* You are viewing the archive for September 23rd, 2011

Question about Registrations

In Trunk, or earlier, is it possible to execute an AGI or any piece of
the Diaplan when a new peer registers successfully?

Set (MONITOR_FILENAME=……………..) for queuing recording calls

Hi All;

I noticed in the queues.conf the configuration for recording the calls in the queuing, and regarding to the filename (or any other parameter), it is written that I can determine the filename using the command:


But it should be called from the dialing plan, but really i did not understand how to call it from the dialing plan.

Well, for example this is my dialing plan to route for the queuing, how I can set the filename:


include => Internal

exten => s,1,Queue(CustomerSupport,t,,,120)
exten => s,2,Macro(voicemail,SIP/reception)

By the way, I need in the filename to appear the following:
The SIP username for the IP Phone that the call is routed for it
The calling number
The Time of the call

Actually for the outbound recording, I am using the below command (I mentioned it to declare the time format I am using and to declare how the filename to be named):

exten => _9ZXXXXXXXX,1,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)

So I hope if someone can help me to write the Set(MONITOR_FILENAME=foo) in a way to acheive same format the filename of the recorded outgoing calls (in addition that until now I am not able to know where I have to place the Set(MONITOR_FILENAME=foo).

For example, should I place it as following:
exten => s,1,Set(MONITOR_FILENAME=……………..)
exten => s,2,Queue(CustomerSupport,t,,,120)
exten => s,3,Macro(voicemail,SIP/reception)

Appreciate if someone help me plz.

Asterisk Now Available

The Asterisk Development Team announces the release of Asterisk

ThisĀ  release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk resolves several issues reported by the community and would have not been possible without your participation.
Thank you!

Please note that a significant numbers of changes and fixes have gone into features.c in this release (call parking, built-in transfers, call pickup, etc.).


Recently, we were notified that the mechanism included in our Asterisk source code releases to download and build support for the iLBC codec had stopped working correctly; a little investigation revealed that this occurred because of some changes on the ilbcfreeware.org website. These changes occurred as a result of Google’s acquisition of GIPS, who produced (and provided licenses for) the iLBC codec.

If you are a user of Asterisk and iLBC together, and you’ve already executed a license agreement with GIPS, we believe you can continue using iLBC with Asterisk. If you are a user of Asterisk and iLBC together, but you had not executed a license agreement with GIPS, we encourage you to research the situation and consult with your own legal representatives to determine what actions you may want to take (or avoid taking).

More information is available on the Asterisk blog:


The following is a sample of the issues resolved in this release:

* Added the ‘storesipcause’ option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function.

We’ve decided to disable this feature by default in future 1.8 versions. This would be an unexpected behavior change for anyone depending on that SIP_CAUSE update in their dialplan. Please refer to the asterisk-dev mailing list more information:


* Significant fixes and improvements to parking lots.

(Closes issues ASTERISK-17183, ASTERISK-17870, ASTERISK-17430, ASTERISK-17452, ASTERISK-17452, ASTERISK-15792. Reported by: David Cabrejos, Remi Quezada, Philippe Lindheimer, David Woolley, Mat Murdock. Patched by: rmudgett)

* Numerous issues have been reported for deadlocks that are caused by a blocking read in res_timing_timerfd on a file descriptor that will never be written to.

A change to Asterisk adds some checks to make sure that the timerfd is both valid and armed before calling read(). Should fix: ASTERISK-18142, ASTERISK-18197, ASTERISK-18166 and possibly others.

(In essence, this change should make res_timing_timerfd usable.)

* Resolve segfault when publishing device states via XMPP and not connected.

(Closes issue ASTERISK-18078. Reported, patched by: Michael L. Young. Tested by Jonathan Rose)

* Refresh peer address if DNS unavailable at peer creation.

(Closes issue ASTERISK-18000)

* Fix the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration.

(Closes issue ASTERISK-18496. Reported by Sean Darcy. Patched by Richard Mudgett)

* Remove unnecessary libpri dependency checks in the configure script.

(Closes issue ASTERISK-18535. Reported by Michael Keuter. Patched by Richard Mudgett)

* Update get_ilbc_source.sh script to work again.

(Closes issue ASTERISK-18412)

For a full list of changes in this release, please see the ChangeLog:


Thank you for your continued support of Asterisk!

Postgresql Reconnect on connection failure

Currently if asterisk loses its connection to the postgresql it does not attempt to reconnect. I have searched all over for a setting that would have asterisk attempt to reconnect but I can not find anything. Is there something I am missing?


sending fax using chan_capi


I tried to sendfax a text file, it was received successfully and the context
were in ascii format (readable form). As I tried to send a fax in .tiff
format (converted from pdf format using ghostscript), the context I received
in fax is in binary form. The dial plan is listed below;

exten => 100,1,Verbose(::::::::::::> Sending Dialogic Diva Fax…)
exten => 100,n,set(BeforeFaxTime=${EPOCH})
exten => 100,n,capicommand(sendfax,/tmp/out.tiff,732-XXX-XXXX,Dialogic Diva
Test Sendfax)
exten => 100,n,HangUp()
exten => h,1,set(ElapsedFaxTime=$[${EPOCH}-${BeforeFaxTime}])
exten =>

Please advice, how can I send fax in image format using T.30

Digium ISDN card

compare the prices between sangoma and digium pri boards!
Sangoma’s oards here in Germany are cheaper as the ones from digium.

if you need detailed help, you can contact me, and I can workout for you
something as well as helping you setting up your pbx!


Am 23.09.2011 15:01, schrieb michael k:
> Hi All,
> I am new in asterisk. In my office we have purchased ISDN
> pri line with 30 channels. we have more than 60 soft phone nodes and the
> internal asterisk connectivity between extensions are working with soft
> phones. Can anybody tell me which pci or pci express digium card can be
> used to connect my asterisk server and the ISDN pri line with 30
> channels ? Please assist me to do if possible
> Michael.k
> –
> _____________________________________________________________________
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