01-24 17:33:36 +0100 (Wed, 24 Jan 2007) | 3 lines

Don't let all iaxmodem processes register at the same moment.
And not every 60 seconds. Every 5 minutes suffices.

------------------------------------------------------------------------
r9923 | raarts | 2007-01-23 11:30:06 +0100 (Tue, 23 Jan 2007) | 2 lines

Try not to crash on malformed activate records.

------------------------------------------------------------------------
r9905 | raarts | 2007-01-22 11:46:35 +0100 (Mon, 22 Jan 2007) | 2 lines

Only test for allow_dial_voip if this is a voip trunk.

------------------------------------------------------------------------
r9881 | ivar | 2007-01-18 14:07:19 +0100 (Thu, 18 Jan 2007) | 1 line

(NeoNova 26545) Fix in astcfgd - postfix.php : add myhostname = company hostname
------------------------------------------------------------------------
r9880 | raarts | 2007-01-17 23:32:07 +0100 (Wed, 17 Jan 2007) | 6 lines

Callee transfer didnot work when the call got to you
through a call that was forwarded from a queue with honour forwarding on.
Reported by Rosmax. Fixed.

Also fixed a crash in astcfgd.

------------------------------------------------------------------------
r9841 | raarts | 2007-01-14 13:55:13 +0100 (Sun, 14 Jan 2007) | 2 lines

Unified all the trunk_* tables into one: the trunk table.

------------------------------------------------------------------------
r9839 | raarts | 2007-01-13 10:54:16 +0100 (Sat, 13 Jan 2007) | 3 lines

Skip deleted phonelines on voicemail retention checking, and
skip phonelines without a voicemail  box as well.

------------------------------------------------------------------------
r9838 | raarts | 2007-01-13 10:49:05 +0100 (Sat, 13 Jan 2007) | 2 lines

Do your work quietly, so cron will not mail root every time this script runs.

------------------------------------------------------------------------
r9837 | raarts | 2007-01-13 10:47:45 +0100 (Sat, 13 Jan 2007) | 2 lines

Fix sql error.

------------------------------------------------------------------------
r9770 | raarts | 2007-01-09 00:02:12 +0100 (Tue, 09 Jan 2007) | 3 lines

If a call is sent to voicemail, conference box, forwarded, or recorded,
note that in the userfield of the CDR, so we can bill it later if we want to.

------------------------------------------------------------------------
r9766 | raarts | 2007-01-08 14:19:07 +0100 (Mon, 08 Jan 2007) | 3 lines

Try to limit the max amount of incoming lines. Does not
work for Zap channels yet. Needs more work.

------------------------------------------------------------------------
r9765 | raarts | 2007-01-08 13:21:36 +0100 (Mon, 08 Jan 2007) | 3 lines

Set variables so we can determine the max amount of
incoming calls on SIP and IAX in the dialplan.

------------------------------------------------------------------------
r9761 | raarts | 2007-01-08 12:39:26 +0100 (Mon, 08 Jan 2007) | 2 lines

Support for max calls in the dialplan

------------------------------------------------------------------------
r9760 | raarts | 2007-01-08 12:36:45 +0100 (Mon, 08 Jan 2007) | 2 lines

Plugin support in the dialplan.

------------------------------------------------------------------------
r9744 | raarts | 2007-01-06 20:43:38 +0100 (Sat, 06 Jan 2007) | 2 lines

plugin should be plugins of course

------------------------------------------------------------------------
r9743 | raarts | 2007-01-06 20:42:15 +0100 (Sat, 06 Jan 2007) | 2 lines

Renamed package to plugin

------------------------------------------------------------------------
r9742 | raarts | 2007-01-06 20:41:43 +0100 (Sat, 06 Jan 2007) | 2 lines

renamed package to plugin.

------------------------------------------------------------------------
r9741 | raarts | 2007-01-06 20:10:05 +0100 (Sat, 06 Jan 2007) | 2 lines

Of course the variable userid should persist across channels.

------------------------------------------------------------------------
r9740 | raarts | 2007-01-06 17:53:38 +0100 (Sat, 06 Jan 2007) | 3 lines

Add extra code to identify an employee on external incoming calls
(by callerid)

------------------------------------------------------------------------
r9689 | raarts | 2007-01-01 23:45:26 +0100 (Mon, 01 Jan 2007) | 3 lines

Infrastructure to add applications as packages to the Astium.
Well, it's just a small start.

------------------------------------------------------------------------
r9682 | raarts | 2006-12-29 17:37:26 +0100 (Fri, 29 Dec 2006) | 2 lines

FD PIN support in the stdastium dialplan.

------------------------------------------------------------------------
r9680 | raarts | 2006-12-29 17:29:18 +0100 (Fri, 29 Dec 2006) | 2 lines

fdpin support.

------------------------------------------------------------------------
r9674 | raarts | 2006-12-29 13:53:47 +0100 (Fri, 29 Dec 2006) | 3 lines

When restarting zaptel, first stop asterisk, and start it afterwards.
This tries to prevent drivers crashes..

------------------------------------------------------------------------
r9639 | raarts | 2006-12-27 23:47:07 +0100 (Wed, 27 Dec 2006) | 4 lines

The phoneline function 'normal' is split into 'contact' and 'free'.
Adapt astcfgd and the stdastium.inc so they will use 'contact'
and the lookupid field as the contact id.

------------------------------------------------------------------------
r9620 | raarts | 2006-12-24 23:13:05 +0100 (Sun, 24 Dec 2006) | 5 lines

Enable *21, *22, *23 codes for new forwarding options.
Add *24 for forwarding when logged off.
Renamed followme (was *24) to *27.


------------------------------------------------------------------------
r9539 | raarts | 2006-12-18 16:37:25 +0100 (Mon, 18 Dec 2006) | 2 lines

Use proper wildcard pattern matching.

------------------------------------------------------------------------
r9537 | raarts | 2006-12-18 15:46:03 +0100 (Mon, 18 Dec 2006) | 3 lines

Don't write a phone context for each company any more, use the
generalized one.

------------------------------------------------------------------------
r9536 | raarts | 2006-12-18 15:39:56 +0100 (Mon, 18 Dec 2006) | 4 lines

Do not generate a special phone context for each company any
more. It is now generalized enough that we can put it in 
/etc/asterisk/extensions.conf

------------------------------------------------------------------------
r9520 | raarts | 2006-12-17 17:08:41 +0100 (Sun, 17 Dec 2006) | 2 lines

Add Reset scripts for ZAP channels. Called when asterisk is restarted.

------------------------------------------------------------------------
r9513 | raarts | 2006-12-16 22:04:10 +0100 (Sat, 16 Dec 2006) | 3 lines

penalty starts at 1 for end-users, but at '0' for us programmers.
Also fixed a bug where IAX phones could not be members of queues.

------------------------------------------------------------------------
r9454 | raarts | 2006-12-14 14:49:04 +0100 (Thu, 14 Dec 2006) | 2 lines

Ensure that Swissvoice phones upgrade to newest firmware.

------------------------------------------------------------------------
r9432 | raarts | 2006-12-13 23:32:37 +0100 (Wed, 13 Dec 2006) | 3 lines

The subscrlen field does not exist any more in the
company table..

------------------------------------------------------------------------
r9430 | raarts | 2006-12-13 23:00:48 +0100 (Wed, 13 Dec 2006) | 3 lines

Process the new sipdomain parameter instead of the 
previous domainname.

------------------------------------------------------------------------
r9362 | ivar | 2006-12-12 11:24:07 +0100 (Tue, 12 Dec 2006) | 4 lines

Merged revisions 9043-9360 via svnmerge from
/astcfgd/branches/1.2


------------------------------------------------------------------------
r9361 | ivar | 2006-12-12 11:23:50 +0100 (Tue, 12 Dec 2006) | 1 line

init svnmerge
------------------------------------------------------------------------
r9352 | nnrelman | 2006-12-11 16:51:30 +0100 (Mon, 11 Dec 2006) | 1 line


------------------------------------------------------------------------
r9264 | nnrelman | 2006-12-08 10:42:20 +0100 (Fri, 08 Dec 2006) | 1 line

moved some setup code from astium-setup to the astcfgd specfile.
------------------------------------------------------------------------
r9091 | nnrelman | 2006-11-30 15:50:27 +0100 (Thu, 30 Nov 2006) | 1 line

Updated version to 1.4
------------------------------------------------------------------------
r9042 | nnrelman | 2006-11-29 17:05:56 +0100 (Wed, 29 Nov 2006) | 1 line

Created branche 1.4 for astcfgd
------------------------------------------------------------------------
r8945 | raarts | 2006-11-24 13:07:56 +0100 (Fri, 24 Nov 2006) | 2 lines

If the fax part of astcfgd changes /etc/inittab, tel the init process about it.

------------------------------------------------------------------------
r8858 | raarts | 2006-11-21 15:09:37 +0100 (Tue, 21 Nov 2006) | 2 lines

SetLanguage is deprecated, use Set(LANGUAGE()=xx) instead.

------------------------------------------------------------------------
r8835 | raarts | 2006-11-20 12:53:47 +0100 (Mon, 20 Nov 2006) | 2 lines

Small typo fixes

------------------------------------------------------------------------
r8832 | ivar | 2006-11-20 12:26:33 +0100 (Mon, 20 Nov 2006) | 1 line

(NeoNova 24079) properly set status of phonelines to 'deleted' if associated conference is deleted.
------------------------------------------------------------------------
r8831 | raarts | 2006-11-20 11:58:15 +0100 (Mon, 20 Nov 2006) | 2 lines

Record outgoing calls. This feature was broken.

------------------------------------------------------------------------
r8818 | raarts | 2006-11-20 10:24:41 +0100 (Mon, 20 Nov 2006) | 3 lines

Fix setextcallerid (again), and fix several cases of
inappropriate use of the LEN() function.

------------------------------------------------------------------------
r8810 | raarts | 2006-11-19 20:29:58 +0100 (Sun, 19 Nov 2006) | 3 lines

The '==' operator does not exist in asterisk expression. Use '='
to test if two things are equal.

------------------------------------------------------------------------
r8804 | raarts | 2006-11-18 19:16:03 +0100 (Sat, 18 Nov 2006) | 5 lines

Improved logging levels. Now queries are echoed when debug = 3 instead of 2.
Created new function log_system that executes a command and logs its output to 
log_system.out in the temp directory. 
Generate internationalprefix variable in globals.conf.

------------------------------------------------------------------------
r8803 | raarts | 2006-11-18 19:11:20 +0100 (Sat, 18 Nov 2006) | 2 lines

Replace leading '+' with the international prefix (usually '00')..

------------------------------------------------------------------------
r8802 | raarts | 2006-11-18 15:02:20 +0100 (Sat, 18 Nov 2006) | 2 lines

userid variable should be an inheriting variable.

------------------------------------------------------------------------
r8801 | raarts | 2006-11-18 00:00:22 +0100 (Sat, 18 Nov 2006) | 4 lines

If an outside call is forwarded, take ownership of the call.
Must be done because else you cannot do Callee transfer back into
the PBX.

------------------------------------------------------------------------
r8800 | raarts | 2006-11-17 17:23:43 +0100 (Fri, 17 Nov 2006) | 2 lines

Do not use phone->maxcalls variable. That is wrong. Use $maxcalls instead.

------------------------------------------------------------------------
r8786 | ivar | 2006-11-17 11:02:11 +0100 (Fri, 17 Nov 2006) | 1 line

(NeoNova 22878) add missing slash, should solve ping.cgi problem
------------------------------------------------------------------------
r8772 | raarts | 2006-11-16 10:09:24 +0100 (Thu, 16 Nov 2006) | 2 lines

Fixed small typo in setting callerid in outbound calls

------------------------------------------------------------------------
r8770 | raarts | 2006-11-15 17:09:03 +0100 (Wed, 15 Nov 2006) | 2 lines

Fix in TAPI support contexts.

------------------------------------------------------------------------
r8769 | raarts | 2006-11-15 16:32:21 +0100 (Wed, 15 Nov 2006) | 3 lines

The music on hold as set up in the company screen, should also
be the music on hold for calls that are initiated by phones.

------------------------------------------------------------------------
r8765 | raarts | 2006-11-15 13:19:41 +0100 (Wed, 15 Nov 2006) | 3 lines

Extend maintenance, thus that whenever DayLight Savings time
rolls over, all phones will be reprovisioned.

------------------------------------------------------------------------
r8737 | raarts | 2006-11-14 23:04:54 +0100 (Tue, 14 Nov 2006) | 2 lines

Don't generate a CDR record if inhibit_cdr == 'yes'.

------------------------------------------------------------------------
r8733 | raarts | 2006-11-14 18:22:10 +0100 (Tue, 14 Nov 2006) | 4 lines

Write all phone info in a file called phones.conf. This is callable
through the manager interface, without setting any variables,
so you can use this to connect external dialers.

------------------------------------------------------------------------
r8732 | raarts | 2006-11-14 15:34:57 +0100 (Tue, 14 Nov 2006) | 2 lines

First step in supporting local dialplan settings.

------------------------------------------------------------------------
r8694 | raarts | 2006-11-11 11:26:26 +0100 (Sat, 11 Nov 2006) | 2 lines

Added log server syntax. Not used yet.

------------------------------------------------------------------------
r8692 | raarts | 2006-11-10 23:55:24 +0100 (Fri, 10 Nov 2006) | 2 lines

Put a nicer text on the display if nobody is logged on.

------------------------------------------------------------------------
r8689 | raarts | 2006-11-10 22:45:09 +0100 (Fri, 10 Nov 2006) | 2 lines

Fix omission in generating programmable keys on 9112i.

------------------------------------------------------------------------
r8688 | raarts | 2006-11-10 22:30:09 +0100 (Fri, 10 Nov 2006) | 2 lines

Small counter bug prevented 9122i from working.

------------------------------------------------------------------------
r8687 | raarts | 2006-11-10 22:28:56 +0100 (Fri, 10 Nov 2006) | 2 lines

Give timeout of 5 seconds, and no retries when rebooting Aastra phones.

------------------------------------------------------------------------
r8686 | raarts | 2006-11-10 21:51:44 +0100 (Fri, 10 Nov 2006) | 10 lines

Callee transfer now follows the permissions of the user that dialed
the callee. This means that if I cannot dial international calls,
and I dial a colleague on his cellphone, he will not be able to
transfer the call to an international number using callee transfer.

On the other hand, if I dial the colleague's internal number, and
he has his phone forwarded to his cellphone, *then* he can use
callee transfer to an international number if he's normally allowed
to dial internationally from his internal phone.

------------------------------------------------------------------------
r8684 | raarts | 2006-11-10 21:22:56 +0100 (Fri, 10 Nov 2006) | 3 lines

Ensure that a callee can not only transfer to internal 
extensions, but also to outside numbers.

------------------------------------------------------------------------
r8659 | raarts | 2006-11-09 23:30:09 +0100 (Thu, 09 Nov 2006) | 4 lines

Don't optimize Local channels out of the dialpath when
they are called from queues. This ensures callee transfer
keeps working, for example.

------------------------------------------------------------------------
r8658 | raarts | 2006-11-09 23:28:47 +0100 (Thu, 09 Nov 2006) | 3 lines

Fix Busy macro incorrectly checking for the FROM_QUEUE variable
Fix parse error in s-fwd part of Macro extension.

------------------------------------------------------------------------
r8657 | raarts | 2006-11-09 21:53:30 +0100 (Thu, 09 Nov 2006) | 2 lines

Only save last dialed number if it is *not* the '##' (redial last numer) sequence.

------------------------------------------------------------------------
r8656 | raarts | 2006-11-09 21:18:29 +0100 (Thu, 09 Nov 2006) | 2 lines

Fix 9992, it didnt say which department you belong to. Now it does.

------------------------------------------------------------------------
r8652 | raarts | 2006-11-09 16:53:27 +0100 (Thu, 09 Nov 2006) | 3 lines

Backend changes to support the Set External Caller ID option in the 
outbound dial plans.

------------------------------------------------------------------------
r8651 | ivar | 2006-11-09 16:47:48 +0100 (Thu, 09 Nov 2006) | 1 line

bugfix: re-generate tftp phone configuration when changing register_ip
------------------------------------------------------------------------
r8578 | raarts | 2006-11-07 15:32:35 +0100 (Tue, 07 Nov 2006) | 2 lines

Fix disa without password

------------------------------------------------------------------------
r8562 | nnrelman | 2006-11-07 13:05:45 +0100 (Tue, 07 Nov 2006) | 1 line

moved current to 1.2.
------------------------------------------------------------------------
r8554 | raarts | 2006-11-06 17:35:29 +0100 (Mon, 06 Nov 2006) | 4 lines

oh my, these Aastra phones are not easy to provision correctly...
Ensure that speeddial and monitor buttons work on a separate
SIP line.

------------------------------------------------------------------------
r8553 | raarts | 2006-11-06 17:16:27 +0100 (Mon, 06 Nov 2006) | 3 lines

It seems older AAstra phones need next-server, and newer
need option tftp-server-name in DHCP.

------------------------------------------------------------------------
r8552 | raarts | 2006-11-06 14:19:39 +0100 (Mon, 06 Nov 2006) | 2 lines

Just saving how to reboot a cisco by telnet

------------------------------------------------------------------------
r8549 | raarts | 2006-11-06 10:06:34 +0100 (Mon, 06 Nov 2006) | 2 lines

Fix parse error.

------------------------------------------------------------------------
r8547 | raarts | 2006-11-05 23:42:23 +0100 (Sun, 05 Nov 2006) | 2 lines

Fix linebutton allocation for Aastra phones

------------------------------------------------------------------------
r8545 | raarts | 2006-11-05 11:43:57 +0100 (Sun, 05 Nov 2006) | 2 lines

Let's not loose our SIP registration shall we?

------------------------------------------------------------------------
r8544 | raarts | 2006-11-04 17:07:43 +0100 (Sat, 04 Nov 2006) | 2 lines

Support loading the users' fastdial list to the Aastra phones.

------------------------------------------------------------------------
r8542 | raarts | 2006-11-04 16:39:35 +0100 (Sat, 04 Nov 2006) | 2 lines

Aastra 9133 Support

------------------------------------------------------------------------
r8541 | raarts | 2006-11-04 12:24:27 +0100 (Sat, 04 Nov 2006) | 2 lines

Use the admin password that we give to the phone to make it reboot. (Aastra)

------------------------------------------------------------------------
r8531 | raarts | 2006-11-02 23:32:20 +0100 (Thu, 02 Nov 2006) | 2 lines

Wait 2 seconds before rebooting Grandstream phones. More reliable.

------------------------------------------------------------------------
r8529 | raarts | 2006-11-02 22:50:44 +0100 (Thu, 02 Nov 2006) | 2 lines

Support for Grandstream GXV-3000 Video Phones.

------------------------------------------------------------------------
r8518 | raarts | 2006-11-02 16:29:45 +0100 (Thu, 02 Nov 2006) | 6 lines

Fix intercom. I did not understand how it worked.
The intercom key on the Aastra is like a dialtone key.
Press it, then enter the extension of your colleague,
press dial. Then his phone will jump to handsfree mode.


------------------------------------------------------------------------
r8504 | raarts | 2006-11-02 13:56:27 +0100 (Thu, 02 Nov 2006) | 2 lines

Only generate function keys when someone is logged in to the phone.

------------------------------------------------------------------------
r8493 | raarts | 2006-11-02 12:40:11 +0100 (Thu, 02 Nov 2006) | 2 lines

Support for Aastra 480i function keys

------------------------------------------------------------------------
r8479 | raarts | 2006-11-02 10:45:54 +0100 (Thu, 02 Nov 2006) | 2 lines

Support for Aastra phones. Not the softkeys yet, that's for the next commit.

------------------------------------------------------------------------
r8471 | raarts | 2006-11-01 17:42:55 +0100 (Wed, 01 Nov 2006) | 3 lines

Fix IAXy provisioning (and astcfgd crashing when provisioning failed)
Aastra now displays the owner of the phone more nicely.

------------------------------------------------------------------------
r8467 | raarts | 2006-11-01 16:54:47 +0100 (Wed, 01 Nov 2006) | 2 lines

Set proper DHCP parms for Aastra phones.

------------------------------------------------------------------------
r8458 | raarts | 2006-11-01 10:23:12 +0100 (Wed, 01 Nov 2006) | 2 lines

Polycom behind NAT is now supported.

------------------------------------------------------------------------
r8435 | raarts | 2006-10-30 12:28:16 +0100 (Mon, 30 Oct 2006) | 2 lines

Set pickupgroup and callgroup explicitly to zero for phonelines (not for phones of course)

------------------------------------------------------------------------
r8430 | raarts | 2006-10-30 10:52:14 +0100 (Mon, 30 Oct 2006) | 2 lines

Revert Thomson to firmware 1.48. The 1.50 is unreliable.

------------------------------------------------------------------------
r8408 | raarts | 2006-10-27 10:13:04 +0200 (Fri, 27 Oct 2006) | 3 lines

Improve handling of phone login. Together with the new SNOM 6.2.3
firmware now it always succeeds and phones always get reprovisioned.

------------------------------------------------------------------------
r8403 | raarts | 2006-10-26 15:58:33 +0200 (Thu, 26 Oct 2006) | 2 lines

New Thomson firmware support, including suppressing the missed calls indication.

------------------------------------------------------------------------
r8399 | raarts | 2006-10-26 13:14:05 +0200 (Thu, 26 Oct 2006) | 2 lines

Let's make the Polycom 601 extension pad work!

------------------------------------------------------------------------
r8395 | raarts | 2006-10-25 17:32:18 +0200 (Wed, 25 Oct 2006) | 2 lines

Fix timing out in IVR menus

------------------------------------------------------------------------
r8375 | raarts | 2006-10-23 15:15:04 +0200 (Mon, 23 Oct 2006) | 2 lines

Moved a line from httpd.conf to astium.conf in /etc/httpd/conf.d

------------------------------------------------------------------------
r8353 | raarts | 2006-10-19 23:21:39 +0200 (Thu, 19 Oct 2006) | 2 lines

Improved logging to the eventlog.

------------------------------------------------------------------------
r8341 | raarts | 2006-10-18 17:28:28 +0200 (Wed, 18 Oct 2006) | 2 lines

Make sure the public IP address is in the proper SIP domain so phones can register properly.

------------------------------------------------------------------------
r8274 | raarts | 2006-10-13 16:59:08 +0200 (Fri, 13 Oct 2006) | 2 lines

Added script to scan the attached network for VoIP phones.

------------------------------------------------------------------------
r8271 | raarts | 2006-10-13 11:07:33 +0200 (Fri, 13 Oct 2006) | 4 lines

From now on, whenever astcfgd is upgraded (when running yum upgrade for example)
everyone is logged of from the Web GUI, so when new fields show up,
they also will have online help.

------------------------------------------------------------------------
r8268 | raarts | 2006-10-13 10:20:01 +0200 (Fri, 13 Oct 2006) | 3 lines

When the GUI says: show queue name on phone, it should prepend
the queue name to the callername, instead of replacing it.

------------------------------------------------------------------------
r8267 | raarts | 2006-10-13 10:19:04 +0200 (Fri, 13 Oct 2006) | 3 lines

When a call drops out of a queue, set FROM_QUEUE to no,
otherwise the Macro's refuse and Busy will give wrong results.

------------------------------------------------------------------------
r8265 | raarts | 2006-10-12 17:45:33 +0200 (Thu, 12 Oct 2006) | 2 lines

When you page one specific phone, always put it in talkback mode.

------------------------------------------------------------------------
r8264 | raarts | 2006-10-12 17:44:01 +0200 (Thu, 12 Oct 2006) | 2 lines

when a DISA password is not required, note that in the dialplan.

------------------------------------------------------------------------
r8260 | raarts | 2006-10-12 17:33:38 +0200 (Thu, 12 Oct 2006) | 2 lines

Fix DISA support.

------------------------------------------------------------------------
r8238 | raarts | 2006-10-11 13:58:12 +0200 (Wed, 11 Oct 2006) | 2 lines

An E1/T1 is never a phone, always a line.

------------------------------------------------------------------------
r8236 | ivar | 2006-10-11 11:40:31 +0200 (Wed, 11 Oct 2006) | 1 line

spelling errors
------------------------------------------------------------------------
r8231 | raarts | 2006-10-11 09:07:18 +0200 (Wed, 11 Oct 2006) | 2 lines

Generate an ISDN-30 trunk as a line in /etc/asterisk/zapata.conf

------------------------------------------------------------------------
r8230 | raarts | 2006-10-10 17:19:14 +0200 (Tue, 10 Oct 2006) | 2 lines

Add missing field, so allow_dial_voip is not emtpy generated in zapata.conf.

------------------------------------------------------------------------
r8219 | raarts | 2006-10-10 10:15:01 +0200 (Tue, 10 Oct 2006) | 4 lines

Rely on the extension macro to handle calls, don't generate
queue etc commands ourselves any more. this fixed problems
on our own helpdesk.

------------------------------------------------------------------------
r8200 | raarts | 2006-10-09 10:02:56 +0200 (Mon, 09 Oct 2006) | 2 lines

Oops, don't recreate the company key where doing a 'module all'.

------------------------------------------------------------------------
r8199 | raarts | 2006-10-09 09:54:39 +0200 (Mon, 09 Oct 2006) | 2 lines

Using '==' instead of '=' does not work in the dialplan...

------------------------------------------------------------------------
r8198 | raarts | 2006-10-09 09:50:07 +0200 (Mon, 09 Oct 2006) | 5 lines

If a call was placed using call.php interface or SOAP, and the
target extension was a queue, then the system asked 'agent login'.
Understandable, because nobody logs in to a queue extension, 
but still not what the user has in mind. This is now fixed.

------------------------------------------------------------------------
r8196 | raarts | 2006-10-09 08:06:24 +0200 (Mon, 09 Oct 2006) | 4 lines

Now automagically recreate all config files (except zaptel, which must be 
regenerated explicitly, because it forces an asterisk restart) whenever astcfgd
is upgraded.

------------------------------------------------------------------------
r8183 | raarts | 2006-10-05 21:50:12 +0200 (Thu, 05 Oct 2006) | 3 lines

Part of the dialplan now logs call progress in the eventlog
(if enabled by the astglobal 'eventlog')

------------------------------------------------------------------------
r8182 | raarts | 2006-10-05 21:40:12 +0200 (Thu, 05 Oct 2006) | 4 lines

Generate new file: /etc/asterisk/globals.conf containing various
globals. These globals were previously generated in /etc/asterisk/companyxxxx/globals.conf
and thus were duplicated.

------------------------------------------------------------------------
r8179 | raarts | 2006-10-05 21:16:20 +0200 (Thu, 05 Oct 2006) | 3 lines

Only display the name of the queue on the phone display if
the showdisplay flag = yes.

------------------------------------------------------------------------
r8173 | raarts | 2006-10-04 15:58:08 +0200 (Wed, 04 Oct 2006) | 2 lines

Delete entries from the eventlog that are older than a year.

------------------------------------------------------------------------
r8172 | raarts | 2006-10-04 15:56:54 +0200 (Wed, 04 Oct 2006) | 3 lines

Introduced maintenance script for the Astium.
Currently only deletes CDR records older than a year.

------------------------------------------------------------------------
r8166 | raarts | 2006-10-04 10:09:32 +0200 (Wed, 04 Oct 2006) | 2 lines

Set Digit Timeout, so IVR menu's will be recognized better.

------------------------------------------------------------------------
r8163 | raarts | 2006-10-03 20:45:12 +0200 (Tue, 03 Oct 2006) | 2 lines

Don't set the language. Should be already set elsewhere..

------------------------------------------------------------------------
r8158 | raarts | 2006-10-03 10:43:28 +0200 (Tue, 03 Oct 2006) | 2 lines

Also remove system-shutdown records on astcfgd startup.

------------------------------------------------------------------------
r8157 | raarts | 2006-10-03 09:47:03 +0200 (Tue, 03 Oct 2006) | 5 lines

Fix problems with honour forwarding and the 'Busy' and 'Refuse' settings.
If an extension was set to give a Busy tone when it was busy, calls from
the Queue would get a busy tone as well, so queues could not function on
lines with 'honour forwarding' enabled. This is now fixed.

------------------------------------------------------------------------
r8156 | raarts | 2006-10-02 16:48:33 +0200 (Mon, 02 Oct 2006) | 2 lines

Oops, when you copy a pice of code, you have to check if it works...

------------------------------------------------------------------------
r8154 | ivar | 2006-10-02 09:52:47 +0200 (Mon, 02 Oct 2006) | 1 line

(Neonova 20132) als astcfgd start, alle QUEUED records in activate met type='system-restart' op 'DONE' zetten
------------------------------------------------------------------------
r8152 | raarts | 2006-09-30 16:32:56 +0200 (Sat, 30 Sep 2006) | 2 lines

Fix diabling callwaiting for various types of phones.

------------------------------------------------------------------------
r8151 | raarts | 2006-09-30 16:12:05 +0200 (Sat, 30 Sep 2006) | 2 lines

More fixes for Grandstream provisioning.

------------------------------------------------------------------------
r8149 | raarts | 2006-09-30 02:18:15 +0200 (Sat, 30 Sep 2006) | 4 lines

Enable picking up a call on a blinking led on the Grandstream GXP-2000 phones.
Also this new feature was too easy to add: *8extension
If you know that extension 333 is ringing, you can pick it up by dialing *8333.

------------------------------------------------------------------------
r8147 | raarts | 2006-09-30 01:26:15 +0200 (Sat, 30 Sep 2006) | 2 lines

Grandstream GXP-2000: allow Intercom/paging support, subscribe for MWI.

------------------------------------------------------------------------
r8144 | raarts | 2006-09-30 01:03:32 +0200 (Sat, 30 Sep 2006) | 2 lines

Support for function keys and extension pad of Grandstream GXP-2000

------------------------------------------------------------------------
r8142 | raarts | 2006-09-29 23:16:15 +0200 (Fri, 29 Sep 2006) | 2 lines

Support for more than one line in Grandstream phones.

------------------------------------------------------------------------
r8141 | raarts | 2006-09-29 23:04:20 +0200 (Fri, 29 Sep 2006) | 2 lines

Now we know how to reboot Grandstream phones and ATA's.

------------------------------------------------------------------------
r8126 | raarts | 2006-09-28 21:42:35 +0200 (Thu, 28 Sep 2006) | 2 lines

First cut at Grandstream GXP-2000 support.

------------------------------------------------------------------------
r8120 | raarts | 2006-09-27 13:08:33 +0200 (Wed, 27 Sep 2006) | 2 lines

Allow calleer transfer on all outgoing calls.

------------------------------------------------------------------------
r8111 | raarts | 2006-09-26 19:59:23 +0200 (Tue, 26 Sep 2006) | 2 lines

Ensure all dependant packages are Required in the specfile.

------------------------------------------------------------------------
r8094 | ivar | 2006-09-26 10:00:31 +0200 (Tue, 26 Sep 2006) | 1 line

(NeoNova 19231) update conference script which deletes conferences
------------------------------------------------------------------------
r8089 | raarts | 2006-09-25 16:12:36 +0200 (Mon, 25 Sep 2006) | 4 lines

Corner case: when an incoming call was blind transferred to a number
in the company phonebook, the call was dropped. Fixed.
Calling 911 and 112 was impossible. Fixed.

------------------------------------------------------------------------
r8088 | raarts | 2006-09-25 15:39:10 +0200 (Mon, 25 Sep 2006) | 2 lines

One more fix for Thomson.

------------------------------------------------------------------------
r8087 | raarts | 2006-09-25 15:37:49 +0200 (Mon, 25 Sep 2006) | 2 lines

More fixes for Thomson phones

------------------------------------------------------------------------
r8084 | raarts | 2006-09-25 03:54:49 +0200 (Mon, 25 Sep 2006) | 2 lines

Fixed BLF (Monitor) keys on the Thomson.

------------------------------------------------------------------------
r8080 | raarts | 2006-09-24 22:45:26 +0200 (Sun, 24 Sep 2006) | 3 lines

Revised Thomson provisioning.
Fix SNOM300 DHCP options.

------------------------------------------------------------------------
r8077 | raarts | 2006-09-23 18:18:06 +0200 (Sat, 23 Sep 2006) | 2 lines

Don't process activate records that should be processed on a later date.

------------------------------------------------------------------------
r8072 | raarts | 2006-09-22 17:38:02 +0200 (Fri, 22 Sep 2006) | 2 lines

Create softlink to extensions.conf.master at installation time.

------------------------------------------------------------------------
r8064 | nnrelman | 2006-09-22 09:47:25 +0200 (Fri, 22 Sep 2006) | 1 line

fixed build script
------------------------------------------------------------------------
r8062 | raarts | 2006-09-21 23:16:52 +0200 (Thu, 21 Sep 2006) | 2 lines

Ensure a proper DHCP record exists for the GMT time offset.

------------------------------------------------------------------------
r8048 | raarts | 2006-09-21 12:07:25 +0200 (Thu, 21 Sep 2006) | 2 lines

Reversed a test which was wrong. It didn't work with Call-Me-Now.

------------------------------------------------------------------------
r8047 | raarts | 2006-09-21 11:42:50 +0200 (Thu, 21 Sep 2006) | 3 lines

Add astium.conf file to http directory for reverse proxy.
Fix SNOM bug: phones not rebooting after login.

------------------------------------------------------------------------
r8038 | raarts | 2006-09-19 22:41:10 +0200 (Tue, 19 Sep 2006) | 3 lines

Hand over the original DID number if an incoming call is directly transferred
to an outgoing trunk.

------------------------------------------------------------------------
r8022 | nnrelman | 2006-09-18 13:02:25 +0200 (Mon, 18 Sep 2006) | 1 line

Disabled cyrus support.
------------------------------------------------------------------------
r8021 | raarts | 2006-09-17 23:14:06 +0200 (Sun, 17 Sep 2006) | 3 lines

Make sure the DTMF feature also works with queues. The line shouldn't
be answered before the actual phone answers the line. 

------------------------------------------------------------------------
r8020 | raarts | 2006-09-17 23:09:43 +0200 (Sun, 17 Sep 2006) | 3 lines

Support sending DTMF in a forward field. Created so we can talk to
DECT PBX'es that can address single handset using some DTMF code.

------------------------------------------------------------------------
r8018 | raarts | 2006-09-17 22:24:56 +0200 (Sun, 17 Sep 2006) | 2 lines

Correctly write the echotraining variable to zapata.conf

------------------------------------------------------------------------
r8013 | raarts | 2006-09-17 16:01:01 +0200 (Sun, 17 Sep 2006) | 6 lines

Make callee transfer work. what is that? Suppose some dials your company, and your
phone is forwarded to your mobile. you pick up your mobile, but you decide a
colleague who is at his desk should handle the call. In that case: press '#'
wait for a dialtone, and dial the colleagues extension. The call is immediately
forwarded to his phone, and you will be hung up.

------------------------------------------------------------------------
r8012 | raarts | 2006-09-17 14:30:14 +0200 (Sun, 17 Sep 2006) | 4 lines

when in the directory (*35) only consider real persons and not 
queues, ivr menus etcetera. Also allow pressing '0' to get to
an operator.

------------------------------------------------------------------------
r8010 | raarts | 2006-09-17 13:29:25 +0200 (Sun, 17 Sep 2006) | 3 lines

Fixed the bug where a conversation was being recorded, even
though recording was disabled.

------------------------------------------------------------------------
r8008 | raarts | 2006-09-17 13:04:17 +0200 (Sun, 17 Sep 2006) | 2 lines

Support enabling/disabling allow dial via voip.

------------------------------------------------------------------------
r8000 | raarts | 2006-09-16 23:41:01 +0200 (Sat, 16 Sep 2006) | 2 lines

Handle things correctly if the CallerID num is filled in in the trunk screens.

------------------------------------------------------------------------
r7999 | raarts | 2006-09-16 23:33:06 +0200 (Sat, 16 Sep 2006) | 2 lines

Do not change the caller ID name if it isn't necessary..

------------------------------------------------------------------------
r7958 | ivar | 2006-09-13 12:28:00 +0200 (Wed, 13 Sep 2006) | 1 line

support for setting smtpserver in astglobals. Not tested yet.
------------------------------------------------------------------------
r7951 | raarts | 2006-09-13 11:40:13 +0200 (Wed, 13 Sep 2006) | 2 lines

When register_ip changes, rebuild sip.conf

------------------------------------------------------------------------
r7905 | raarts | 2006-09-08 16:18:51 +0200 (Fri, 08 Sep 2006) | 2 lines

Rotate asterisk logs properly (keep them for a month, then throw them away)

------------------------------------------------------------------------
r7885 | raarts | 2006-09-06 15:27:14 +0200 (Wed, 06 Sep 2006) | 4 lines

While leaving a voicemail, you can now press '0' to reach
an operator.. (The operator's extension must be entereed into
the GUI of course).

------------------------------------------------------------------------
r7884 | raarts | 2006-09-06 15:09:53 +0200 (Wed, 06 Sep 2006) | 2 lines

Forward to voicemail when logged off didn't work. Fixed.

------------------------------------------------------------------------
r7883 | raarts | 2006-09-06 12:44:58 +0200 (Wed, 06 Sep 2006) | 2 lines

Strip and prefix in the outbound routing screen didn't work for Zap channels. Fixed.

------------------------------------------------------------------------
r7819 | raarts | 2006-08-29 21:24:48 +0200 (Tue, 29 Aug 2006) | 2 lines

Regenerate all upwatch file if any astglobal is changed.

------------------------------------------------------------------------
r7810 | raarts | 2006-08-29 09:35:50 +0200 (Tue, 29 Aug 2006) | 5 lines

Do not set the name of the callerid to zero if the number was not
found in the company table.
When setting company variables, use '__' to make them inheritable by
child channels.

------------------------------------------------------------------------
r7805 | raarts | 2006-08-28 12:49:49 +0200 (Mon, 28 Aug 2006) | 2 lines

Fixed pretty stupid programming error.

------------------------------------------------------------------------
r7801 | ivar | 2006-08-28 11:27:08 +0200 (Mon, 28 Aug 2006) | 1 line

(NeoNova 16872) it's a bug - when generating asterisk command which play soundfiles, any .gsm extension should be stripped. But you can't give the strrpos function a string as the needle - it will only take the first character which is a '.'. So anything after a '.' character was stripped.
------------------------------------------------------------------------
r7798 | raarts | 2006-08-26 11:18:51 +0200 (Sat, 26 Aug 2006) | 2 lines

Allow setting default music on hold for a company.

------------------------------------------------------------------------
r7793 | raarts | 2006-08-26 10:42:52 +0200 (Sat, 26 Aug 2006) | 2 lines

Only generate config files for company id > 1

------------------------------------------------------------------------
r7772 | wijnand | 2006-08-24 14:15:38 +0200 (Thu, 24 Aug 2006) | 3 lines

Use jitterbuffer=no for software faxes. 
The iaxmodem author said so on the maillinglist.

------------------------------------------------------------------------
r7758 | raarts | 2006-08-22 15:47:01 +0200 (Tue, 22 Aug 2006) | 2 lines

Allow networks in buildings to access the NTP server.

------------------------------------------------------------------------
r7757 | raarts | 2006-08-22 11:26:04 +0200 (Tue, 22 Aug 2006) | 2 lines

Incorrectly used building id as network id.

------------------------------------------------------------------------
r7756 | raarts | 2006-08-21 19:45:32 +0200 (Mon, 21 Aug 2006) | 2 lines

Fixed SOAP problems.

------------------------------------------------------------------------
r7754 | raarts | 2006-08-21 18:11:58 +0200 (Mon, 21 Aug 2006) | 2 lines

Didn't I fix this before? Remove greenpeace from dialplan.

------------------------------------------------------------------------
r7752 | raarts | 2006-08-21 17:08:03 +0200 (Mon, 21 Aug 2006) | 3 lines

Generate voicemailboxes for ivr's, queues, conferences too
now we can forward and do voicemail on them...

------------------------------------------------------------------------
r7749 | raarts | 2006-08-21 12:33:57 +0200 (Mon, 21 Aug 2006) | 3 lines

bug cms:17262: outbound nr trunks werkte niet correct
Fixed.

------------------------------------------------------------------------
r7739 | perry | 2006-08-08 14:30:12 +0200 (Tue, 08 Aug 2006) | 1 line

fixed recording of sound files.
------------------------------------------------------------------------
r7715 | raarts | 2006-08-03 16:58:37 +0200 (Thu, 03 Aug 2006) | 3 lines

plan-0 does not exist any more, send to plan-1 instead.
Show queue name on phone when a call comes in.

------------------------------------------------------------------------
r7709 | raarts | 2006-08-01 16:35:41 +0200 (Tue, 01 Aug 2006) | 2 lines

Fixed playback of recorded messages

------------------------------------------------------------------------
r7708 | raarts | 2006-08-01 14:36:00 +0200 (Tue, 01 Aug 2006) | 2 lines

Fix boot parameters for SNOM300 en 320.

------------------------------------------------------------------------
r7684 | raarts | 2006-07-27 11:50:13 +0200 (Thu, 27 Jul 2006) | 2 lines

Fix *1... not working

------------------------------------------------------------------------
r7660 | raarts | 2006-07-26 02:09:48 +0200 (Wed, 26 Jul 2006) | 9 lines

Ok, parked calls that timeout because nobody is picking
them up, normally go back to the extension that parked them.
But when that extension was busy, the caller that was parked
got disconnected. not Good. So I hacked it, and now it works.
Sort of.

Call Parking in asterisk sucks bigtime by the way...
 

------------------------------------------------------------------------
r7656 | raarts | 2006-07-25 22:00:33 +0200 (Tue, 25 Jul 2006) | 3 lines

Fixed bug with queue timeouts in the dialplan
Added support for periodic announcements.

------------------------------------------------------------------------
r7634 | raarts | 2006-07-22 21:23:36 +0200 (Sat, 22 Jul 2006) | 5 lines

Make call-forwarding work again.
Make call-parking work. Use *38 to park a call. The system choose
an extension for you between 970 and 989 for you.
You can now record soundprompts from a phone using *55

------------------------------------------------------------------------
r7605 | wijnand | 2006-07-17 16:06:22 +0200 (Mon, 17 Jul 2006) | 2 lines

Please leave cyrus in for a while. We are still using it.

------------------------------------------------------------------------
r7602 | raarts | 2006-07-17 13:07:21 +0200 (Mon, 17 Jul 2006) | 2 lines

Use the proper context for warding calls.

------------------------------------------------------------------------
r7597 | raarts | 2006-07-17 11:49:14 +0200 (Mon, 17 Jul 2006) | 3 lines

Handle blind transfers of outside lines. Send them to company-plan-0,
so the system does not think they are phones, and ask the caller to login to it...

------------------------------------------------------------------------
r7593 | raarts | 2006-07-16 21:14:37 +0200 (Sun, 16 Jul 2006) | 2 lines

Again revised call forwarding options.

------------------------------------------------------------------------
r7585 | raarts | 2006-07-15 21:15:00 +0200 (Sat, 15 Jul 2006) | 5 lines

Expressions fixes in stdastium.inc
Now uses 1,n,n,n etc numbering for most contexts.
Fix monitoring, didn't work before
Fix setting correct accountcode

------------------------------------------------------------------------
r7576 | raarts | 2006-07-14 10:24:45 +0200 (Fri, 14 Jul 2006) | 2 lines

Only sound the Busy signal for 15 seconds. Then hangup.

------------------------------------------------------------------------
r7572 | raarts | 2006-07-13 20:51:28 +0200 (Thu, 13 Jul 2006) | 3 lines

Fix voicemail bug.
Added busy tone if an extension can not be reached.

------------------------------------------------------------------------
r7564 | raarts | 2006-07-13 12:05:25 +0200 (Thu, 13 Jul 2006) | 3 lines

Use 'i' flag on zap channels when a number prefix is set (carrier preselect
on pridialplan=dynamic).

------------------------------------------------------------------------
r7558 | raarts | 2006-07-13 00:01:21 +0200 (Thu, 13 Jul 2006) | 3 lines

Greatly simpplified the code to generate the inbound context,
because we now have the nice new extension macro.

------------------------------------------------------------------------
r7557 | raarts | 2006-07-12 23:45:54 +0200 (Wed, 12 Jul 2006) | 2 lines

Process the special extensions in the extension macro as well.

------------------------------------------------------------------------
r7556 | raarts | 2006-07-12 21:58:37 +0200 (Wed, 12 Jul 2006) | 2 lines

Fixed Trunk handling, and handling of system numbers through the inbound context.

------------------------------------------------------------------------
r7555 | raarts | 2006-07-12 21:04:42 +0200 (Wed, 12 Jul 2006) | 3 lines

Check if voicemail is on before activating it.
Fixed logged off check.

------------------------------------------------------------------------
r7534 | raarts | 2006-07-10 11:59:36 +0200 (Mon, 10 Jul 2006) | 2 lines

Use the correct flag to disable CLID presentation.

------------------------------------------------------------------------
r7531 | raarts | 2006-07-09 23:47:38 +0200 (Sun, 09 Jul 2006) | 3 lines

It is now possible to forward queues, ivr menus etcetera, using the
phoneline (extension) forwarding options.

------------------------------------------------------------------------
r7498 | raarts | 2006-07-03 15:26:01 +0200 (Mon, 03 Jul 2006) | 3 lines

Put phones by default in the highest group possible,
or else they will be seen as outgoing lines...

------------------------------------------------------------------------
r7482 | raarts | 2006-06-28 22:46:25 +0200 (Wed, 28 Jun 2006) | 2 lines

Enable rebooting Linksys-941.

------------------------------------------------------------------------
r7481 | raarts | 2006-06-28 22:45:32 +0200 (Wed, 28 Jun 2006) | 2 lines

Fix astcfgd crash

------------------------------------------------------------------------
r7471 | raarts | 2006-06-27 21:53:06 +0200 (Tue, 27 Jun 2006) | 2 lines

Implemented paging support

------------------------------------------------------------------------
r7456 | raarts | 2006-06-27 00:06:33 +0200 (Tue, 27 Jun 2006) | 2 lines

Generate extensions for paging.

------------------------------------------------------------------------
r7445 | raarts | 2006-06-23 23:30:24 +0200 (Fri, 23 Jun 2006) | 4 lines

Allow SIP trunks with ports other then 5060.
When registering with a SIP server, give OUR main number as SIP contact.
Fix problem with registering with SIP servers and additional auth name.

------------------------------------------------------------------------
r7442 | raarts | 2006-06-23 01:14:12 +0200 (Fri, 23 Jun 2006) | 3 lines

set useclientcode=yes in sip.conf for SNOM phones that
have CMC enabled.

------------------------------------------------------------------------
r7425 | raarts | 2006-06-20 22:21:06 +0200 (Tue, 20 Jun 2006) | 74 lines

------------------------------------------------------------------------
r7249 | raarts | 2006-05-29 11:05:36 +0200 (Mon, 29 May 2006) | 5 lines
Changed paths:
   M /astcfgd/branches/1.0/src/dhcp.php
   M /astcfgd/branches/1.0/src/ntpd.php
   M /astcfgd/branches/1.0/src/stdastium.inc

Support for Polycom sending %23 when '#' is pressed at the beginning 
of a dialstring.
Any astglobal change is now sufficient for regenerating
ntp and dhcp config files.

------------------------------------------------------------------------
r7264 | raarts | 2006-05-31 11:24:15 +0200 (Wed, 31 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Now fastdial entries work without the starting '#'.

------------------------------------------------------------------------
r7266 | raarts | 2006-05-31 11:43:51 +0200 (Wed, 31 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Ensure that rebooting phones from the web works.

------------------------------------------------------------------------
r7275 | raarts | 2006-05-31 22:07:04 +0200 (Wed, 31 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Oops: make dialplans work for other companies than mbalance...

------------------------------------------------------------------------
r7278 | raarts | 2006-06-01 15:25:33 +0200 (Thu, 01 Jun 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/stdastium.inc

Set new departmentid and fromext when a call is being forwarded.
Fixes a problem for incoming calls that are forwarded immediately out.

------------------------------------------------------------------------
r7279 | raarts | 2006-06-01 15:37:31 +0200 (Thu, 01 Jun 2006) | 4 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Set proper variables for inbound IAX lines
Set companyid as well as company while in pre-inbound
Regenerate iax.conf and sip.conf when a company is changed.

------------------------------------------------------------------------
r7286 | raarts | 2006-06-07 14:03:46 +0200 (Wed, 07 Jun 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php
   M /astcfgd/branches/1.0/src/stdastium.inc

Changes to make recording work even on queues, and when transferring calls.
This could be done better, but that is for version 1.2.

------------------------------------------------------------------------
r7295 | wijnand | 2006-06-08 15:40:27 +0200 (Thu, 08 Jun 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Don't use deleted phonelines in user contexts, prevents asterisk warnings and potential problems.

------------------------------------------------------------------------
r7394 | raarts | 2006-06-16 20:17:07 +0200 (Fri, 16 Jun 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/zaptel.php

Also rebuild /etc/zaptel.conf if a zap_channel is changed.


------------------------------------------------------------------------
r7419 | raarts | 2006-06-20 17:21:03 +0200 (Tue, 20 Jun 2006) | 3 lines

Fix basic provisioning of Swissvoice phones.
Add fastdial support without the leading '#'.

------------------------------------------------------------------------
r7413 | raarts | 2006-06-20 09:45:51 +0200 (Tue, 20 Jun 2006) | 2 lines

Fix errors in PATH expressions in the dialplan.

------------------------------------------------------------------------
r7412 | raarts | 2006-06-19 23:40:32 +0200 (Mon, 19 Jun 2006) | 2 lines

Ok, now every company can have 2, 3 or 4 number extensions..

------------------------------------------------------------------------
r7409 | raarts | 2006-06-19 16:13:03 +0200 (Mon, 19 Jun 2006) | 2 lines

Support setting Swissvoice function keys in the backend.

------------------------------------------------------------------------
r7341 | raarts | 2006-06-11 14:12:43 +0200 (Sun, 11 Jun 2006) | 3 lines

Re-enable incoming multiple sip domains, because in asterisk 1.2.9
the bug with subscriptons is fixed. Now we can accept incoming sip calls again!

------------------------------------------------------------------------
r7294 | wijnand | 2006-06-08 15:37:53 +0200 (Thu, 08 Jun 2006) | 2 lines

Don't use deleted phonelines in user contexts, prevents asterisk warnings and potential problems.

------------------------------------------------------------------------
r7274 | raarts | 2006-05-31 22:06:00 +0200 (Wed, 31 May 2006) | 3 lines

Prefix channel variables with double underscores, so they will
be inherited into child channels.

------------------------------------------------------------------------
r7247 | raarts | 2006-05-29 00:17:23 +0200 (Mon, 29 May 2006) | 2 lines

Add 9996 extension to reprovision a phone, fix a small bug with ringtones.

------------------------------------------------------------------------
r7216 | raarts | 2006-05-22 11:55:38 +0200 (Mon, 22 May 2006) | 3 lines

Reworked the login procedure, so the dialplan will not reboot
the phone anymore, but astcfgd will instead.

------------------------------------------------------------------------
r7179 | raarts | 2006-05-16 22:59:13 +0200 (Tue, 16 May 2006) | 4 lines

Improved provisioning for cisco 790x phones. Now uses
the timezone as set in the building table.
Made more things configurable from the GUI.

------------------------------------------------------------------------
r7172 | raarts | 2006-05-16 11:37:25 +0200 (Tue, 16 May 2006) | 2 lines

Fix ringtone support across companies.

------------------------------------------------------------------------
r7160 | raarts | 2006-05-15 12:08:08 +0200 (Mon, 15 May 2006) | 8 lines

------------------------------------------------------------------------
r7159 | raarts | 2006-05-15 12:07:19 +0200 (Mon, 15 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

provisioning filenames for 790x should be in uppercase.


------------------------------------------------------------------------
r7138 | raarts | 2006-05-13 14:10:07 +0200 (Sat, 13 May 2006) | 2 lines

Renamed channel.button and phoneline.button to channel.line and phoneline.line

------------------------------------------------------------------------
r7124 | raarts | 2006-05-12 10:08:21 +0200 (Fri, 12 May 2006) | 17 lines

------------------------------------------------------------------------
r7103 | raarts | 2006-05-09 23:47:31 +0200 (Tue, 09 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

Support for the Sipura 9412-line phone.

------------------------------------------------------------------------
r7123 | raarts | 2006-05-12 09:59:35 +0200 (Fri, 12 May 2006) | 4 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) Do not create a separate SIP.CFG anymore, but put it in 
the phone-specific config file. This is needed for the new polycom firmware,
which is needed because of bugs,


------------------------------------------------------------------------
r7111 | raarts | 2006-05-10 13:43:47 +0200 (Wed, 10 May 2006) | 2 lines

Prevent eternal loops if a phonenumber is forwarded to itself.

------------------------------------------------------------------------
r7106 | raarts | 2006-05-10 00:01:09 +0200 (Wed, 10 May 2006) | 2 lines

Backend support for the linksys 941 2-line phone

------------------------------------------------------------------------
r7098 | raarts | 2006-05-09 21:58:28 +0200 (Tue, 09 May 2006) | 3 lines

Disabled support for incoming sip URI's becasue it messesup
the SNOM lights. Will  maybe fixed in a future asterisk version.

------------------------------------------------------------------------
r7092 | raarts | 2006-05-09 16:04:48 +0200 (Tue, 09 May 2006) | 3 lines

Support for pressing '0' (zero) while hearing voicemail instructions.
This will transfer to an operator.

------------------------------------------------------------------------
r7045 | raarts | 2006-05-08 00:23:09 +0200 (Mon, 08 May 2006) | 33 lines

------------------------------------------------------------------------
r6953 | raarts | 2006-05-02 21:57:24 +0200 (Tue, 02 May 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/astcfgd
   M /astcfgd/branches/1.0/src/asterisk.php
   M /astcfgd/branches/1.0/src/zaptel.php

Fixes for FC5. Fixed some queries with errors that only came up on PostgreSQL 8.1
Removed openxchage and cyrus support.

------------------------------------------------------------------------
r7032 | raarts | 2006-05-05 11:02:21 +0200 (Fri, 05 May 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) Userid and fromext were reversed in the calleridnum statement, which caused setting
callerid for a user and phoneline to not work. Fixed.

------------------------------------------------------------------------
r7033 | raarts | 2006-05-05 12:46:51 +0200 (Fri, 05 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Support NAT in sip.conf

------------------------------------------------------------------------
r7036 | raarts | 2006-05-05 15:11:03 +0200 (Fri, 05 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/stdastium.inc

(bugfix) *50 did not work for 3-digit extensions. Fixed.


------------------------------------------------------------------------
r7037 | raarts | 2006-05-05 23:43:59 +0200 (Fri, 05 May 2006) | 3 lines

Added *26XX. to talk immediately to someones voicemailbox.
Fixes for Thomson phones (ntp, voicemail)

------------------------------------------------------------------------
r7035 | raarts | 2006-05-05 14:45:36 +0200 (Fri, 05 May 2006) | 2 lines

Fix FXS impedance and CID signalling for GRandstream devices.

------------------------------------------------------------------------
r6963 | raarts | 2006-05-04 00:48:39 +0200 (Thu, 04 May 2006) | 11 lines

(feature) Accept incoming SIP URL's and connect them to the corresponding phone.
Needs the domainname being set properly on the company screen and needs a SRV
entry in DNS pointing to the Astium like this:

;setup to support SIP URL's
_sip._udp   SRV  0 0 5060 obelix-office.netland.nl. 

So now I can put <sip:raarts@neonova.nl> on my business card!
Outgoing calls of SIP URL's is another matter though..


------------------------------------------------------------------------
r6937 | raarts | 2006-05-02 13:51:15 +0200 (Tue, 02 May 2006) | 16 lines

------------------------------------------------------------------------
r6861 | raarts | 2006-04-28 16:12:15 +0200 (Fri, 28 Apr 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/BuildRpm
   M /astcfgd/branches/1.0/redhat/astcfgd.spec.in

Build fixes for FC5

------------------------------------------------------------------------
r6936 | raarts | 2006-05-02 13:50:09 +0200 (Tue, 02 May 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Fixed IAX trunks and RSA security.


------------------------------------------------------------------------
r6933 | raarts | 2006-05-02 11:46:58 +0200 (Tue, 02 May 2006) | 2 lines

Quick fix or better hack to make inbound IAX calls work.

------------------------------------------------------------------------
r6925 | raarts | 2006-05-02 00:24:23 +0200 (Tue, 02 May 2006) | 2 lines

Support for streaming mp3 from the internet.

------------------------------------------------------------------------
r6919 | raarts | 2006-05-01 11:09:45 +0200 (Mon, 01 May 2006) | 2 lines

A default musiconhold entry should always exist.

------------------------------------------------------------------------
r6885 | raarts | 2006-04-29 09:21:03 +0200 (Sat, 29 Apr 2006) | 2 lines

Support for enable/disable showing missed calls on Polycon

------------------------------------------------------------------------
r6881 | raarts | 2006-04-29 00:42:08 +0200 (Sat, 29 Apr 2006) | 2 lines

From now on the ntp servers in astglobals override the default one.

------------------------------------------------------------------------
r6879 | raarts | 2006-04-29 00:19:23 +0200 (Sat, 29 Apr 2006) | 3 lines

Support more provisionong options for IAXy's
Give Grandstream equipment a next-server in DHCP

------------------------------------------------------------------------
r6850 | raarts | 2006-04-27 23:34:56 +0200 (Thu, 27 Apr 2006) | 10 lines

------------------------------------------------------------------------
r6834 | raarts | 2006-04-26 15:07:23 +0200 (Wed, 26 Apr 2006) | 4 lines
Changed paths:
   M /astcfgd/branches/1.0/src/extensions.conf.master

(bugfix) when someone calls without callerid, the string 'asterisk' was shown
on the phone display. From now on it will be 'unknown' or whatever the phone prints
when callerid is withheld.


------------------------------------------------------------------------
r6845 | raarts | 2006-04-27 17:23:26 +0200 (Thu, 27 Apr 2006) | 3 lines

Ringing MP3 should be played a loud level. Made a special exception for that.
Also small fixes in the dialplan.

------------------------------------------------------------------------
r6825 | ivar | 2006-04-26 09:59:21 +0200 (Wed, 26 Apr 2006) | 1 line

(NeoNova 13284) have a redirect schedule redirect to an extension instead of an ivr menu
------------------------------------------------------------------------
r6804 | raarts | 2006-04-24 20:05:35 +0200 (Mon, 24 Apr 2006) | 9 lines

------------------------------------------------------------------------
r6797 | raarts | 2006-04-24 09:51:54 +0200 (Mon, 24 Apr 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/get-config.php
   M /astcfgd/branches/1.0/src/tftp.php

Fixes for Flemish language


------------------------------------------------------------------------
r6802 | raarts | 2006-04-24 19:59:55 +0200 (Mon, 24 Apr 2006) | 3 lines

When a call comes in from another companyi on the same astium, 
show it in the phone display.

------------------------------------------------------------------------
r6752 | raarts | 2006-04-20 11:49:11 +0200 (Thu, 20 Apr 2006) | 17 lines

Each user and each phone currently have their own dialplan/user context
in the extensions.conf. This makes the extensions.conf *really* large
if you have lots of users. My goal is to improve this situation
by creating generic contexts for phones and users.

This commit is step 1: create a generic context for phones/users.
It is now possible in asterisk to attach variables to channels in
sip.conf, iax.conf (and I patched it so it works with zapata.conf too).

Every time a user logs into a phone (or something is changed for a user)
then the corresponding sip/iax/zapata.conf is recreated, with new variables.

So now I have dumped the phone contexts, and the user contexts. 
Next step is: dump the user dialplans, and get that live from the database.

Also fixed MWI for Thomson phones by the way.

------------------------------------------------------------------------
r6738 | raarts | 2006-04-18 21:44:09 +0200 (Tue, 18 Apr 2006) | 2 lines

Fix Thomson provisioning error.

------------------------------------------------------------------------
r6737 | raarts | 2006-04-18 21:33:23 +0200 (Tue, 18 Apr 2006) | 2 lines

Fix Thomson provisioning: need proper boot filename in DHCP.

------------------------------------------------------------------------
r6736 | raarts | 2006-04-17 23:18:49 +0200 (Mon, 17 Apr 2006) | 2 lines

Initial support for Thomson phones.

------------------------------------------------------------------------
r6729 | raarts | 2006-04-16 00:36:51 +0200 (Sun, 16 Apr 2006) | 2 lines

For the moment hardcode the Aastra to CET, and DST.

------------------------------------------------------------------------
r6728 | raarts | 2006-04-15 23:50:22 +0200 (Sat, 15 Apr 2006) | 2 lines

Set correct date time format for Polycom, Swissvoice, Grandstream and Aastra phones

------------------------------------------------------------------------
r6725 | raarts | 2006-04-15 23:19:11 +0200 (Sat, 15 Apr 2006) | 4 lines

Enable NAT and QoS for Budgetone
Try to enable NAT for Cisco 79xx
Support NAT via STUN on Aastra

------------------------------------------------------------------------
r6724 | raarts | 2006-04-15 16:45:03 +0200 (Sat, 15 Apr 2006) | 2 lines

Fixed bug in aastra provisioning.

------------------------------------------------------------------------
r6723 | raarts | 2006-04-15 16:37:33 +0200 (Sat, 15 Apr 2006) | 2 lines

Support provisioning the Aastra phones.

------------------------------------------------------------------------
r6706 | ivar | 2006-04-14 10:50:52 +0200 (Fri, 14 Apr 2006) | 1 line

bugfix: don't query using  if you're in a loop processing results in that 
------------------------------------------------------------------------
r6689 | raarts | 2006-04-13 21:41:10 +0200 (Thu, 13 Apr 2006) | 2 lines

(feature) Support for a super company.

------------------------------------------------------------------------
r6683 | raarts | 2006-04-13 20:09:55 +0200 (Thu, 13 Apr 2006) | 23 lines

------------------------------------------------------------------------
r6602 | raarts | 2006-04-11 12:31:55 +0200 (Tue, 11 Apr 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/stdastium.inc

(bugfix) Do NOT play instructions for how to leave a voicemail message
when the user has entered his own away-message.

------------------------------------------------------------------------
r6639 | raarts | 2006-04-12 13:53:56 +0200 (Wed, 12 Apr 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(feature) Support enabling/disabling IAX trunking.

------------------------------------------------------------------------
r6679 | raarts | 2006-04-13 16:54:01 +0200 (Thu, 13 Apr 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/stdastium.inc

(bugfix) look at company extention length when examining phone login/PIN


------------------------------------------------------------------------
r6665 | raarts | 2006-04-13 00:16:09 +0200 (Thu, 13 Apr 2006) | 2 lines

(cms10719) ringtone support.

------------------------------------------------------------------------
r6554 | raarts | 2006-04-08 00:10:54 +0200 (Sat, 08 Apr 2006) | 16 lines

------------------------------------------------------------------------
r6550 | raarts | 2006-04-07 17:02:40 +0200 (Fri, 07 Apr 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) also disable callwaiting when max concurrent calls < 2

------------------------------------------------------------------------
r6552 | raarts | 2006-04-07 17:43:30 +0200 (Fri, 07 Apr 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) Now look at both values for max concurrent calls (in the phoneline and
on the phone). The smallest one wins.


------------------------------------------------------------------------
r6553 | raarts | 2006-04-08 00:10:09 +0200 (Sat, 08 Apr 2006) | 2 lines

(bugfix) fix erroneous use of calleridnum

------------------------------------------------------------------------
r6485 | raarts | 2006-04-05 21:20:18 +0200 (Wed, 05 Apr 2006) | 2 lines

Also regenerate dhcpd.conf when building, subnet, network, host etc changed.

------------------------------------------------------------------------
r6483 | nnrelman | 2006-04-05 17:29:11 +0200 (Wed, 05 Apr 2006) | 1 line

Changed Copyright into License, fc5 does not support it
------------------------------------------------------------------------
r6437 | raarts | 2006-04-02 14:56:59 +0200 (Sun, 02 Apr 2006) | 2 lines

Bumped version number to 1.2

------------------------------------------------------------------------
r6433 | raarts | 2006-03-31 23:08:42 +0200 (Fri, 31 Mar 2006) | 55 lines

------------------------------------------------------------------------
r6368 | raarts | 2006-03-27 23:31:59 +0200 (Mon, 27 Mar 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/stdastium.inc

(bugfix) Some telcos return PRI Cause code CHANNEL UNAVAILABLE when a line is busy.
Weird. So now we indicate a busy signal when that happens.

------------------------------------------------------------------------
r6369 | raarts | 2006-03-28 09:43:30 +0200 (Tue, 28 Mar 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) Use European start/stop dates for Daylight Savings Time in
stead of the USA ones.

------------------------------------------------------------------------
r6370 | raarts | 2006-03-28 09:56:23 +0200 (Tue, 28 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) fix syntax error.

------------------------------------------------------------------------
r6407 | raarts | 2006-03-29 22:45:25 +0200 (Wed, 29 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) initial support for Polycom 301/501/601 models.

------------------------------------------------------------------------
r6416 | raarts | 2006-03-30 18:09:22 +0200 (Thu, 30 Mar 2006) | 4 lines
Changed paths:
   M /astcfgd/branches/1.0/src/check-sync.pl

(bugfix) Enabled sipura reboot support. Was missing before.
Use the current ip address field from the database to talk to a phone.
Log reboots in the system log.

------------------------------------------------------------------------
r6419 | raarts | 2006-03-31 14:57:34 +0200 (Fri, 31 Mar 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) Fix for 7905/7912: These phones ask for other filenames then we thought.
Fixed. Also disabled VLAN en CDP discovery for these phones.

------------------------------------------------------------------------
r6429 | raarts | 2006-03-31 22:41:10 +0200 (Fri, 31 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/stdastium.inc

(cms11085) call forward when logged off. Dialplan support.


------------------------------------------------------------------------
r6420 | raarts | 2006-03-31 15:33:01 +0200 (Fri, 31 Mar 2006) | 2 lines

(bugfix) only generate networks/subnets/hosts that are not deleted.

------------------------------------------------------------------------
r6398 | wijnand | 2006-03-29 10:28:21 +0200 (Wed, 29 Mar 2006) | 2 lines

Oops, I should fix the comments when copy/pasting :-)

------------------------------------------------------------------------
r6388 | wijnand | 2006-03-28 18:39:22 +0200 (Tue, 28 Mar 2006) | 2 lines

make sure hylafax has a setup.cache when starting.

------------------------------------------------------------------------
r6378 | wijnand | 2006-03-28 14:27:47 +0200 (Tue, 28 Mar 2006) | 2 lines

Oops, forgot to add fax.php

------------------------------------------------------------------------
r6377 | wijnand | 2006-03-28 14:12:07 +0200 (Tue, 28 Mar 2006) | 3 lines

Allow creation of a hosts.hfaxd.local file. The contents will be included in the generated hosts.hfaxd
We need this because we need to allow the netland-03 to fax without authentication.

------------------------------------------------------------------------
r6373 | wijnand | 2006-03-28 12:37:18 +0200 (Tue, 28 Mar 2006) | 2 lines

Fix the parse error from r6370 in current too

------------------------------------------------------------------------
r6367 | wijnand | 2006-03-27 17:28:35 +0200 (Mon, 27 Mar 2006) | 2 lines

Do some things in a nicer order, it seems to prevent asterisk crashes.

------------------------------------------------------------------------
r6360 | raarts | 2006-03-26 00:47:59 +0100 (Sun, 26 Mar 2006) | 95 lines

------------------------------------------------------------------------
r6312 | raarts | 2006-03-18 10:47:03 +0100 (Sat, 18 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Only generate defaultip= line when we actually HAVE an ip address

------------------------------------------------------------------------
r6315 | raarts | 2006-03-18 18:07:26 +0100 (Sat, 18 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/ntpd.php

(bugfix) generate correct ntpd.conf looking at register_ip variable

------------------------------------------------------------------------
r6316 | raarts | 2006-03-18 18:12:59 +0100 (Sat, 18 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) define the extension length as variable extlen for every phone

------------------------------------------------------------------------
r6320 | raarts | 2006-03-19 21:27:36 +0100 (Sun, 19 Mar 2006) | 8 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) set proper defaults for the functionkeys on swissvoice phones
as follows:

F1 FUNCTION HOLD
F2 FUNCTION TRANSFER
F3 FUNCTION VM
F4 FUNCTION DND

------------------------------------------------------------------------
r6321 | raarts | 2006-03-19 21:32:39 +0100 (Sun, 19 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

Make the third button on a Swissvoice the MUTE button.

------------------------------------------------------------------------
r6339 | raarts | 2006-03-22 21:07:17 +0100 (Wed, 22 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/Makefile.am
   M /astcfgd/branches/1.0/src/asterisk.php
   A /astcfgd/branches/1.0/src/astium_set_company_key

(bugfix) Generate a private/public keypair using astgenkey when a new company is created

------------------------------------------------------------------------
r6342 | raarts | 2006-03-22 21:15:08 +0100 (Wed, 22 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/redhat/astcfgd.spec.in

Forgot to add script to redhat spec file.

------------------------------------------------------------------------
r6344 | raarts | 2006-03-22 22:17:02 +0100 (Wed, 22 Mar 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/astium_set_company_key

(bugfix) Modified key generation script to also generate .pub en .key
files in /var/lib/asteris/keys

------------------------------------------------------------------------
r6346 | raarts | 2006-03-23 17:03:27 +0100 (Thu, 23 Mar 2006) | 4 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) the default target for outbound calls should also include
phonenumbers starting with an asterisk. Or we would not be able to
send *21 etc codes to the telco.

------------------------------------------------------------------------
r6347 | raarts | 2006-03-23 17:08:36 +0100 (Thu, 23 Mar 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(feature) You can now send *any* code to the telco by prefixing
(escaping) it with '**'.

------------------------------------------------------------------------
r6350 | ivar | 2006-03-24 10:51:29 +0100 (Fri, 24 Mar 2006) | 1 line
Changed paths:
   M /astcfgd/branches/1.0/src/conferences

(NeoNova 12564) bugfix: fix conference script
------------------------------------------------------------------------
r6351 | ivar | 2006-03-24 11:37:54 +0100 (Fri, 24 Mar 2006) | 1 line
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(NeoNova 12564) when company is modified, re-generate meetme.conf too.

------------------------------------------------------------------------
r6354 | raarts | 2006-03-24 23:08:47 +0100 (Fri, 24 Mar 2006) | 3 lines

Sometimes you want to answer the line before entering a queue (or you don't hear moh).
sometimes (for example when the call is initiated from a callfile) you don't.

------------------------------------------------------------------------
r6334 | ivar | 2006-03-21 20:55:44 +0100 (Tue, 21 Mar 2006) | 1 line

astcfgd, bugfix: when giving ivr a internal number, include ivr in 'from' part of query
------------------------------------------------------------------------
r6333 | wijnand | 2006-03-20 18:35:16 +0100 (Mon, 20 Mar 2006) | 6 lines

(NeoNova 12488) 
Remove telinit hack again, replace it with a new one to prevent the inittab to reload 64 times.
Cut iaxmodem some slack, wait 10 seconds before restarting the other fax stuff, sometimes iaxmodem is slow with the first registration.

While there, set the registration time for the iaxmodem devices to 60 seconds, hoping to prevent asterisk warnings.

------------------------------------------------------------------------
r6330 | ivar | 2006-03-20 16:08:53 +0100 (Mon, 20 Mar 2006) | 1 line

bug in asterisk.php: queue.name was not selected in query, but it was used in generated dialplan. Calling queues internally didn't work.
------------------------------------------------------------------------
r6324 | wijnand | 2006-03-20 11:07:42 +0100 (Mon, 20 Mar 2006) | 2 lines

(NeoNova 12488) software fax support in astcfgd

------------------------------------------------------------------------
r6301 | wijnand | 2006-03-16 18:57:39 +0100 (Thu, 16 Mar 2006) | 3 lines

Abuse the services module to be able to reload inittab
I think it's a nice but ugly hack.

------------------------------------------------------------------------
r6273 | ivar | 2006-03-14 16:38:29 +0100 (Tue, 14 Mar 2006) | 2 lines

svn merge

------------------------------------------------------------------------
r6262 | raarts | 2006-03-13 17:54:41 +0100 (Mon, 13 Mar 2006) | 2 lines

Forgot to include company-wide DHCP options. Noticed by Jan Willem.

------------------------------------------------------------------------
r6260 | raarts | 2006-03-13 16:20:31 +0100 (Mon, 13 Mar 2006) | 2 lines

Redirect all output to zero, because it crashes astcfgd

------------------------------------------------------------------------
r6259 | raarts | 2006-03-13 16:04:04 +0100 (Mon, 13 Mar 2006) | 2 lines

Write a message in the logfile when a phone is rebooted.

------------------------------------------------------------------------
r6255 | raarts | 2006-03-13 13:08:21 +0100 (Mon, 13 Mar 2006) | 2 lines

feature: resetting phone from the web now works as advertised.

------------------------------------------------------------------------
r6253 | ivar | 2006-03-13 12:45:39 +0100 (Mon, 13 Mar 2006) | 20 lines


svnmerge: checking latest revision of http://svn.neonova.nl/svn/private/astcfgd/branches/1.0
svnmerge: latest revision of http://svn.neonova.nl/svn/private/astcfgd/branches/1.0 is 6252
svn log --incremental -v -r 6147:6252 http://svn.neonova.nl/svn/private/astcfgd/branches/1.0
------------------------------------------------------------------------
r6228 | raarts | 2006-03-12 09:41:47 +0100 (Sun, 12 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php
   M /astcfgd/branches/1.0/src/zaptel.php

Added support for the TDM2400P

------------------------------------------------------------------------
r6250 | raarts | 2006-03-13 12:18:30 +0100 (Mon, 13 Mar 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/tftp.php

Fixed duplicate registration type for Polycom phones.


------------------------------------------------------------------------
r6249 | raarts | 2006-03-13 12:04:03 +0100 (Mon, 13 Mar 2006) | 2 lines

(bugfix) Get the phone's ip address from the correct database field

------------------------------------------------------------------------
r6248 | ivar | 2006-03-13 11:48:25 +0100 (Mon, 13 Mar 2006) | 18 lines

svnmerge and resolved some conflicts in asterisk.php

svnmerge: checking latest revision of http://svn.neonova.nl/svn/private/astcfgd/ivar/9911
svnmerge: latest revision of http://svn.neonova.nl/svn/private/astcfgd/ivar/9911 is 6246
svn log --incremental -v -r 5623:6246 http://svn.neonova.nl/svn/private/astcfgd/ivar/9911
------------------------------------------------------------------------
r5629 | ivar | 2006-01-24 10:00:05 +0100 (Tue, 24 Jan 2006) | 1 line
Changed paths:
   M /astcfgd/ivar/9911/src/asterisk.php

(NeoNova 9911) Changes required for ivr menu's to get an extension
------------------------------------------------------------------------
r5739 | ivar | 2006-01-31 15:40:06 +0100 (Tue, 31 Jan 2006) | 1 line
Changed paths:
   M /astcfgd/ivar/9911

need to commit svnmerge init?

------------------------------------------------------------------------
r6247 | ivar | 2006-03-13 11:37:02 +0100 (Mon, 13 Mar 2006) | 1 line

must commit 'svnmerge init' first?
------------------------------------------------------------------------
r6246 | ivar | 2006-03-13 11:31:23 +0100 (Mon, 13 Mar 2006) | 3 lines

not sure what is in this commit...


------------------------------------------------------------------------
r6245 | raarts | 2006-03-13 10:55:37 +0100 (Mon, 13 Mar 2006) | 2 lines

Only set defaultip= when we actuallly have a default ip address

------------------------------------------------------------------------
r6244 | raarts | 2006-03-13 10:54:18 +0100 (Mon, 13 Mar 2006) | 2 lines

Don't skip phones with an empty IP address

------------------------------------------------------------------------
r6198 | raarts | 2006-03-11 08:39:00 +0100 (Sat, 11 Mar 2006) | 4 lines

(feature) DHCP Management. One can now manage buildings and the networks,
IP subnets, and hosts. Also its possible to assign a phone to a network
so the system will choose a working IP address/gateway for the phone.

------------------------------------------------------------------------
r6190 | raarts | 2006-03-10 16:16:55 +0100 (Fri, 10 Mar 2006) | 2 lines

(feature) implemented call-me-now by entering a phone number on a webpage.

------------------------------------------------------------------------
r6147 | ivar | 2006-03-08 10:04:11 +0100 (Wed, 08 Mar 2006) | 18 lines


svnmerge: checking latest revision of http://svn.neonova.nl/svn/private/astcfgd/branches/1.0
svnmerge: latest revision of http://svn.neonova.nl/svn/private/astcfgd/branches/1.0 is 6146
svn log --incremental -v -r 6095:6146 http://svn.neonova.nl/svn/private/astcfgd/branches/1.0
------------------------------------------------------------------------
r6134 | ivar | 2006-03-07 16:44:44 +0100 (Tue, 07 Mar 2006) | 1 line
Changed paths:
   M /astcfgd/branches/1.0/src/astium-functions.php
   M /astcfgd/branches/1.0/src/check-sync.pl
   M /astcfgd/branches/1.0/src/dhcp.php
   M /astcfgd/branches/1.0/src/firewall.php
   M /astcfgd/branches/1.0/src/network.php
   M /astcfgd/branches/1.0/src/ntpd.php
   M /astcfgd/branches/1.0/src/postfix.php
   M /astcfgd/branches/1.0/src/tftp.php

(NeoNova 12067) remove interface admin from web - part 2

------------------------------------------------------------------------
r6110 | ivar | 2006-03-03 15:18:11 +0100 (Fri, 03 Mar 2006) | 4 lines


Initialized merge tracking via svnmerge with revisions 1-5622 from
http://svn.neonova.nl/svn/private/astcfgd/ivar/9911

------------------------------------------------------------------------
r6095 | ivar | 2006-03-02 10:10:58 +0100 (Thu, 02 Mar 2006) | 4 lines


Merged revisions 5991-6094 via svnmerge from
/astcfgd/branches/1.0

------------------------------------------------------------------------
r6077 | raarts | 2006-03-01 12:53:56 +0100 (Wed, 01 Mar 2006) | 2 lines

(bugfix) Do NOT remove ALL grandstream files, when regenerating the config for only ONE phone.

------------------------------------------------------------------------
r6075 | raarts | 2006-03-01 12:17:17 +0100 (Wed, 01 Mar 2006) | 2 lines

Order the extensions in the dialplan (the hint part) by extension

------------------------------------------------------------------------
r6058 | raarts | 2006-02-25 23:45:02 +0100 (Sat, 25 Feb 2006) | 2 lines

(bugfix) give call-recording and speech generation lower CPU priority

------------------------------------------------------------------------
r6054 | ivar | 2006-02-24 16:59:31 +0100 (Fri, 24 Feb 2006) | 1 line

create company directory for mp3 files
------------------------------------------------------------------------
r5991 | raarts | 2006-02-23 00:03:16 +0100 (Thu, 23 Feb 2006) | 101 lines

------------------------------------------------------------------------
r5894 | raarts | 2006-02-14 10:20:22 +0100 (Tue, 14 Feb 2006) | 8 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix). Generated the correct confiuration to make the lights
on SNOM phones working. This means that a light on your
phone can show the fact that a colleague is on the phone
(or her phone rings, which will show a blinking LED).

The only thing we now need is a way to configure these buttons
from the web. These are not fastdials, but monitoring lights.

------------------------------------------------------------------------
r5913 | raarts | 2006-02-14 17:24:15 +0100 (Tue, 14 Feb 2006) | 6 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php
   M /astcfgd/branches/1.0/src/stdastium.inc

(bugfix cms:11600) Afsluittoon in Astium.
If a number is unallocated (out of service), now 
generate the proper tones to indicate that, and tell the
user. Busy is also better handled now, because
it will not be retried using a fallback route.

------------------------------------------------------------------------
r5922 | raarts | 2006-02-16 11:11:44 +0100 (Thu, 16 Feb 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/redhat/astcfgd.spec.in
   M /astcfgd/branches/1.0/src/Makefile.am
   A /astcfgd/branches/1.0/src/meetme.conf
   M /astcfgd/branches/1.0/src/stdastium.inc

Added meetme.conf to astcfgd package where it belongs.

------------------------------------------------------------------------
r5923 | raarts | 2006-02-16 11:14:46 +0100 (Thu, 16 Feb 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) set the language to the default company language
for incoming calls routed into conference boxes.

------------------------------------------------------------------------
r5925 | raarts | 2006-02-16 12:21:28 +0100 (Thu, 16 Feb 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) Setting callerid values changes in asterisk 1.2.x,
so use the new way from now on.

------------------------------------------------------------------------
r5927 | raarts | 2006-02-16 12:39:00 +0100 (Thu, 16 Feb 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) if the outgoing callerid is set to 0 (zero) suppress
the callerid on the outgoing line.

------------------------------------------------------------------------
r5958 | raarts | 2006-02-19 15:40:30 +0100 (Sun, 19 Feb 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(feature) callforward-not-answer-after is now implemented in the backend.

