similar to: Asterisk Debugging

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

Hello! I'm tryuing to bring up an R2 connection but eventhough I've followed the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems to be missing. When an incomming call is generated I get: Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 <- 0001 [1/ 1/Idle /Idle ] Nov 24 06:01:17 WARNING[-197416016]:

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

how to adjust voice volume for sipura 2000 and cisco ata186?

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is when dialing to the Harris PBX, it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play a prompt as soon as it is picked up, the promt is beign played as

I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created from regexten in sip.conf, and I do an 'extensions reload', I lose all the priority 1 NoOps! This can't be right... this means that in a production environment, if you make a change to your dialplan and do an 'extensions reload', you lose your ability to terminate calls to phones on this system.

Ok, I can't quite figure out how to make this work. I want to simultaneously log everything for facility local5 in a local file and a remote syslog-ng server. local7 is working fine getting the boot.log log entries transferred over to the syslog-ng server, but not so much with local5. Local logging of local5.* on the client is working fine now. SERVER: /etc/syslog-ng/syslog-ng.conf

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug.

It seems that when you use Realtime static and possibly realtime realtime for sip users, that Asterisk fails to create the regexten context for DUNDi. Someone else had the same problem back in July. Doesn't look like they ever had a resolution. <http://lists.digium.com/pipermail/asterisk-users/2006-July/160105.html>

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm

This is a continuation of the thread: "yum update fails: CentOS 5.2". I installed a fresh Centos 5.2 and ran yum update. It ran correctly, except for these issues: 1. As previously mentioned, after deciding what it is going to do, It doesn't ask if its ok. But if I type y<enter> it reports: Is this ok [y/N]: Downloading Packages: 2. During the long period it

I don't know if this is a recent issue... When the read() application is given a file that does not exist, it aborts the ENTIRE dial plan. That can't be right. Playback() and Background() don't do this. Couldn't find a bug in mantis for it... [Apr 26 17:20:12] VERBOSE[14611] logger.c: -- Executing [s@macro-VMDepositIntro:5] Read("SIP/xx.yy.69.210-009c3f40",

Anyone know if there's a way to match a dialplan extension, execute some code, say set a variable, and then continue with the dialplan? I want to set a variable when the dialplan flows beyond a certain context. This would be a great feature. Doug.

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten =>