similar to: Problems with Asterisk 1.4.14 and Queue app

<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

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

I'm using Queue app with Asterisk 1.4.6 It was working 5 days without problems and then it crash. When I did #gdb asterisk core.xxx I see... #0 ast_senddigit_end (chan=0x0, digit=54 '6', duration=0) at channel.c:2691 #1 0xb780c7d5 in agent_answer (ast=0x925cb78) at chan_agent.c:398 #2 0x08081877 in ast_senddigit (chan=0x57, digit=54 '6') at channel.c:2692 #3 0x0805d20d in

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 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>

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

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:

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>

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

In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash with this message asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so: undefined symbol: PQescapeStringConn Is this a knowed error? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071201/a3aecc1d/attachment.htm

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:

I found this property in queue.conf ; Calls may be recorded using Asterisk's monitor resource ; This can be enabled from within the Queue application, starting recording ; when the call is actually picked up; thus, only successful calls are ; recorded, and you are not recording while people are listening to MOH. ; To enable monitoring, simply specify "monitor-format"; it will be

Hi! I want to have this behabior in my queue. When a call come in, if there are unavailable agents or and busy agents, the queue reject the call. Thanks ! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070711/a05ed825/attachment.htm

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 ! 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 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 think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. I've looked at the sample config files for 1.4 and nothing seems to jump out at me as to what

I cannot access my voicemail and get the following warning in my console: [Nov 25 10:26:43] WARNING[5628]: app.c:935 ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/8900/Old': File exists I have also noticed that Asterisk will crash several minutes later after this warning message. I am using the latest SVN 1.4 branch of Asterisk (Revision 48007) and