------------------------------------------------------------------------
r5962 | raarts | 2006-02-19 16:36:06 +0100 (Sun, 19 Feb 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(Bugfix) regenerate *all* configuration files, when a 'module asterisk'
activation is done, or when a company is added.

------------------------------------------------------------------------
r5964 | raarts | 2006-02-20 11:40:20 +0100 (Mon, 20 Feb 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

(bugfix) Fix voicemail button for older SNOM phones.

------------------------------------------------------------------------
r5973 | raarts | 2006-02-21 17:26:31 +0100 (Tue, 21 Feb 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/stdastium.inc

Send User Events when CallForward is enabled/disabled, so
the Personal Switchboard can display that.

------------------------------------------------------------------------
r5989 | raarts | 2006-02-22 23:55:51 +0100 (Wed, 22 Feb 2006) | 5 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php
   M /astcfgd/branches/1.0/src/tftp.php

(bugfix) Fixed generating empty queue entries, even though none
were being defined in the GUI. 
(bugfix) Set the language on all incoming lines to the 
default company language.


------------------------------------------------------------------------
r5990 | raarts | 2006-02-22 23:58:44 +0100 (Wed, 22 Feb 2006) | 2 lines

Ran svnmerge init on this tree.

------------------------------------------------------------------------
r5954 | ivar | 2006-02-17 15:35:27 +0100 (Fri, 17 Feb 2006) | 1 line

svnmerge init again
------------------------------------------------------------------------
r5952 | ivar | 2006-02-17 15:33:17 +0100 (Fri, 17 Feb 2006) | 4 lines


Merged revisions 5874-5946 via svnmerge from
/astcfgd/branches/1.0

------------------------------------------------------------------------
r5944 | ivar | 2006-02-17 12:32:31 +0100 (Fri, 17 Feb 2006) | 1 line

init merge
------------------------------------------------------------------------
r5940 | ivar | 2006-02-17 12:16:44 +0100 (Fri, 17 Feb 2006) | 1 line

implement generating configuration files for music on hold.
------------------------------------------------------------------------
r5934 | raarts | 2006-02-16 16:59:57 +0100 (Thu, 16 Feb 2006) | 6 lines

(feature: paging) Added page macro. This enables dialing
another phone and have it auto-pickup on handsfree.
Tested on SNOM. Does not work on firmware 3.56q.
This feature is not fully ready yet, just doing some 
preparing work.

------------------------------------------------------------------------
r5890 | ivar | 2006-02-14 09:40:54 +0100 (Tue, 14 Feb 2006) | 1 line

(NeoNova 10962) new feature: enable/disable firewall with checkbox
------------------------------------------------------------------------
r5885 | ivar | 2006-02-13 17:22:13 +0100 (Mon, 13 Feb 2006) | 1 line

copy 1.0 branch of astcfgd to current branch
------------------------------------------------------------------------
r5870 | raarts | 2006-02-10 16:58:06 +0100 (Fri, 10 Feb 2006) | 3 lines

bugfix: when nobody is logged in to a phone: *DO* play
the greeting message.

------------------------------------------------------------------------
r5845 | ivar | 2006-02-07 14:22:51 +0100 (Tue, 07 Feb 2006) | 1 line

(NeoNova 10169) bugfix: give all trunks_'s a 'status' field, and change that to 'deleted' when deleting a trunk_* record.
------------------------------------------------------------------------
r5839 | raarts | 2006-02-07 12:18:58 +0100 (Tue, 07 Feb 2006) | 3 lines

(bugfix, reported on the phone by bruno) 
IAX client authentication is always md5.

------------------------------------------------------------------------
r5834 | ivar | 2006-02-07 11:17:15 +0100 (Tue, 07 Feb 2006) | 1 line

(NeoNova 10169) bugfix: give staticroute a 'status' field, and change that to 'deleted' when deleting a staticroute record.
------------------------------------------------------------------------
r5822 | ivar | 2006-02-06 20:26:08 +0100 (Mon, 06 Feb 2006) | 1 line

(NeoNova 10169) bugfix: give all card_'s a 'status' field, and change that to 'deleted' when deleting a card_* record.
------------------------------------------------------------------------
r5821 | raarts | 2006-02-06 16:39:38 +0100 (Mon, 06 Feb 2006) | 2 lines

(bugfix cms11625) Now generates proper contexts for analog phonelines.

------------------------------------------------------------------------
r5782 | raarts | 2006-02-05 22:55:48 +0100 (Sun, 05 Feb 2006) | 8 lines

Prepare for High Availability. Currently HA is determined by 
the softlink /etc/asterisk/extensions.conf. When Astium is master
this link points to /etc/asterisk/extensions.conf.master.
When Astium is running as slave, this link points to a file named
/etc/asterisk.extensions,conf.slave. This way going from master to
slave involves only pointing a softlink to another file, and
giving an asterisk reload.

------------------------------------------------------------------------
r5781 | raarts | 2006-02-03 15:45:55 +0100 (Fri, 03 Feb 2006) | 2 lines

Add support for SNOM 320 to the background deamon

------------------------------------------------------------------------
r5765 | ivar | 2006-02-01 17:21:04 +0100 (Wed, 01 Feb 2006) | 1 line

(NeoNova 10169) bugfix: give soundfile a 'status' field, and change that to 'deleted' when deleting a soundfile record.
------------------------------------------------------------------------
r5758 | ivar | 2006-02-01 12:08:29 +0100 (Wed, 01 Feb 2006) | 1 line

(NeoNova 10169) bugfix: give conference a 'status' field, and change that to 'deleted' when deleting a conference record.
------------------------------------------------------------------------
r5753 | ivar | 2006-02-01 11:33:01 +0100 (Wed, 01 Feb 2006) | 1 line

(NeoNova 10169) bugfix: give queue a 'status' field, and change that to 'deleted' when deleting a queue record.
------------------------------------------------------------------------
r5743 | raarts | 2006-01-31 21:55:43 +0100 (Tue, 31 Jan 2006) | 3 lines

Set default language for Queues in the extension list
Remove inboundid from query, it was not used.

------------------------------------------------------------------------
r5741 | ivar | 2006-01-31 16:26:03 +0100 (Tue, 31 Jan 2006) | 1 line

move creation of sounds directory to astcfgd
------------------------------------------------------------------------
r5732 | ivar | 2006-01-31 15:12:34 +0100 (Tue, 31 Jan 2006) | 1 line

(NeoNova 10169) bugfix: give ivr a 'status' field, and change that to 'deleted' when deleting an ivr record.
------------------------------------------------------------------------
r5721 | ivar | 2006-01-30 17:06:02 +0100 (Mon, 30 Jan 2006) | 1 line

(bugfix) use table prefix when accessing status field
------------------------------------------------------------------------
r5717 | ivar | 2006-01-30 16:31:31 +0100 (Mon, 30 Jan 2006) | 1 line

(NeoNova 10169) bugfix: give inbound a 'status' field, and change that to 'deleted' when deleting an inbound record.
------------------------------------------------------------------------
r5699 | ivar | 2006-01-30 11:38:15 +0100 (Mon, 30 Jan 2006) | 1 line

(NeoNova 10169) bugfix: give callrouting a 'status' field, and change that to 'deleted' when deleting a callrouting record.
------------------------------------------------------------------------
r5696 | raarts | 2006-01-28 00:47:25 +0100 (Sat, 28 Jan 2006) | 3 lines

Try to set the proper accountcode for incoming calls as well,
so Ivar gets a much easier job displaying the call statistics.

------------------------------------------------------------------------
r5695 | raarts | 2006-01-28 00:07:34 +0100 (Sat, 28 Jan 2006) | 8 lines

(bugfix) When a call is forwarded, now two CDR's are created:
- one for the original call, this contains the original desctination and
  a comment in the userfield which shows where it was forwarded to, 
- one for the owner of the forwarding phone, showing original source
  and a comment in the userfield which shows by which extension it was 
  forwarded.
Multiple forwards generate multiple CDR's and billing info is preserved. 

------------------------------------------------------------------------
r5684 | raarts | 2006-01-27 11:21:07 +0100 (Fri, 27 Jan 2006) | 4 lines

(bugfix, found by me) Forwarded calls should also be written
to the CDR table. Fixed. Now it's up to Ivar to make them show
up in the management web.

------------------------------------------------------------------------
r5683 | raarts | 2006-01-27 10:53:06 +0100 (Fri, 27 Jan 2006) | 3 lines

(bugfix: noticed by me) regenerate voicemail.conf when 
adding a company or regenerating all asterisk configs.

------------------------------------------------------------------------
r5673 | raarts | 2006-01-26 10:37:20 +0100 (Thu, 26 Jan 2006) | 2 lines

(bugfix conferences): did not issue the query correctly to create a confbox.

------------------------------------------------------------------------
r5672 | raarts | 2006-01-26 10:17:41 +0100 (Thu, 26 Jan 2006) | 3 lines

(bugfix made while on the phone with eurid). Forgot to skip
deleted phones while generating zapata.conf.

------------------------------------------------------------------------
r5671 | raarts | 2006-01-26 00:27:13 +0100 (Thu, 26 Jan 2006) | 4 lines

(housekeeping) move some files form the asterisk rpm to this one
(feature cms:9691) Conference creation by the web.
(bugfix) set the default telnet password for cisco phones to 'astium'.

------------------------------------------------------------------------
r5606 | raarts | 2006-01-20 23:54:39 +0100 (Fri, 20 Jan 2006) | 2 lines

Use the new query constructed by Ivar to create conferences.

------------------------------------------------------------------------
r5603 | raarts | 2006-01-20 14:26:39 +0100 (Fri, 20 Jan 2006) | 2 lines

(bugfix) Overriding callerid did not work since the 1.0 version. Fixed.

------------------------------------------------------------------------
r5586 | raarts | 2006-01-18 17:40:33 +0100 (Wed, 18 Jan 2006) | 2 lines

Also added extensions.conf to astcfgd.

------------------------------------------------------------------------
r5585 | raarts | 2006-01-18 17:37:00 +0100 (Wed, 18 Jan 2006) | 3 lines

Moved the file stdastium.inc to the astcfgd package.
This is where it really belongs.

------------------------------------------------------------------------
r5570 | ivar | 2006-01-18 09:31:33 +0100 (Wed, 18 Jan 2006) | 1 line

(bugfix) properly create config for Cisco 7912
------------------------------------------------------------------------
r5527 | raarts | 2006-01-16 13:49:59 +0100 (Mon, 16 Jan 2006) | 1 line

Promote the trunk (based on asterisk 1.2) to official 1.0
------------------------------------------------------------------------
r5525 | raarts | 2006-01-16 13:49:24 +0100 (Mon, 16 Jan 2006) | 25 lines

------------------------------------------------------------------------
r5425 | raarts | 2006-01-06 11:22:01 +0100 (Fri, 06 Jan 2006) | 2 lines
Changed paths:
   M /astcfgd/branches/1.0/src/upwatch.php

Fixed typo and missing declarations in generating upwatch configuration.

------------------------------------------------------------------------
r5460 | raarts | 2006-01-09 14:44:50 +0100 (Mon, 09 Jan 2006) | 4 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Use joins in a statement instead of where ....
This way there is more chance of analog phonelines
getting configured properly.

------------------------------------------------------------------------
r5466 | raarts | 2006-01-10 09:44:24 +0100 (Tue, 10 Jan 2006) | 3 lines
Changed paths:
   M /astcfgd/branches/1.0/src/asterisk.php

Fix bug where a zap trunk is defined for a company, but no
channels exist within the channelgroup given.


------------------------------------------------------------------------
r5465 | raarts | 2006-01-10 09:30:28 +0100 (Tue, 10 Jan 2006) | 3 lines

Fix bug where a zap trunk is defined for a company, but no
channels exist within the channelgroup given.

------------------------------------------------------------------------
r5461 | raarts | 2006-01-09 14:45:58 +0100 (Mon, 09 Jan 2006) | 4 lines

Use joins in a statement instead of where ....
This way there is more chance of analog phonelines
getting configured properly.

------------------------------------------------------------------------
r5422 | raarts | 2006-01-02 13:46:48 +0100 (Mon, 02 Jan 2006) | 5 lines

Tell Polycom phones which timeserver they should be using.
Do the same for Cisco.
Small change in generating the filename for Sipura, because
we might also support Linksys stuff.

------------------------------------------------------------------------
r5415 | raarts | 2005-12-23 16:35:53 +0100 (Fri, 23 Dec 2005) | 2 lines

Small change to Sipura provisioning

------------------------------------------------------------------------
r5413 | raarts | 2005-12-23 14:42:15 +0100 (Fri, 23 Dec 2005) | 2 lines

Create pre-provisioning files for Sipura phones.

------------------------------------------------------------------------
r5375 | raarts | 2005-12-16 15:41:42 +0100 (Fri, 16 Dec 2005) | 2 lines

Support for Polycom SoundStation IP 4000 in the backend.

------------------------------------------------------------------------
r5333 | raarts | 2005-12-13 16:24:57 +0100 (Tue, 13 Dec 2005) | 2 lines

BUG: Enable/disable call forward according to the setting in the web.

------------------------------------------------------------------------
r5331 | raarts | 2005-12-13 15:49:09 +0100 (Tue, 13 Dec 2005) | 3 lines

asterisk 1.2 has a different layout for 'show peer xxxxx' output.
Because of this, phones did not reboot. This fix makes that work again.

------------------------------------------------------------------------
r5324 | raarts | 2005-12-13 11:31:26 +0100 (Tue, 13 Dec 2005) | 3 lines

Merged revisions 5271-5323 via svnmerge from
/astcfgd/branches/1.0

------------------------------------------------------------------------
r5323 | raarts | 2005-12-13 11:29:42 +0100 (Tue, 13 Dec 2005) | 2 lines

Used svnmerge on this tree

------------------------------------------------------------------------
r5317 | raarts | 2005-12-12 09:37:33 +0100 (Mon, 12 Dec 2005) | 2 lines

Fix wrong placement of quote.

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r5277 | raarts | 2005-12-07 10:15:08 +0100 (Wed, 07 Dec 2005) | 2 lines

Fix parse error when *xx commands are dialed.

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r5270 | raarts | 2005-12-06 17:02:01 +0100 (Tue, 06 Dec 2005) | 2 lines

Merged all fixes made to 1.0 into the current trunk version.

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r5243 | raarts | 2005-12-01 14:43:10 +0100 (Thu, 01 Dec 2005) | 3 lines

Use .inf files for swissvoice phones. They seem to like that.
Fixed some parse errors in the dialplan.

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r5188 | raarts | 2005-11-28 14:01:01 +0100 (Mon, 28 Nov 2005) | 2 lines

Do not display connects to the manager interface in the CLI.

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r5187 | raarts | 2005-11-28 13:59:20 +0100 (Mon, 28 Nov 2005) | 1 line

copy branches/1.0 to trunk
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r5184 | wijnand | 2005-11-28 12:47:58 +0100 (Mon, 28 Nov 2005) | 4 lines

We have to wait 1 minute before shutdown or reboot. We have to be absolutely sure astcfgd marks the shutdown and reboot activate records as done. Otherwise reboot cycles can occur.

Fixes CMS 10062

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r5149 | raarts | 2005-11-24 23:47:58 +0100 (Thu, 24 Nov 2005) | 5 lines

Also regenerate if a te110p is modified.
Fixed speling error
Fixed autologout so that it also reboots the phones involved.
Fix bug in checksync if the username is not found.

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r5137 | raarts | 2005-11-23 14:43:43 +0100 (Wed, 23 Nov 2005) | 2 lines

Enlarged the echotraining time to 800 ms.

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r5121 | raarts | 2005-11-22 12:25:39 +0100 (Tue, 22 Nov 2005) | 2 lines

Fixed typo in generating jumps to forbidden extension numbers.

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r5119 | raarts | 2005-11-22 11:29:39 +0100 (Tue, 22 Nov 2005) | 4 lines

Instead of leaving out the numbers you are not allowed to dial,
we now specifically send these numbers to a context
stating 'this extensionnumber is invalid'.

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r5072 | raarts | 2005-11-19 20:01:13 +0100 (Sat, 19 Nov 2005) | 2 lines

Small bugfix: skip the default target

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r5065 | raarts | 2005-11-18 23:09:14 +0100 (Fri, 18 Nov 2005) | 4 lines

Authorize outgoing connections based on the new costclass variable
in the callrouting table. All this was started with a bugreport from
Scarlet Curacao (#9462).

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r5046 | raarts | 2005-11-18 09:21:01 +0100 (Fri, 18 Nov 2005) | 2 lines

Must generate queues.conf as well, when a user logs in!

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r5045 | raarts | 2005-11-18 00:10:36 +0100 (Fri, 18 Nov 2005) | 2 lines

Ensure that on Polycom phones the volume is not reset to default after a call.

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r5043 | raarts | 2005-11-17 16:55:14 +0100 (Thu, 17 Nov 2005) | 3 lines

First look at the phonereboot astglobal before trying to reboot a phone.
You can override this with '-f' on the commandline.

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r5036 | raarts | 2005-11-17 15:54:29 +0100 (Thu, 17 Nov 2005) | 5 lines

A bit of a hack, until we have triggers really working.
Only generate the phonefiles that *need* to be regenerated
in case of user login/logout. In all other cases regenerate all,
as we did before.

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r5035 | raarts | 2005-11-17 15:07:06 +0100 (Thu, 17 Nov 2005) | 2 lines

Prepare for unifying SQL queries for phones.

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r5034 | raarts | 2005-11-17 11:47:50 +0100 (Thu, 17 Nov 2005) | 3 lines

Move the determination of our ip address to one spot.
Saves at least one SQL query per phone.

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r5033 | raarts | 2005-11-17 11:34:25 +0100 (Thu, 17 Nov 2005) | 3 lines

When creating .inc files, the same query was used over and over again.
Fixed.

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r5032 | raarts | 2005-11-17 11:25:54 +0100 (Thu, 17 Nov 2005) | 4 lines

Took another query out of a loop, and reused the
results instead. Performance Optimisation for the 
background dialplan generator.

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r5031 | raarts | 2005-11-17 11:19:53 +0100 (Thu, 17 Nov 2005) | 3 lines

Took a query out of a loop, to reduce the number of times it is performed
Save and reuse the results instead.

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r5030 | raarts | 2005-11-17 11:07:05 +0100 (Thu, 17 Nov 2005) | 2 lines

Folded two queries into one, and removed one superfluous query.

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r5029 | raarts | 2005-11-17 11:00:29 +0100 (Thu, 17 Nov 2005) | 2 lines

Ensure that ALL queries are being printed when debug > 1.

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r5027 | raarts | 2005-11-16 17:07:59 +0100 (Wed, 16 Nov 2005) | 2 lines

Erase unused phonelines on a Cisco 7940/60.

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r5022 | raarts | 2005-11-16 12:50:02 +0100 (Wed, 16 Nov 2005) | 2 lines

Also regenerate when a channel is modified.

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r5020 | raarts | 2005-11-15 17:42:33 +0100 (Tue, 15 Nov 2005) | 2 lines

Generate correct filename for cisco 7905/7912

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r5018 | raarts | 2005-11-15 17:26:37 +0100 (Tue, 15 Nov 2005) | 2 lines

No quotes around ip addresses are allowed .

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r5013 | wijnand | 2005-11-15 15:13:45 +0100 (Tue, 15 Nov 2005) | 7 lines

Remove the
ln: `/etc/cron.daily/autologout.pl': File exists
error: %post(astcfgd-1.0-5011) scriptlet failed, exit status 1

error when upgrading astcfgd.


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r5011 | raarts | 2005-11-15 12:17:09 +0100 (Tue, 15 Nov 2005) | 2 lines

Add timeout. Cannot wait forever...

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r5007 | wijnand | 2005-11-15 09:45:35 +0100 (Tue, 15 Nov 2005) | 4 lines

Only one at job is needed for a astcfgd restart

I highly doubt it is really needed, you only want to restart it when it is not running. It cannot restart itself when not running.

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r5005 | raarts | 2005-11-14 22:17:36 +0100 (Mon, 14 Nov 2005) | 2 lines

Small fixes for restarting ourselves.

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r4996 | raarts | 2005-11-14 12:50:32 +0100 (Mon, 14 Nov 2005) | 5 lines

When calling on of your own inbound lines, don't allocate
a real line, but take a shortcut in the dialplan.
Used for supporting connecting Skype users to an incoming
IVR menu.

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r4995 | raarts | 2005-11-14 12:49:07 +0100 (Mon, 14 Nov 2005) | 3 lines

Can now reboot multiple phones in one run.
Fix reboot when phone not reachable.

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r4993 | wijnand | 2005-11-14 12:35:12 +0100 (Mon, 14 Nov 2005) | 12 lines

Astcfgd can now restart services and reboot the astium.

action is always module
type:
system-restart reboots the astium
system-shutdown halts the astium
service-servicename-restart restarts the service with name servicename
service-servicename-reload reloads the service with name servicename

Please note, I named the module and class services, we already had this class, but I moved it to its own file now since it is getting too big for astium-functions.php. This does not affect restart calls made by other modules (eg racoon.php etc).


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r4975 | raarts | 2005-11-11 14:25:01 +0100 (Fri, 11 Nov 2005) | 4 lines

Add and activate the autologout script. If the astglobals 'autologout'
is set to 'yes', everybody will be logged out of his/her phone
at 4am.

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r4947 | raarts | 2005-11-10 14:26:46 +0100 (Thu, 10 Nov 2005) | 2 lines

Fixes in authenticating using rsa public and private keys for IAX trunks.

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r4936 | raarts | 2005-11-09 16:56:59 +0100 (Wed, 09 Nov 2005) | 2 lines

Generate user Manager event when someone dials a number

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r4919 | raarts | 2005-11-09 09:35:09 +0100 (Wed, 09 Nov 2005) | 2 lines

Generate correct global variables for iax devices.

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r4917 | raarts | 2005-11-08 17:18:41 +0100 (Tue, 08 Nov 2005) | 2 lines

Support IAXy, second stage: provisioning, and generate iax.conf.

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r4915 | raarts | 2005-11-08 15:45:37 +0100 (Tue, 08 Nov 2005) | 2 lines

Should also generate new zapata.conf when company added

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r4912 | wijnand | 2005-11-08 13:59:45 +0100 (Tue, 08 Nov 2005) | 5 lines

Customers want usernames containing dots, postgresql does not like that.
From now on we will convert every dot to _dot_ internally when postgresql is involved.

I don't know if connecting to postgresql for normal users is documented, if so, document this too!

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r4911 | raarts | 2005-11-08 12:44:12 +0100 (Tue, 08 Nov 2005) | 5 lines

Create config file for IAXy, and activate (provision) if needed.
The IAXy is different in that it does not retrieve it configs by
TFTP, but an external utility must *send* the configfile to the IAXy
to activate the configuration.

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r4910 | raarts | 2005-11-08 12:42:44 +0100 (Tue, 08 Nov 2005) | 2 lines

Added script to activate an IAXy ATA.

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r4888 | raarts | 2005-11-07 13:54:47 +0100 (Mon, 07 Nov 2005) | 3 lines

Renamed sip_username to username, sip_secret to secret and sip_dtmfmode to dtmfmode.
This is a preparation for IAX channels, which are added becasue we will support the IAXy.

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r4872 | wijnand | 2005-11-04 16:01:27 +0100 (Fri, 04 Nov 2005) | 3 lines

Quick fix, for every activate record racoon was reconfigured.
Totally useless, and messes up your debug output :-)

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r4866 | raarts | 2005-11-04 13:08:48 +0100 (Fri, 04 Nov 2005) | 2 lines

Used the wrong XML attribute for max calls on Polycom phones.

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r4864 | raarts | 2005-11-04 12:38:23 +0100 (Fri, 04 Nov 2005) | 2 lines

Prevent dupe entries in globals.conf

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r4853 | raarts | 2005-11-03 15:17:46 +0100 (Thu, 03 Nov 2005) | 2 lines

Added the IAXy provisioning tools to the astcfgd rpm.

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r4851 | raarts | 2005-11-03 14:52:45 +0100 (Thu, 03 Nov 2005) | 7 lines

The phones weren't rebooted any more since I started
querying asterisk for the full contact address.
There were two causes: asterisk could not be found in the path
if run by the asterisk user, AND asterisk refused to run
if started by a non-root user.
This patch fixes the first cause.

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r4846 | raarts | 2005-11-03 11:34:30 +0100 (Thu, 03 Nov 2005) | 2 lines

Support for the Cisco 7905/7912 Series Phones

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r4818 | raarts | 2005-10-31 16:28:16 +0100 (Mon, 31 Oct 2005) | 3 lines

At last reboot works for (I hope all) SNOM phones.
And for ciscos I have a new method. Untested yet.

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r4817 | raarts | 2005-10-31 10:03:45 +0100 (Mon, 31 Oct 2005) | 3 lines

Created a ChangeLog file.
Set register timeout for swissvoice to 3600 seconds instead of 1.

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r4804 | raarts | 2005-10-27 13:05:00 +0200 (Thu, 27 Oct 2005) | 3 lines

Functionkey support for swissvoice phones. Are automatically
filled with the first four entries from the personal phone directory.

------------------------------------------------------------------------
r4796 | raarts | 2005-10-27 01:00:56 +0200 (Thu, 27 Oct 2005) | 3 lines

callwaiting setting is changed from yes/no to on/off,
due to the bug in scarlet callcenter.

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r4770 | raarts | 2005-10-20 12:00:59 +0200 (Thu, 20 Oct 2005) | 3 lines

Also generate department id as variable. Need that later for 
determining outgoing callerid.

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r4738 | raarts | 2005-10-18 12:53:35 +0200 (Tue, 18 Oct 2005) | 2 lines

Another quote fix for GotoIF..

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r4737 | raarts | 2005-10-18 12:50:30 +0200 (Tue, 18 Oct 2005) | 4 lines

The GotoIf function is pretty picky. If you compare strings, they
should be enclosed with qoutes on both sides of the comparison operator.
If you compare numbers, do not enclose them in quotes..

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r4736 | raarts | 2005-10-18 12:35:21 +0200 (Tue, 18 Oct 2005) | 2 lines

Fix some GotoIf expressions, needed to use quotes.

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r4710 | raarts | 2005-10-13 16:03:00 +0200 (Thu, 13 Oct 2005) | 2 lines

If a phone overrides its netmask, put that in the generated dhcpd.conf

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r4705 | raarts | 2005-10-13 14:01:23 +0200 (Thu, 13 Oct 2005) | 2 lines

Well, if you change a queue, we should regenerate extensions.conf and inbound.conf as well I think

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r4703 | raarts | 2005-10-13 13:14:04 +0200 (Thu, 13 Oct 2005) | 3 lines

We have multiple lines per user. when generating user and
phone contexts, use the line on button one.

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r4702 | raarts | 2005-10-13 13:02:04 +0200 (Thu, 13 Oct 2005) | 6 lines

- set QoS for the SIP channel as well as for the RTP (sound) channel.
- only allow more then one call per line if callwaiting is enabled
- use the largest value for the maximum number of calls per line (the largest
  of the max calls per phoneline and the max calls per phone).
- fixed missing join in query. This resulted in far too large polycom config files.

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r4698 | ivar | 2005-10-12 17:12:13 +0200 (Wed, 12 Oct 2005) | 1 line

bugfix: don't include deleted phonelines in globals.conf
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r4690 | raarts | 2005-10-12 13:37:10 +0200 (Wed, 12 Oct 2005) | 4 lines

Implement max calls per line for polycom phones
When you remove the default music from the queue screen, there
will be no music-on-hold.

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r4681 | ivar | 2005-10-12 10:21:35 +0200 (Wed, 12 Oct 2005) | 1 line

don't create tftp files for deleted phones
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r4676 | raarts | 2005-10-11 12:29:12 +0200 (Tue, 11 Oct 2005) | 2 lines

Move check-sync.pl to /usr/local/astium/sbin

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r4669 | raarts | 2005-10-11 10:41:08 +0200 (Tue, 11 Oct 2005) | 2 lines

Fixes ivr menu handling

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r4667 | raarts | 2005-10-11 09:15:40 +0200 (Tue, 11 Oct 2005) | 3 lines

If a local dhcpd.conf exists, include it.
Fix php compile bug.

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r4662 | raarts | 2005-10-10 12:59:06 +0200 (Mon, 10 Oct 2005) | 2 lines

Replaced command that gave an error, with the correct one.

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r4661 | raarts | 2005-10-10 12:35:47 +0200 (Mon, 10 Oct 2005) | 2 lines

Support remote reboot of Swissvoice phones.

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r4660 | raarts | 2005-10-10 12:16:43 +0200 (Mon, 10 Oct 2005) | 2 lines

Support provisioning SwissVoice IP10S phones.

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r4658 | raarts | 2005-10-07 22:03:33 +0200 (Fri, 07 Oct 2005) | 4 lines

Fix check-sync script to work with Polycom and SNOM phones.
Should work with Cisco too but didn't test it yet. Can't
get it to work with Grandstream

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r4654 | ivar | 2005-10-07 14:59:47 +0200 (Fri, 07 Oct 2005) | 1 line

fixed bug: deleted phones were also generated in sip.conf
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r4644 | ivar | 2005-10-07 09:55:53 +0200 (Fri, 07 Oct 2005) | 1 line

fixed little bug: missing \n
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r4634 | raarts | 2005-10-06 12:35:56 +0200 (Thu, 06 Oct 2005) | 2 lines

Give every user specific rights on certain events.

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r4633 | raarts | 2005-10-06 11:54:22 +0200 (Thu, 06 Oct 2005) | 2 lines

Let the proxy tell asterisk it wants to receive ALL events.

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r4632 | raarts | 2005-10-06 11:16:42 +0200 (Thu, 06 Oct 2005) | 3 lines

Now also regenerate astmanproxy.conf if the data for user number 2
are changed.

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r4622 | raarts | 2005-10-05 16:05:57 +0200 (Wed, 05 Oct 2005) | 3 lines

Generate manager.conf, so all users 
will be able to connect to astmanproxy.

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r4620 | wijnand | 2005-10-05 13:20:39 +0200 (Wed, 05 Oct 2005) | 2 lines

Convert the username to lowercase when checking if the user excists in postgresql

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r4619 | wijnand | 2005-10-05 12:37:10 +0200 (Wed, 05 Oct 2005) | 5 lines

Quick but not perfect fix.
Refuse to create users when the username is not valid for postgresql.
only a-z, 0-9 _ and - are allowed.
It seems to work, except for -, will fix that as soon as possible.

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r4617 | wijnand | 2005-10-05 10:29:11 +0200 (Wed, 05 Oct 2005) | 4 lines

:%s/nummeric/numeric/g

stupid stupid stupid me.

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r4616 | wijnand | 2005-10-05 10:20:13 +0200 (Wed, 05 Oct 2005) | 4 lines

If the contact username starts with a digit prepend a 'n'.
otherwise Postgresql does not accept the new user.


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r4612 | raarts | 2005-10-04 13:41:14 +0200 (Tue, 04 Oct 2005) | 2 lines

Add descriptive comments to the generated /etc/dhcpd.conf

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r4611 | raarts | 2005-10-04 12:29:40 +0200 (Tue, 04 Oct 2005) | 3 lines

DHCP fixes. Should work OK now.
also regenerate iax.conf and sip.conf on company change/add

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r4608 | raarts | 2005-10-04 11:30:05 +0200 (Tue, 04 Oct 2005) | 3 lines

Small typo in generating Grandstream configs resulted in phones not being
able to login.

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r4607 | raarts | 2005-10-04 11:25:37 +0200 (Tue, 04 Oct 2005) | 2 lines

Only look at active phonelines when generating configs for phones.

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r4598 | raarts | 2005-10-04 00:56:31 +0200 (Tue, 04 Oct 2005) | 3 lines

Re-arranged generating IVR menus. New and improved. 
Now it is possible to set timeouts, languages, and dialing of extensions..

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r4575 | raarts | 2005-10-01 12:20:02 +0200 (Sat, 01 Oct 2005) | 3 lines

Small error in query which creates queue.conf: also 
look at which button the channel is on.

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r4559 | raarts | 2005-09-29 17:36:41 +0200 (Thu, 29 Sep 2005) | 2 lines

Improve writing queue.conf

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r4557 | raarts | 2005-09-29 15:01:36 +0200 (Thu, 29 Sep 2005) | 2 lines

Fixed problem with incorrect dialplan numbering

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r4556 | raarts | 2005-09-29 14:52:35 +0200 (Thu, 29 Sep 2005) | 2 lines

Globals did not include Zap channels. Now it does.

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r4555 | raarts | 2005-09-29 14:13:03 +0200 (Thu, 29 Sep 2005) | 5 lines

Skip analog phones, and phones without a sip username
when generating sip.conf
Tell Polycomphones to just reboot when told, and not look
if the configfiles have changed on the FTP server.

------------------------------------------------------------------------
r4549 | raarts | 2005-09-29 11:56:28 +0200 (Thu, 29 Sep 2005) | 5 lines

- Set the company shortname label on the phone when 
  nobody is logged on, else display the extension number.
- Also create entries in sip.conf for phones where 
  nobody is logged on.

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r4538 | raarts | 2005-09-29 10:02:08 +0200 (Thu, 29 Sep 2005) | 2 lines

Fix missing quote in query

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r4534 | raarts | 2005-09-29 00:36:45 +0200 (Thu, 29 Sep 2005) | 3 lines

Provisioning and dialplan support for configuring separate buttons
on phones with separate functions.

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r4485 | raarts | 2005-09-27 09:40:53 +0200 (Tue, 27 Sep 2005) | 2 lines

Skip deleted phonelines when generating sip.conf

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r4484 | raarts | 2005-09-26 10:20:22 +0200 (Mon, 26 Sep 2005) | 2 lines

Only remove astcfg from chkconfig if totally uninstalling it.

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r4482 | raarts | 2005-09-25 21:34:26 +0200 (Sun, 25 Sep 2005) | 3 lines

It is now possible to give each phone its own extension. This cannot be
forwarded, does not have voicemail, and will not be recorded.

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r4442 | wijnand | 2005-09-22 16:03:18 +0200 (Thu, 22 Sep 2005) | 4 lines

Matthias loves lot's of empty lines.

I hate them, and I know I am not the only one.

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r4441 | wijnand | 2005-09-22 15:20:03 +0200 (Thu, 22 Sep 2005) | 3 lines

New module: postgresql. Astcfgd now creates a postgresql user for every active astium user.
Currently those users are only member of the astium_users group (this group has select access one some views) and these users have NO other rights on other objects.

------------------------------------------------------------------------
r4430 | raarts | 2005-09-21 11:41:19 +0200 (Wed, 21 Sep 2005) | 5 lines

Also use wctdm driver where available (this driver replaces wcfxs 
in newer asterisk versions).
Add timeout extensions to some strategix places in the dialplan.
Don't strip zeroes on incoming numbers in the dialplan.

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r4404 | raarts | 2005-09-19 15:27:48 +0200 (Mon, 19 Sep 2005) | 2 lines

Support in the background configurator for Digium 2-port ISDN-30 cards.

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r4394 | raarts | 2005-09-16 17:17:14 +0200 (Fri, 16 Sep 2005) | 2 lines

Set accountcode in phone channel and not in dialplan.

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r4391 | raarts | 2005-09-16 15:25:32 +0200 (Fri, 16 Sep 2005) | 2 lines

Ok, start generating CDR for incoming calls that get connected to queues.

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r4369 | raarts | 2005-09-14 10:47:37 +0200 (Wed, 14 Sep 2005) | 3 lines

New global: emailfrom.
Fix generating CDR with 'h' in it.

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r4339 | raarts | 2005-09-09 23:01:10 +0200 (Fri, 09 Sep 2005) | 2 lines

After an upgrade, restart astcfgd if it is running

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r4283 | raarts | 2005-09-07 17:40:37 +0200 (Wed, 07 Sep 2005) | 2 lines

Implemented queue priorities for inbound phone numbers.

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r4270 | raarts | 2005-09-06 17:00:53 +0200 (Tue, 06 Sep 2005) | 2 lines

Fix voicemail button not working on SNOM phones

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r4247 | raarts | 2005-09-05 15:06:42 +0200 (Mon, 05 Sep 2005) | 2 lines

Only look at non-deleted phones when generating dhcpd.conf

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r4237 | raarts | 2005-09-01 15:58:39 +0200 (Thu, 01 Sep 2005) | 4 lines

Generate better tos parameters to use DiffServ Expedited Forwarding.
in sip.conf and iax.conf. 
Do the same in the Polycom Configuration Files.

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r4213 | raarts | 2005-08-31 09:41:10 +0200 (Wed, 31 Aug 2005) | 2 lines

Perry forgot a ';'

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r4196 | perry | 2005-08-29 15:11:48 +0200 (Mon, 29 Aug 2005) | 1 line

Added option queue pause
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r4193 | perry | 2005-08-29 12:03:03 +0200 (Mon, 29 Aug 2005) | 1 line

fixed syntax error
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r4192 | perry | 2005-08-29 11:58:16 +0200 (Mon, 29 Aug 2005) | 1 line

Only add voicemail for contact's that are active'
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r4191 | perry | 2005-08-29 11:57:17 +0200 (Mon, 29 Aug 2005) | 1 line

Bug fixed for Scarlet Telecom
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r4189 | perry | 2005-08-29 11:23:49 +0200 (Mon, 29 Aug 2005) | 1 line

added support to astcfg for openntpd
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r4188 | perry | 2005-08-29 11:23:07 +0200 (Mon, 29 Aug 2005) | 1 line

Commiting a bug fix from Ron
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r4173 | raarts | 2005-08-26 17:07:39 +0200 (Fri, 26 Aug 2005) | 6 lines

Added and tested outbound SIP trunk support. Haven't tested
inbound so far.
Fixed bug for pre-inbound context, where we need to do
SetVar(company=$shortname), otherwise voicemail will not
work on external incoming lines.

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r4141 | raarts | 2005-08-23 10:40:59 +0200 (Tue, 23 Aug 2005) | 2 lines

Fixed bug, noticed by DNS-BE: missing underscores in dialplan expressions

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r4128 | raarts | 2005-08-04 15:21:22 +0200 (Thu, 04 Aug 2005) | 3 lines

Added script to reboot Polycom phones remotely. I tested it and it works.
Will come in handy later.

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r4122 | raarts | 2005-08-02 13:54:55 +0200 (Tue, 02 Aug 2005) | 4 lines

Ensure the agents are sorted in the queue the same way, as they are sorted 
in the GUI screen that lists the users in a queue.:wq


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r4121 | raarts | 2005-08-02 13:40:24 +0200 (Tue, 02 Aug 2005) | 3 lines

Should only specify monitor-format in queues.conf when we 
actually *want* to start monitoring

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r4112 | raarts | 2005-08-01 12:28:31 +0200 (Mon, 01 Aug 2005) | 2 lines

Implemented call waiting for Cisco's. 

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r4103 | raarts | 2005-07-31 22:44:02 +0200 (Sun, 31 Jul 2005) | 3 lines

Made DHCP server authoratative. do man dncpd.conf if you want 
to know what this means.

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r4093 | raarts | 2005-07-31 14:02:01 +0200 (Sun, 31 Jul 2005) | 2 lines

Implemented logo_url, directory_url, services_url for cisco phones

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r4090 | raarts | 2005-07-31 13:31:02 +0200 (Sun, 31 Jul 2005) | 2 lines

Fixed bug for generating phone configs where nobody was logged in.

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r4087 | raarts | 2005-07-30 15:53:32 +0200 (Sat, 30 Jul 2005) | 2 lines

Fixed two bugs which should have surfaced much earlier..

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r4079 | raarts | 2005-07-29 00:17:15 +0200 (Fri, 29 Jul 2005) | 3 lines

Set language to default company language for phones
that have noone loggin in.

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r4076 | raarts | 2005-07-29 00:06:40 +0200 (Fri, 29 Jul 2005) | 4 lines

Fixed hardcoded value for company in generating dialplans
Fixed illegal keywords in queues.conf (monitor_join and monitor_format)
Now correctly retrieves the from extension from the database.

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r4068 | raarts | 2005-07-26 14:40:05 +0200 (Tue, 26 Jul 2005) | 3 lines

From now on generates an agents.conf file.
Don't know when this will be used yet.

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r4067 | raarts | 2005-07-26 13:06:39 +0200 (Tue, 26 Jul 2005) | 4 lines

Previously you could only put PEOPLE in queues, but of course
that should be extensions (roles in the future). Fixed.
Also added a flag to honour call-forwards and voicemail in queues.

