Larry Shields
2004-Aug-27 09:19 UTC
[Asterisk-Users] No audio on PRI channel answered by Playback() or MeetMe()
Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a single DID number that rings in from the NEC IPX on PRI Span 1, trunk group 1. If I assign the inbound DID to ring an extension on Asterisk, ie. SIP/2000, it in fact rings and when answered I have a complete 2-way voice path. If I change the destination of the inbound DID from SIP/2000 to MeetMe() or Playback(), Asterisk will answer and I can see from the CLI the .gsm file being played but there is no playback audio heard on the calling extension. If I assign the DID to ring extension SIP/2000 and then after time-out send it to MeetMe() or Playback() it works and the caller hears the .gsm file. Any assistance in solving this problem is appreciated. What follows are two examples from what I tried in extensions.conf: This works but is not desirable: [nec_pri] ; Digital PRI from the NEAX2400 exten => 2688,1,Wait,1 exten => 2688,2,Dial(SIP/2000,3,Tr) exten => 2688,3,Wait,1 exten => 2688,4,MeetMe,|Mps exten => 2688,5,Hangup This will answer, but there is no audible playback on the channel: [nec_pri] ; Digital PRI from the NEAX2400 exten => 2688,1,Wait,3 exten => 2688,2,MeetMe,|Mps exten => 2688,3,Hangup This is what is displayed from the CLI while the calling station is connected via PRI: -- Accepting call from '2502' to '2688' on channel 0/4, span 1 -- Executing Wait("Zap/4-1", "3") in new stack -- Executing MeetMe("Zap/4-1", "|Mps") in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Executing Hangup("Zap/4-1", "") in new stack == Spawn extension (nec_pri, 2688, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' MDBRIDGE*CLI> Thank you, --LJ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040827/2be9395d/attachment.htm
Scott Stingel
2004-Aug-27 09:38 UTC
[Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe()
You should be able to hear the audio - a sound card is not involved. Try inserting an "answer" command in the dialplan before you try to play something. Like Answer Wait (if you want) Playback Hangup Should work (using the proper dialplan commands) Regards Scott Stingel Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England www.evtmedia.com _____________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Larry Shields Sent: Friday, August 27, 2004 9:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe() Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a single DID number that rings in from the NEC IPX on PRI Span 1, trunk group 1. If I assign the inbound DID to ring an extension on Asterisk, ie. SIP/2000, it in fact rings and when answered I have a complete 2-way voice path. If I change the destination of the inbound DID from SIP/2000 to MeetMe() or Playback(), Asterisk will answer and I can see from the CLI the .gsm file being played but there is no playback audio heard on the calling extension. If I assign the DID to ring extension SIP/2000 and then after time-out send it to MeetMe() or Playback() it works and the caller hears the .gsm file. Any assistance in solving this problem is appreciated. What follows are two examples from what I tried in extensions.conf: This works but is not desirable: [nec_pri] ; Digital PRI from the NEAX2400 exten => 2688,1,Wait,1 exten => 2688,2,Dial(SIP/2000,3,Tr) exten => 2688,3,Wait,1 exten => 2688,4,MeetMe,|Mps exten => 2688,5,Hangup This will answer, but there is no audible playback on the channel: [nec_pri] ; Digital PRI from the NEAX2400 exten => 2688,1,Wait,3 exten => 2688,2,MeetMe,|Mps exten => 2688,3,Hangup This is what is displayed from the CLI while the calling station is connected via PRI: -- Accepting call from '2502' to '2688' on channel 0/4, span 1 -- Executing Wait("Zap/4-1", "3") in new stack -- Executing MeetMe("Zap/4-1", "|Mps") in new stack -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Playing 'conf-getconfno' (language 'en') -- Executing Hangup("Zap/4-1", "") in new stack == Spawn extension (nec_pri, 2688, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' MDBRIDGE*CLI> Thank you, --LJ