similar to: Outgoing agent´s calls

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn?t found wath opition configure this functionality. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071212/cfed2687/attachment.htm

Is a normal functionality? when I do #dahdi_cfg -vvvvvv In my Asterisk console shows this.... [Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I do this a lot of times...then [Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24

Is there any way to do this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120307/77764e4b/attachment.htm>

<asterisk-users@lists.digium.com>Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. This behavior still happend in 1.4.1 version. Thanks a lot. -------------- next

Hi ! I have a python script that create and move .call files to /var/spool/asterisk/outgoing Sometimes...(in this case after 500 successfull calls) Asterisk don?t make the calls and the .call files are in the "outgoing" forever... Any Ideas? I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior) In my python script I move .call files using ... import shutil

I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel:

I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081229/3545ab8c/attachment.htm

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

Hi! I need to use text to speech, what is the best application? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070829/bc69eb9d/attachment.htm

Hi! I have this errors compiling Asterisk 1.4.5 cdr_tds.c:86:2: warning: #warning "You have older TDS, you should upgrade!" cdr_tds.c: In function `tds_log': cdr_tds.c:213: error: too many arguments to function `tds_process_simple_query' cdr_tds.c: In function `mssql_connect': cdr_tds.c:326: error: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:326:

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>

Yamaban wrote: > On Tue, 22 Dec 2015 14:29, James Hogarth wrote: >> On 22 December 2015 at 10:33, Sylvain CANOINE wrote >>> ----- Mail original ----- >>>> De: "Marcelo Ricardo Leitner" >>>> ?: "centos" >>>> Envoy?: Lundi 21 D?cembre 2015 21:46:10 >>>> Objet: Re: [CentOS] Network services start before network is

I''ve gone down (hopefully the correct) route of breaking up my layout(s) into common/generic elements such as the html head etc, but when the page renders there is no consistency in indenting - it''s all over the place (I''ve de-tabbed everything with the same result). Is there a standard/existing way of pretty-printing output? I''ve searched without any real

Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 <- call1 queue1 <- call2 queue2 <- call3 queue2 <- call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could

Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT <*prosody rate*=\'.8\' >Hello World\, this is a test\,</*prosody* >|0|1 Would I use a scape character? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/7132be4c/attachment.htm

Hi, Some background: I would like to call my girlfriend over the internet. We are both behind a nat router and I want to avoid portmapping. I've heard that you can call someone behind a firewall (nat router) with the IAX protocol, but I'm not sure. The questions: Do I have to set up my own PBX asterisk server or are there any other (free) servers where I can register on and connect

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As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match