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r4057 | raarts | 2005-07-25 15:48:33 +0200 (Mon, 25 Jul 2005) | 2 lines

Now login and logout of your phone really works

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r4052 | raarts | 2005-07-21 17:09:04 +0200 (Thu, 21 Jul 2005) | 2 lines

Skip deleted phonelines when generating voicemail.conf

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r4050 | raarts | 2005-07-18 17:48:23 +0200 (Mon, 18 Jul 2005) | 3 lines

The test for redirect_on should be in the redirect query, not when generating
the dialplan.

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r4041 | raarts | 2005-07-18 15:39:54 +0200 (Mon, 18 Jul 2005) | 2 lines

Make the message button work for polycom phones

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r4040 | raarts | 2005-07-18 15:27:53 +0200 (Mon, 18 Jul 2005) | 2 lines

Put correct NTP server and gmt offset into /etc/dhcpd.conf

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r4039 | raarts | 2005-07-18 14:06:37 +0200 (Mon, 18 Jul 2005) | 2 lines

Don't add deleted extensions to the company extension list.

------------------------------------------------------------------------
r4037 | raarts | 2005-07-15 14:07:05 +0200 (Fri, 15 Jul 2005) | 2 lines

Support for Octo BRI card.

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r4033 | raarts | 2005-07-14 15:53:06 +0200 (Thu, 14 Jul 2005) | 2 lines

Strip comma's from firstname/lastname, because they mess up voicemail.conf

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r4015 | raarts | 2005-07-06 16:08:42 +0200 (Wed, 06 Jul 2005) | 2 lines

Remove verbose printing which made the daemon go exit.

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r4011 | raarts | 2005-07-06 14:09:02 +0200 (Wed, 06 Jul 2005) | 2 lines

Unified all snom-*.php into ONE snom.php

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r4009 | raarts | 2005-07-06 10:17:17 +0200 (Wed, 06 Jul 2005) | 2 lines

From now on Grandstream phones use the Astium as their logging server

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r4008 | raarts | 2005-07-05 17:00:29 +0200 (Tue, 05 Jul 2005) | 2 lines

Improved Grandstream configuration

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r3998 | raarts | 2005-07-04 16:14:22 +0200 (Mon, 04 Jul 2005) | 2 lines

Now correctly generates static routes into /etc/sysconfig/network-scripts/route-*

------------------------------------------------------------------------
r3993 | wijnand | 2005-06-30 12:17:56 +0200 (Thu, 30 Jun 2005) | 2 lines

Only copy files that exist, prevents some php warnings.

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r3988 | raarts | 2005-06-27 15:47:00 +0200 (Mon, 27 Jun 2005) | 3 lines

Finally fix multiple number patterns for mobile (for example)
Small bugs in IVR menus (dup '*' and '#')

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r3955 | raarts | 2005-06-10 16:25:19 +0200 (Fri, 10 Jun 2005) | 3 lines

Fix putting in correct ipaddresses for option-150.
Fix genetaring syntax errors in dhcpd.conf

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r3954 | raarts | 2005-06-10 14:58:09 +0200 (Fri, 10 Jun 2005) | 3 lines

Only do DHCP for phone which have actual IP
and mac info filled into their webpage

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r3951 | wijnand | 2005-06-09 13:41:22 +0200 (Thu, 09 Jun 2005) | 3 lines

Don't try to deliver mail locally when mailservice='no' in the domain table. While there, respect the internal mailserver setting if there.


------------------------------------------------------------------------
r3892 | wijnand | 2005-05-18 12:46:04 +0200 (Wed, 18 May 2005) | 2 lines

Kill the output of shar.

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r3891 | wijnand | 2005-05-18 12:02:19 +0200 (Wed, 18 May 2005) | 2 lines

Make postmap silent

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r3890 | wijnand | 2005-05-18 11:51:01 +0200 (Wed, 18 May 2005) | 2 lines

Don't try to print a entry in /etc/postfix/virtual for a default emailaddress when the actual emailaddress is missing

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r3885 | wijnand | 2005-05-11 11:33:15 +0200 (Wed, 11 May 2005) | 2 lines

Make amavisd less verbose in the logs. Loglevel 0 is enough (same as NBS)

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r3815 | raarts | 2005-04-25 15:01:23 +0200 (Mon, 25 Apr 2005) | 4 lines

Don't write dupe CDR records for Queues
Generalize Call recording
Fix bug in generating backup routes

------------------------------------------------------------------------
r3743 | raarts | 2005-04-18 15:45:13 +0200 (Mon, 18 Apr 2005) | 2 lines

Drop at most 0.5% of VoIP packets.

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r3741 | raarts | 2005-04-15 12:28:43 +0200 (Fri, 15 Apr 2005) | 2 lines

Don't use _. pattern any more in the dialplan, it messes up the CDR

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r3701 | raarts | 2005-04-08 14:14:24 +0200 (Fri, 08 Apr 2005) | 2 lines

Now actually *uses* the qualify setting from the web.

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r3700 | raarts | 2005-04-08 14:12:45 +0200 (Fri, 08 Apr 2005) | 2 lines

Coding error: forgot to reset a variable

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r3699 | raarts | 2005-04-08 11:15:55 +0200 (Fri, 08 Apr 2005) | 2 lines

Always properly put quotes around an expression!

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r3688 | raarts | 2005-04-06 15:44:49 +0200 (Wed, 06 Apr 2005) | 6 lines

Now you can assign an extension the 'trunk' status.
If that is set, you can forward incoming calls into
an outgoing trunk.
Also fixed the overlapdial parameter. You could set it in the 
web, but otherwise it was ignored by the backend.

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r3676 | raarts | 2005-04-04 16:41:42 +0200 (Mon, 04 Apr 2005) | 2 lines

Small bug introduced with call-recording

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r3674 | raarts | 2005-04-04 14:08:47 +0200 (Mon, 04 Apr 2005) | 2 lines

Support outbound call recording

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r3673 | raarts | 2005-04-01 21:38:02 +0200 (Fri, 01 Apr 2005) | 2 lines

Added unload entry for the TE110P card.

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r3671 | raarts | 2005-03-31 17:37:54 +0200 (Thu, 31 Mar 2005) | 2 lines

Fixed old cardid bug in zaptel.php

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r3668 | raarts | 2005-03-31 17:08:05 +0200 (Thu, 31 Mar 2005) | 2 lines

Support for the TE110P card

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r3666 | raarts | 2005-03-31 00:59:53 +0200 (Thu, 31 Mar 2005) | 2 lines

Better set default outgoing routing as last

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r3661 | raarts | 2005-03-30 23:46:41 +0200 (Wed, 30 Mar 2005) | 2 lines

Support for IAX trunks, asterisk@home, preliminary for sip trunks (don't use yet!)

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r3609 | raarts | 2005-03-16 18:01:43 +0100 (Wed, 16 Mar 2005) | 2 lines

Support for call-forward while being in a queue.

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r3592 | raarts | 2005-03-15 11:23:58 +0100 (Tue, 15 Mar 2005) | 2 lines

Now fully support Cisco 79xx phones.

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r3581 | raarts | 2005-03-14 15:22:28 +0100 (Mon, 14 Mar 2005) | 2 lines

Support for upgrading cisco phones from CM firmware to SIP

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r3580 | raarts | 2005-03-14 13:51:41 +0100 (Mon, 14 Mar 2005) | 2 lines

First cut at full cisco support

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r3321 | raarts | 2005-03-13 15:17:28 +0100 (Sun, 13 Mar 2005) | 1 line

create 1.0 branch for astcfgd
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r3175 | raarts | 2005-03-07 15:23:04 +0100 (Mon, 07 Mar 2005) | 2 lines

Also respond to user changes

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r3136 | raarts | 2005-03-04 17:26:32 +0100 (Fri, 04 Mar 2005) | 4 lines

Use different length of lable texts for different types
of polycom phones because they use different fonts on different
displays.

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r3133 | raarts | 2005-03-04 16:49:42 +0100 (Fri, 04 Mar 2005) | 3 lines

Presence support (SNOM function keys, and Ploycom Buddy lists)
No more dialing '0' for outgoing calls.

------------------------------------------------------------------------
r3054 | raarts | 2005-03-01 16:30:43 +0100 (Tue, 01 Mar 2005) | 3 lines

Support the voicemail buttons on the Polycom.
(Message Center).

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r3053 | raarts | 2005-02-28 16:03:57 +0100 (Mon, 28 Feb 2005) | 3 lines

Always generate a complete zap channel definition from now on.
Also sort by channel number

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r3052 | raarts | 2005-02-28 13:40:57 +0100 (Mon, 28 Feb 2005) | 2 lines

Fixed hardcoded value, shoud have been a variable.

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r3051 | raarts | 2005-02-26 23:19:18 +0100 (Sat, 26 Feb 2005) | 2 lines

Completed company fastdial support

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r3049 | raarts | 2005-02-26 21:05:06 +0100 (Sat, 26 Feb 2005) | 2 lines

Fixed small typo

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r3038 | raarts | 2005-02-22 22:40:22 +0100 (Tue, 22 Feb 2005) | 2 lines

Say the right words when entering DISA

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r3037 | raarts | 2005-02-22 22:32:53 +0100 (Tue, 22 Feb 2005) | 2 lines

Implement inbound redirection tables

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r3020 | raarts | 2005-02-21 16:41:26 +0100 (Mon, 21 Feb 2005) | 3 lines

Amavis and openexchage fixes by wijnand
Implemented disa for ourselves

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r3009 | raarts | 2005-02-21 10:05:29 +0100 (Mon, 21 Feb 2005) | 3 lines

Reflect the actual asterisk default settings, so we only generate
non-default settings into zapata.conf

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r3006 | raarts | 2005-02-21 00:01:51 +0100 (Mon, 21 Feb 2005) | 3 lines

Applied the improved generation of /etc/zaptel.conf to all
cards with an E1 or T1 port.

------------------------------------------------------------------------
r3005 | raarts | 2005-02-20 23:57:10 +0100 (Sun, 20 Feb 2005) | 17 lines

Fixes as a result of starting to use the Astium ourselves:
- Now by default create a directory '/etc/asterisk/companyNNNN/sounds' where
  apache can write its uploaded files. Also give asterisk access to that files
  and when using them in the dialplan use the full path.
- zap_channels were not properly generated in queues.conf
- add SetLanguage when jumping to a Queue on inbound calls
- use the maxtime variable wherever you jump to a Queue
- fix calling extensions and queues in IVR menus.
- when a call was auto-forwarded to an outside phone (say a mobile) that
  phone should be able to transfer the call back in. Adapted the dialplan
  to enable this.
- fixes for analog signalling in zapata.conf
- now generate aliaslist into /etc/postfix/virtual
- generate defaultzone in /etc/zaptel.conf so we have an analog dialtone..
- rewrote part of /etc/zaptel.conf generation to accomodate for channelbanks
  using channelbanks is still very awkward though...

------------------------------------------------------------------------
r2954 | raarts | 2005-02-02 15:37:49 +0100 (Wed, 02 Feb 2005) | 2 lines

First cut at Polycom provisioning.

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r2951 | raarts | 2005-02-02 14:53:47 +0100 (Wed, 02 Feb 2005) | 2 lines

Prevent PHP warnings

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r2950 | raarts | 2005-02-02 14:49:16 +0100 (Wed, 02 Feb 2005) | 3 lines

Oops, repaired bug that only happened when a country has
more then one prefix for (for example) mobile calls

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r2917 | raarts | 2005-01-28 17:30:52 +0100 (Fri, 28 Jan 2005) | 2 lines

Implemented '##', redial last number.

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r2910 | raarts | 2005-01-28 12:46:17 +0100 (Fri, 28 Jan 2005) | 2 lines

Implemented fastdial support in the dialplan generation

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r2905 | raarts | 2005-01-25 23:18:28 +0100 (Tue, 25 Jan 2005) | 3 lines

Regenerate queues.conf when a queue_contact changes
Regenerate all inc files wnen a company disappears.

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r2901 | raarts | 2005-01-25 17:33:05 +0100 (Tue, 25 Jan 2005) | 2 lines

Support for other extension sizes than 4.

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r2889 | raarts | 2005-01-20 17:01:42 +0100 (Thu, 20 Jan 2005) | 3 lines

Make it possible to point incoming numbers directly
to a queue.

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r2879 | raarts | 2005-01-17 16:45:41 +0100 (Mon, 17 Jan 2005) | 3 lines

Moved outbound processing from system to company. It's
now generated in extensions.conf

------------------------------------------------------------------------
r2878 | ivar | 2005-01-17 16:40:59 +0100 (Mon, 17 Jan 2005) | 1 line

have default language come from deflang field in company table
------------------------------------------------------------------------
r2875 | ivar | 2005-01-17 15:51:49 +0100 (Mon, 17 Jan 2005) | 3 lines

update asterisk.php to correctly play soundfiles in ivrmenu,
jump to phoneline or new menu after welcome prompt,
work with function extensions.
------------------------------------------------------------------------
r2871 | raarts | 2005-01-16 15:56:34 +0100 (Sun, 16 Jan 2005) | 4 lines

Queues actually do work now
Fixed error in /etc/dhcpd.conf where tftp-servers
were not generated correctly for SNOM phones.

------------------------------------------------------------------------
r2868 | raarts | 2005-01-16 01:23:01 +0100 (Sun, 16 Jan 2005) | 2 lines

Added generating of queues.conf. More echocancel fixes.

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r2854 | raarts | 2005-01-12 17:05:51 +0100 (Wed, 12 Jan 2005) | 2 lines

Fix for *1 (at last it works)

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r2853 | raarts | 2005-01-12 14:45:28 +0100 (Wed, 12 Jan 2005) | 2 lines

Always explicitly state the echocancel setting

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r2819 | matthias | 2005-01-03 12:26:36 +0100 (Mon, 03 Jan 2005) | 1 line

added comments
------------------------------------------------------------------------
r2801 | raarts | 2004-12-28 13:12:54 +0100 (Tue, 28 Dec 2004) | 3 lines

Fixed generating dhcpd.conf independant of the number of interfaces
Fixed generating Grandstream configfiles both for ATA-286 and BUDGETONE

------------------------------------------------------------------------
r2790 | matthias | 2004-12-24 12:34:10 +0100 (Fri, 24 Dec 2004) | 1 line

also generate an amavis config on first startup of the daemon
------------------------------------------------------------------------
r2781 | raarts | 2004-12-23 23:22:49 +0100 (Thu, 23 Dec 2004) | 3 lines

Take the 'function' field into account when generating
voicemail.conf and other files

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r2737 | raarts | 2004-12-20 17:04:13 +0100 (Mon, 20 Dec 2004) | 2 lines

Improved error message

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r2734 | raarts | 2004-12-20 16:17:38 +0100 (Mon, 20 Dec 2004) | 3 lines

Added (inter)nationalprefix, this way asterisk prefixes incoming
phone calls with the proper prefixes.

------------------------------------------------------------------------
r2726 | matthias | 2004-12-20 12:26:36 +0100 (Mon, 20 Dec 2004) | 1 line

remove a debug line that slipped through svn
------------------------------------------------------------------------
r2722 | raarts | 2004-12-20 11:45:41 +0100 (Mon, 20 Dec 2004) | 2 lines

Improved layout in outbound.conf

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r2712 | raarts | 2004-12-17 14:22:45 +0100 (Fri, 17 Dec 2004) | 3 lines

The system function writes to standard output, causing the
daemon to stop. The exec() doesn't

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r2705 | matthias | 2004-12-16 15:05:23 +0100 (Thu, 16 Dec 2004) | 1 line

ipsec and postfix fixes
------------------------------------------------------------------------
r2693 | raarts | 2004-12-15 16:46:59 +0100 (Wed, 15 Dec 2004) | 2 lines

Fix small error in SQL query

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r2686 | matthias | 2004-12-15 10:22:07 +0100 (Wed, 15 Dec 2004) | 1 line

only backup certs, if the certificate is regenerated
------------------------------------------------------------------------
r2684 | matthias | 2004-12-14 16:38:24 +0100 (Tue, 14 Dec 2004) | 1 line

add a function to put the certs in a single shar into the database
------------------------------------------------------------------------
r2681 | raarts | 2004-12-14 16:22:19 +0100 (Tue, 14 Dec 2004) | 4 lines

Fixed errors in generating /etc/sysconfig/zaptel
Corrected generating /etc/asterisk/zapata.conf. I did not
correctly process 4-port PRI cards.

------------------------------------------------------------------------
r2678 | raarts | 2004-12-14 12:01:34 +0100 (Tue, 14 Dec 2004) | 5 lines

Implemented generating IVR menus in dialplan.
Removed lots of 'echo' statements that made astcfgd crash.
Please remember: writing to standard output when a PHP script
is in the background as a daemon, crashes the script!

------------------------------------------------------------------------
r2669 | matthias | 2004-12-13 10:26:37 +0100 (Mon, 13 Dec 2004) | 1 line

add beter ox groups support
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r2655 | raarts | 2004-12-08 17:46:57 +0100 (Wed, 08 Dec 2004) | 2 lines

Typo in asterisk.php

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r2600 | raarts | 2004-12-06 09:13:23 +0100 (Mon, 06 Dec 2004) | 3 lines

Finally fixed irritating DHCP error
Fixed error in generating macro(extension,..) in the dialplan

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r2536 | matthias | 2004-11-30 08:59:52 +0100 (Tue, 30 Nov 2004) | 1 line

removed debug lines
------------------------------------------------------------------------
r2519 | raarts | 2004-11-29 11:49:31 +0100 (Mon, 29 Nov 2004) | 6 lines

Implemented hot-desking.
Removed falling back to 5 sec wait after 500 secs, perry
write to Linus if you think the sleep() function is 
inefficient (or to Rasmus if you think PHP is).
Added debugging for all SQL queries

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r2518 | matthias | 2004-11-29 11:42:10 +0100 (Mon, 29 Nov 2004) | 1 line

modified postfix configuration, external mail receiving is working now
------------------------------------------------------------------------
r2497 | matthias | 2004-11-24 14:49:19 +0100 (Wed, 24 Nov 2004) | 1 line

small bugfixes in the script to get daemonized scanning working again
------------------------------------------------------------------------
r2490 | matthias | 2004-11-24 13:44:33 +0100 (Wed, 24 Nov 2004) | 1 line

add config for main.cf, we must trust friendly interfaces not all :)
------------------------------------------------------------------------
r2484 | raarts | 2004-11-24 12:37:59 +0100 (Wed, 24 Nov 2004) | 5 lines

Implemented Hot Desking (first steps only many things don't work yet,
like callgroups, voicemail detection by your phone, and your name on your
phone when you are logged in)
astcfgd now finally has commandline options for debugging (use astcfgd -d1)

------------------------------------------------------------------------
r2453 | matthias | 2004-11-19 12:20:27 +0100 (Fri, 19 Nov 2004) | 1 line

integrate amavis into astcfgd
------------------------------------------------------------------------
r2447 | matthias | 2004-11-18 16:44:31 +0100 (Thu, 18 Nov 2004) | 1 line

bring amavis support
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r2429 | matthias | 2004-11-17 12:32:21 +0100 (Wed, 17 Nov 2004) | 1 line

fixed up pptp support for fixed ip
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r2425 | matthias | 2004-11-17 10:20:46 +0100 (Wed, 17 Nov 2004) | 1 line

disable dhcpc killing, works not good yet
------------------------------------------------------------------------
r2422 | matthias | 2004-11-16 15:54:14 +0100 (Tue, 16 Nov 2004) | 1 line

add fixed ip support for pptp users
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r2417 | matthias | 2004-11-16 12:25:43 +0100 (Tue, 16 Nov 2004) | 1 line

kill dhcp client before restarting the network
------------------------------------------------------------------------
r2352 | raarts | 2004-11-03 21:33:06 +0100 (Wed, 03 Nov 2004) | 2 lines

the 'span=1,1..' line is needed for *every* BRI port.

------------------------------------------------------------------------
r2351 | raarts | 2004-11-03 21:30:19 +0100 (Wed, 03 Nov 2004) | 3 lines

incominglimit and outcominglimit are deprecated and will be removed
in a future asterisk version.

------------------------------------------------------------------------
r2344 | raarts | 2004-11-03 13:11:45 +0100 (Wed, 03 Nov 2004) | 2 lines

Implemented inbound processing, fixed ZEROCONF=no in network

------------------------------------------------------------------------
r2312 | raarts | 2004-10-28 15:29:45 +0200 (Thu, 28 Oct 2004) | 4 lines

Temporary fix for the fact that if you dial 0205628282, the ISDN line strips
the starting zero from that number, before telling it to asterisk.
I contacted Junghanns in Germany, and they will implement it in the next upgrade.

