Robert Jackson
2004-Aug-27 09:31 UTC
[Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe()
>-----Original Message----- >From: Larry Shields [mailto:LJ.Shields@Verizon.net] >Sent: Friday, August 27, 2004 12:20 PM >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] No audio on PRI channel answered byPlayback() orMeetMe()>If I assign the DID to ring extension SIP/2000 and then after time-outsend>it to MeetMe() or Playback() it works and the caller hears the .gsmfile.>Any assistance in solving this problem is appreciated. > >[nec_pri] >; Digital PRI from the NEAX2400 > >exten => 2688,1,Wait,3 >exten => 2688,2,MeetMe,|Mps >exten => 2688,3,Hangup >I had a similar problem with my system, and I was able to fix the problem by executing Answer before I entered any other applications. Using your previous example: exten => 2688,1,Answer exten => 2688,2,Wait,3 exten => 2688,3,MeetMe,|Mps exten => 2688,4,Hangup Hope this helps, Robert Jackson
Larry Shields
2004-Aug-27 12:48 UTC
[Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()
Robert,
Thanks for the reply. I tried that initially and it did not work. To verify
I went back and tried again. It answers and still no sound is heard. From
the CLI I can see it answer and ask for "conf-getconfno" three times
before
executing the hangup... But no sound. Yet if I point the DID to a SIP
extension it rings, upon answer there is 2-way speech path. Any other
ideas?
-- Accepting call from '8541' to '2688' on channel 0/2, span
1
-- Executing Wait("Zap/2-1", "3") in new stack
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Wait("Zap/2-1", "1") in new stack
-- Executing MeetMe("Zap/2-1", "|Mps") in new stack
-- Playing 'conf-getconfno' (language 'en')
-- Playing 'conf-getconfno' (language 'en')
-- Playing 'conf-getconfno' (language 'en')
-- Executing Hangup("Zap/2-1", "") in new stack
== Spawn extension (nec_pri, 2688, 5) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Jackson
Sent: Friday, August 27, 2004 11:31 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] No audio on PRI channel answered by
Playback()orMeetMe()
>-----Original Message-----
>From: Larry Shields [mailto:LJ.Shields@Verizon.net]
>Sent: Friday, August 27, 2004 12:20 PM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] No audio on PRI channel answered by
Playback() orMeetMe()
>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.
>
>[nec_pri]
>; Digital PRI from the NEAX2400
>
>exten => 2688,1,Wait,3
>exten => 2688,2,MeetMe,|Mps
>exten => 2688,3,Hangup
>
I had a similar problem with my system, and I was able to fix the
problem by executing
Answer before I entered any other applications.
Using your previous example:
exten => 2688,1,Answer
exten => 2688,2,Wait,3
exten => 2688,3,MeetMe,|Mps
exten => 2688,4,Hangup
Hope this helps,
Robert Jackson
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Larry Shields
2004-Aug-29 13:40 UTC
[Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()
Robert,
Thanks for the reply. I tried that initially and it did not work. To verify
I went back and tried again. It answers and still no sound is heard. From
the CLI I can see it answer and ask for "conf-getconfno" three times
before
executing the hangup... But no sound. Yet if I point the DID to a SIP
extension it rings, upon answer there is 2-way speech path. Any other
ideas?
-- Accepting call from '8541' to '2688' on channel 0/2, span
1
-- Executing Wait("Zap/2-1", "3") in new stack
-- Executing Answer("Zap/2-1", "") in new stack
-- Executing Wait("Zap/2-1", "1") in new stack
-- Executing MeetMe("Zap/2-1", "|Mps") in new stack
-- Playing 'conf-getconfno' (language 'en')
-- Playing 'conf-getconfno' (language 'en')
-- Playing 'conf-getconfno' (language 'en')
-- Executing Hangup("Zap/2-1", "") in new stack
== Spawn extension (nec_pri, 2688, 5) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Jackson
Sent: Friday, August 27, 2004 11:31 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] No audio on PRI channel answered by
Playback()orMeetMe()
>-----Original Message-----
>From: Larry Shields [mailto:LJ.Shields@Verizon.net]
>Sent: Friday, August 27, 2004 12:20 PM
>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] No audio on PRI channel answered by
Playback() orMeetMe()
>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.
>
>[nec_pri]
>; Digital PRI from the NEAX2400
>
>exten => 2688,1,Wait,3
>exten => 2688,2,MeetMe,|Mps
>exten => 2688,3,Hangup
>
I had a similar problem with my system, and I was able to fix the
problem by executing
Answer before I entered any other applications.
Using your previous example:
exten => 2688,1,Answer
exten => 2688,2,Wait,3
exten => 2688,3,MeetMe,|Mps
exten => 2688,4,Hangup
Hope this helps,
Robert Jackson
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users