similar to: Asterisk on Fedora Core 4

Hello All, How can I SET EXTENSION from context? This is my context: - [docall-usa] exten => _NXXNXXXXXX,1,Answer exten => _NXXNXXXXXX,n,Set() ; <<What do I need to set here>> exten => _NXXNXXXXXX,n,DeadAGI(dousacall.php|1) exten => _NXXNXXXXXX,n,Hangup I need to add 1 in front of ${EXTEN} and then send the call to dousa.php. Set(CALLERID(number)=1${EXTEN}) will set

The Asterisk development team has released version 1.4.11. This version contains numerous bug fixes. One of these is for a security issue in chan_sip. The issue is that SIP dialog history was being stored in memory regardless if the option for this was turned on or off. This could be abused to cause a system using chan_sip to run out of memory. The security issue is documented in

The Asterisk development team has released version 1.4.11. This version contains numerous bug fixes. One of these is for a security issue in chan_sip. The issue is that SIP dialog history was being stored in memory regardless if the option for this was turned on or off. This could be abused to cause a system using chan_sip to run out of memory. The security issue is documented in

Hi, is anyone on the list using the Siemens Openstage phones together with asterisk? If yes, is it possible to use the programmable keys of these phones together with Asterisk? Thanks for any hints, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

Hello all, After installing Asterisk, i have installed the docs by "make progdocs". But i don't know where to locate this documentation. please Help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm

Hi List; If I configured one SIP account or one IAX account [sipuser1] or [iaxuser1] then how many calls can be originate/terminate using the same account [sipuser1] or [iaxuser1]? In other words, can 10 IP Phones (users) do a calls via Asterisk using the same account (SIP or IAX2)? If yes, how can I control the number of calls per account? Regards Bilal

Hi all, does anybody know any way that when it run "reload app_queue" in the asterisk cli it don't lose the informations from "show queue (queue name)" ? I'm passing for this trouble, because I need this informations (http://www.voip-info.org/wiki/index.php?page=asterisk+cli+command+show+queue) that asterisk cli command "show queue (queue name)" show me

Can anyone help with BLF for Snom 300s ? (Asterisk 1.4.10.1) I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing

Hi all, I am seeing on my logs this message: Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 (repeated much more then what I will show here). I see that it comes from static void* do_monitor(void *data) in chan_zap.c, but I do not understand what does it

Hi List; I noticed that if I disabled secret in the context by placing ( ; ) before it, then at the asterisk the log will be: -- Registered SIP 'bilal_sip' at 0.0.0.0 port 5060 expired The IP address of the endpoint was not captured!!! Why? If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution?

hi folks, I've built a simple PHP-script utilizing the AGI-interface. in extensions.conf I trigger the script and pass a single value as first argument: exten => h,1,DeadAGI(process.php|${Enter}) On the Asterisk-console, I can actually see that the script is called correctly (something like "DeadAGI(process.php|1234)"). However, when I read stdin in the PHP script, I receive

Hi List; In the outbound, I read in the documents the Wildcard match "by using the . (period)", but I did not understand how Wildcard will work (like what)? As I know that Wildcard is a term used with the Diguim TDM card (FXO and FXS), so what is the relation between such cards and the matching in the dial plan? Any help? Regards Bilal

I am trying to use PHP to reload the extensions in an Asterisk installation. I keep getting this error: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) when I run the script by visiting the URL; however, if I run the script from the command line, it runs just fine (works perfect, actually). I think it is permissions related. Does anyone have any ideas? <php

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

Hi: I want to use asterisk1.2 but I don't know which version of asterisk1.2 and zaptel1.2 is best.Please offer me one version of asterisk and zaptel and libpri.How about asterisk1.2.24 and zaptel1.2.20.1 and libpri1.2.5?And do they work togather well? Best regards. --------------------------------- Pinpoint customers who are looking for what you sell. -------------- next part

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!