------------------------------------------------------------------------
r2305 | raarts | 2004-10-28 13:31:07 +0200 (Thu, 28 Oct 2004) | 2 lines

Outbound fixes, re-added zaptel to astcfgd

------------------------------------------------------------------------
r2268 | raarts | 2004-10-25 17:44:01 +0200 (Mon, 25 Oct 2004) | 4 lines

Added feature to require a pin before dialing expensive numbers
Split up initial variable setting in 'phone' variables and 'user' variables,
so we can change user info by jumping into a context.

------------------------------------------------------------------------
r2260 | raarts | 2004-10-25 14:54:58 +0200 (Mon, 25 Oct 2004) | 3 lines

Added outbound call routing support.
Added pre-auth.

------------------------------------------------------------------------
r2255 | raarts | 2004-10-22 16:45:28 +0200 (Fri, 22 Oct 2004) | 2 lines

Support for International dialplans (Starting with Belgium)

------------------------------------------------------------------------
r2249 | matthias | 2004-10-22 12:36:14 +0200 (Fri, 22 Oct 2004) | 1 line

fixed Makefile.am
------------------------------------------------------------------------
r2248 | matthias | 2004-10-22 11:27:43 +0200 (Fri, 22 Oct 2004) | 1 line

latest support for ox
------------------------------------------------------------------------
r2209 | perry | 2004-10-15 11:20:20 +0200 (Fri, 15 Oct 2004) | 7 lines

The default idle time is not static anymore.
If the astcfgd daemon is 500 time idle, 
the idle time changes to 5 seconds.
When there is one item in the activate queue the 
idle time changes back to 1 second.


------------------------------------------------------------------------
r2200 | matthias | 2004-10-13 15:34:25 +0200 (Wed, 13 Oct 2004) | 1 line

small fix
------------------------------------------------------------------------
r2191 | perry | 2004-10-12 16:41:53 +0200 (Tue, 12 Oct 2004) | 1 line

fixed a bug
------------------------------------------------------------------------
r2189 | perry | 2004-10-12 16:13:13 +0200 (Tue, 12 Oct 2004) | 2 lines

Budgetone 101 uses ntp1.netland.nl as ntp
server
------------------------------------------------------------------------
r2188 | perry | 2004-10-12 16:09:12 +0200 (Tue, 12 Oct 2004) | 2 lines

Budgetone 101 uses ntp1.netland.nl as ntp
server
------------------------------------------------------------------------
r2186 | matthias | 2004-10-12 14:59:58 +0200 (Tue, 12 Oct 2004) | 1 line

wins support in astcfgd
------------------------------------------------------------------------
r2184 | matthias | 2004-10-12 09:35:37 +0200 (Tue, 12 Oct 2004) | 1 line

fixed return of the spatie bug
------------------------------------------------------------------------
r2180 | raarts | 2004-10-11 15:17:34 +0200 (Mon, 11 Oct 2004) | 3 lines

Create /etc/sysconfig/zaptel with an RMODULES line, so
/etc/init.d/zaptel knows how to unload all modules

------------------------------------------------------------------------
r2176 | matthias | 2004-10-11 14:04:07 +0200 (Mon, 11 Oct 2004) | 1 line

ipsec code cleanup and support for racoon rc script
------------------------------------------------------------------------
r2175 | raarts | 2004-10-11 12:51:33 +0200 (Mon, 11 Oct 2004) | 4 lines

Implemented hints for SNOM keyboard lights
WISIP support, (always has user=phone, so SIP entry and username MUST be same as phonenumber)
pridialplan and prilocaldialplan setting for BRI/PRI entries.

------------------------------------------------------------------------
r2173 | perry | 2004-10-11 12:22:58 +0200 (Mon, 11 Oct 2004) | 1 line

fixed a bug
------------------------------------------------------------------------
r2168 | raarts | 2004-10-10 08:45:56 +0200 (Sun, 10 Oct 2004) | 4 lines

Generate definitions for outside lines from the database.
Also for company-specific zap channels
Don't restart ntpd every time you see an activate record.

------------------------------------------------------------------------
r2167 | perry | 2004-10-08 14:22:08 +0200 (Fri, 08 Oct 2004) | 1 line

added option version
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r2166 | perry | 2004-10-08 14:21:31 +0200 (Fri, 08 Oct 2004) | 1 line

added a new function
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r2165 | perry | 2004-10-08 14:12:21 +0200 (Fri, 08 Oct 2004) | 1 line

some more grandstream budget100 support
------------------------------------------------------------------------
r2152 | raarts | 2004-10-06 22:40:18 +0200 (Wed, 06 Oct 2004) | 3 lines

Fixed dup lbo values in output. Adjusted for
moving the signalling to zap_channel

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r2145 | matthias | 2004-10-06 12:36:50 +0200 (Wed, 06 Oct 2004) | 1 line

include zaptel in build proces
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r2142 | perry | 2004-10-05 16:49:49 +0200 (Tue, 05 Oct 2004) | 1 line

added tftp server to config
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r2141 | matthias | 2004-10-05 16:22:08 +0200 (Tue, 05 Oct 2004) | 1 line

small ipsec fixes
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r2139 | perry | 2004-10-05 15:55:38 +0200 (Tue, 05 Oct 2004) | 1 line

Grandstream budget 101 phone can now connect to the astium
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r2134 | raarts | 2004-10-04 17:33:29 +0200 (Mon, 04 Oct 2004) | 2 lines

Now only generate the modules we really need

------------------------------------------------------------------------
r2132 | raarts | 2004-10-04 17:20:49 +0200 (Mon, 04 Oct 2004) | 2 lines

Now auto-generates /etc/zaptel.conf and /etc/sysconfig/zaptel

------------------------------------------------------------------------
r2131 | perry | 2004-10-04 15:05:08 +0200 (Mon, 04 Oct 2004) | 1 line

Added code for the budget 101 voiceip telephone
------------------------------------------------------------------------
r2129 | matthias | 2004-10-04 12:22:31 +0200 (Mon, 04 Oct 2004) | 1 line

fixed ipsec module to work in php daemon mode
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r2124 | perry | 2004-10-01 10:32:37 +0200 (Fri, 01 Oct 2004) | 1 line

/etc/network wil now als be created when adding/deleting/changing staticroutes, was needed because of default gateway
------------------------------------------------------------------------
r2122 | perry | 2004-10-01 10:05:51 +0200 (Fri, 01 Oct 2004) | 1 line

removed test file
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r2121 | matthias | 2004-09-30 17:06:03 +0200 (Thu, 30 Sep 2004) | 1 line

small fixes
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r2119 | matthias | 2004-09-30 15:33:06 +0200 (Thu, 30 Sep 2004) | 1 line

removed all stdout output from ipsec.php
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r2117 | perry | 2004-09-30 12:39:55 +0200 (Thu, 30 Sep 2004) | 1 line

fixed a typo
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r2113 | raarts | 2004-09-29 23:02:59 +0200 (Wed, 29 Sep 2004) | 2 lines

The Z matches 1-9 in a dialplan, the N matches 2-9

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r2111 | raarts | 2004-09-29 15:27:52 +0200 (Wed, 29 Sep 2004) | 2 lines

Now sets variable 'useridnum' on every call

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r2108 | perry | 2004-09-29 12:05:14 +0200 (Wed, 29 Sep 2004) | 1 line

relaxed the access permissions to the ntp server
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r2107 | perry | 2004-09-29 11:19:26 +0200 (Wed, 29 Sep 2004) | 1 line

dhcpd only listens to interfaces that are not external interfaces
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r2104 | raarts | 2004-09-28 23:11:15 +0200 (Tue, 28 Sep 2004) | 2 lines

Added support for phone IP gateway settings (option routers)

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r2097 | perry | 2004-09-28 14:54:53 +0200 (Tue, 28 Sep 2004) | 1 line

added quoates to a string
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r2092 | raarts | 2004-09-28 09:39:05 +0200 (Tue, 28 Sep 2004) | 2 lines

Generate correct tftp-boot-server lines for phones

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r2087 | raarts | 2004-09-27 15:41:18 +0200 (Mon, 27 Sep 2004) | 2 lines

Set variables when a phone sets up a call. Other fixes

------------------------------------------------------------------------
r2036 | raarts | 2004-09-24 14:29:13 +0200 (Fri, 24 Sep 2004) | 3 lines

Improved debugging, moved astglobals loading to the real start
of the loop.

------------------------------------------------------------------------
r2034 | raarts | 2004-09-24 14:25:15 +0200 (Fri, 24 Sep 2004) | 4 lines

Added generating /etc/asterisk/voicemail.conf
Added outbound calling permissions for phones
Made sure onyl services that are activated are restarted

------------------------------------------------------------------------
r2020 | raarts | 2004-09-24 12:18:50 +0200 (Fri, 24 Sep 2004) | 4 lines

Removed ALL print statements to stdout from astcfgd
and modules, because the daemon crashes on them.
From now on, use 'if ($debug > 0) print ..." !!!!!!!!

------------------------------------------------------------------------
r2010 | raarts | 2004-09-24 09:53:04 +0200 (Fri, 24 Sep 2004) | 2 lines

Added tz, dialback and dialout contexts to voicemail definitions

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r2005 | raarts | 2004-09-24 00:51:47 +0200 (Fri, 24 Sep 2004) | 3 lines

Fixed MWI for SNOM phones, added subscribe context
Added dialout= and callback= to voicemail extensions

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r1996 | raarts | 2004-09-23 14:55:06 +0200 (Thu, 23 Sep 2004) | 3 lines

Lots of improvements to asterisk configuration, as a result
of redesigning the dialplan together with michiel.

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r1882 | perry | 2004-09-15 11:48:47 +0200 (Wed, 15 Sep 2004) | 1 line

fixed a bug
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r1881 | perry | 2004-09-15 10:51:31 +0200 (Wed, 15 Sep 2004) | 2 lines

ntp module uses the tmpfiles class now to create
tmp config files
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r1879 | raarts | 2004-09-15 02:01:24 +0200 (Wed, 15 Sep 2004) | 2 lines

Fixed card ID

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r1872 | raarts | 2004-09-14 21:09:19 +0200 (Tue, 14 Sep 2004) | 2 lines

Added generating inboud.conf, rewrote some parts

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r1868 | perry | 2004-09-14 15:39:50 +0200 (Tue, 14 Sep 2004) | 1 line

removed a file
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r1865 | perry | 2004-09-14 14:13:23 +0200 (Tue, 14 Sep 2004) | 1 line

Removed a tmp file
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r1864 | perry | 2004-09-14 14:12:41 +0200 (Tue, 14 Sep 2004) | 1 line

Added a test file
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r1863 | perry | 2004-09-14 14:01:46 +0200 (Tue, 14 Sep 2004) | 1 line

fixed a syntax buf
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r1862 | perry | 2004-09-14 12:55:43 +0200 (Tue, 14 Sep 2004) | 2 lines

Added a function to make it simple to 
create tmp config files and activate them
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r1860 | raarts | 2004-09-14 11:52:00 +0200 (Tue, 14 Sep 2004) | 3 lines

That damned global debug flag again, when will
someone implement ./astcfgd -d?

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r1858 | raarts | 2004-09-14 11:47:35 +0200 (Tue, 14 Sep 2004) | 3 lines

Removed superfluous debug info
Cisco support (initial)

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r1856 | perry | 2004-09-14 11:38:39 +0200 (Tue, 14 Sep 2004) | 1 line

added tmpfiles.php
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r1834 | raarts | 2004-09-13 12:51:32 +0200 (Mon, 13 Sep 2004) | 3 lines

Added accountcode to extension config, so ot gets in the CDR
Added tftp.php to Makefile.am

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r1826 | raarts | 2004-09-11 23:36:34 +0200 (Sat, 11 Sep 2004) | 6 lines

First shot at configuring GrandStream phones by tftp, yes
its been reverse-engineered. See:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg43052.html
Moved all IPv4 includes to one spot.
More debug info

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r1823 | raarts | 2004-09-11 00:27:03 +0200 (Sat, 11 Sep 2004) | 4 lines

Fixed generation of asterisk files, only do dhcp for
internal network cards, repaired company_by_*() functions
Always recreate ntp.conf for now

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r1815 | perry | 2004-09-09 13:16:45 +0200 (Thu, 09 Sep 2004) | 1 line


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r1812 | perry | 2004-09-09 12:17:22 +0200 (Thu, 09 Sep 2004) | 4 lines

Asterisk did not start up after reboot.
Because /etc/asterisk/pg.conf did not contain a valid login.
From now astcfgd create's a pg.conf with a valid account.

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r1811 | perry | 2004-09-09 11:35:59 +0200 (Thu, 09 Sep 2004) | 2 lines

d

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r1810 | perry | 2004-09-09 11:35:42 +0200 (Thu, 09 Sep 2004) | 4 lines

Astium wil now startup after postgresql.
Cleaned up debugging output stuff.
Added ($debug > 0) to debugging stuff.

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r1807 | raarts | 2004-09-09 01:15:45 +0200 (Thu, 09 Sep 2004) | 2 lines

Oops, debug should be 0 in production environments

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r1804 | raarts | 2004-09-09 01:02:32 +0200 (Thu, 09 Sep 2004) | 11 lines

Lots of rewrites, moved code around:
Now every time all astglobas are re-read before processing starts
into a global array $astglobals, (and $astglobalname, and $astglobalcat),
so functions to get the astglobals are not needed any more.
Rewrote network.php a bit.
Added log_msg(), please USE THIS FUNCTION!
Renamed $astcfg['debug'] to $debug.
Changed the interfaces table to contain the name of
the interface, and its type too (public, internal etc)
The routing table should be next.

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r1799 | perry | 2004-09-08 15:51:51 +0200 (Wed, 08 Sep 2004) | 1 line

renamed netland to neonova
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r1798 | perry | 2004-09-08 15:49:25 +0200 (Wed, 08 Sep 2004) | 1 line

renamed user netland to neonova
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r1795 | perry | 2004-09-08 15:11:46 +0200 (Wed, 08 Sep 2004) | 1 line

ron replaced the word retreive by the correct spelling retrieve
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r1791 | perry | 2004-09-08 11:00:57 +0200 (Wed, 08 Sep 2004) | 1 line

added a new module to the make file.
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r1790 | perry | 2004-09-08 10:55:37 +0200 (Wed, 08 Sep 2004) | 1 line

test commit
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r1789 | perry | 2004-09-08 10:13:16 +0200 (Wed, 08 Sep 2004) | 1 line

moved a class
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r1788 | perry | 2004-09-07 15:43:03 +0200 (Tue, 07 Sep 2004) | 1 line

bug fixe
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r1787 | perry | 2004-09-07 13:59:21 +0200 (Tue, 07 Sep 2004) | 1 line

Fixed a few bugs in the ntpd module
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r1786 | perry | 2004-09-07 13:35:01 +0200 (Tue, 07 Sep 2004) | 1 line

ntpd does things the astium way now.
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r1785 | perry | 2004-09-07 12:30:14 +0200 (Tue, 07 Sep 2004) | 1 line

ntpd.conf uses the local_network function to give access to the ntp server.
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r1784 | perry | 2004-09-07 12:26:22 +0200 (Tue, 07 Sep 2004) | 1 line

Added a function to retrieve al local network
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r1783 | perry | 2004-09-07 12:05:33 +0200 (Tue, 07 Sep 2004) | 1 line

Added network restrict to ntp config file.
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r1781 | perry | 2004-09-07 10:57:35 +0200 (Tue, 07 Sep 2004) | 1 line

Fixed a bug
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r1780 | perry | 2004-09-07 10:48:19 +0200 (Tue, 07 Sep 2004) | 1 line

Added reload optie to ntpd module
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r1779 | perry | 2004-09-07 10:32:58 +0200 (Tue, 07 Sep 2004) | 1 line

added a generated by XXX function
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r1778 | perry | 2004-09-07 10:16:36 +0200 (Tue, 07 Sep 2004) | 1 line

added ntpd modules to astcfgd
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r1777 | perry | 2004-09-07 10:14:25 +0200 (Tue, 07 Sep 2004) | 1 line

added some new general functions (used by ntpd config module).
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r1776 | perry | 2004-09-07 09:38:52 +0200 (Tue, 07 Sep 2004) | 1 line

First edition of ntpd config stuff
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r1772 | perry | 2004-09-06 15:30:36 +0200 (Mon, 06 Sep 2004) | 1 line

added firewall restart to network stuff
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r1769 | raarts | 2004-09-06 13:49:11 +0200 (Mon, 06 Sep 2004) | 3 lines

Added getzapcfg so we can auto-detect all telephony hardware later on
moved some general functions to astium-functions.php

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r1764 | perry | 2004-09-06 11:38:40 +0200 (Mon, 06 Sep 2004) | 2 lines

Moved some general functions from asterisk.php to file astium-functions.php.

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r1754 | raarts | 2004-09-05 23:45:38 +0200 (Sun, 05 Sep 2004) | 2 lines

better use require_once instead of require

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r1752 | raarts | 2004-09-05 14:31:00 +0200 (Sun, 05 Sep 2004) | 2 lines

Removed dhcp.conf. Old module.

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r1751 | raarts | 2004-09-05 14:30:28 +0200 (Sun, 05 Sep 2004) | 7 lines

Split phonesetup.php into dhcp.php and snom???.php.
Moved snom???.php into astium-confweb package.
Also removed demo-perry.php, because it was used only
in testing.
Fix bug with repeatedly starting a transaction as astcfgd
removed opengroupware

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r1749 | raarts | 2004-09-04 17:24:03 +0200 (Sat, 04 Sep 2004) | 2 lines

Removed opengroupware functionality

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r1746 | perry | 2004-09-03 16:20:54 +0200 (Fri, 03 Sep 2004) | 1 line

Added some more general functions
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r1744 | perry | 2004-09-03 14:52:21 +0200 (Fri, 03 Sep 2004) | 1 line

Added a module for general astium functions, not is use yet
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r1743 | perry | 2004-09-03 14:51:49 +0200 (Fri, 03 Sep 2004) | 1 line

Added a module for general astium functions, not is use yet
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r1742 | perry | 2004-09-03 14:51:35 +0200 (Fri, 03 Sep 2004) | 1 line

Added a module for general astium functions, not is use yet
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r1740 | perry | 2004-09-03 11:51:14 +0200 (Fri, 03 Sep 2004) | 1 line

Fixed a syntax error
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r1739 | perry | 2004-09-03 11:49:20 +0200 (Fri, 03 Sep 2004) | 1 line

Added a function to delete file from a deleted SIP phone
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r1738 | perry | 2004-09-03 11:34:09 +0200 (Fri, 03 Sep 2004) | 1 line

Bug fixes
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r1737 | perry | 2004-09-03 11:23:29 +0200 (Fri, 03 Sep 2004) | 1 line

First commit of module phonesetup
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r1722 | raarts | 2004-09-01 12:51:41 +0200 (Wed, 01 Sep 2004) | 2 lines

Small but important bug: return correct companyid

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r1720 | perry | 2004-09-01 12:13:15 +0200 (Wed, 01 Sep 2004) | 1 line

Creating backup of development files.
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r1713 | perry | 2004-09-01 10:09:27 +0200 (Wed, 01 Sep 2004) | 1 line

added a test script
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r1678 | raarts | 2004-08-30 13:58:33 +0200 (Mon, 30 Aug 2004) | 3 lines

-j option should be uppercase
external_to_internal was wrongly generated

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r1676 | raarts | 2004-08-30 13:39:24 +0200 (Mon, 30 Aug 2004) | 2 lines

Implemented firewall support

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r1486 | raarts | 2004-08-24 17:23:19 +0200 (Tue, 24 Aug 2004) | 2 lines

Fix finding headers in right location

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r1415 | raarts | 2004-08-19 23:44:15 +0200 (Thu, 19 Aug 2004) | 2 lines

Added BuildRpm

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r1397 | raarts | 2004-08-19 16:56:58 +0200 (Thu, 19 Aug 2004) | 2 lines

Test commit, to see if mailing to nl-nbs works

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r1396 | raarts | 2004-08-19 16:39:21 +0200 (Thu, 19 Aug 2004) | 2 lines

Now builds correctly.

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r1391 | raarts | 2004-08-19 15:48:31 +0200 (Thu, 19 Aug 2004) | 3 lines

I am committing this, but it is not ready, but svn says i MUST
commit because I want to overwrite a file that I also moved. (Dig?)

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r1390 | raarts | 2004-08-19 15:12:20 +0200 (Thu, 19 Aug 2004) | 1 line

initial (unfinished) import
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