Re: [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()
Tobias Jönsson wrote: Sorry, I did not know these american specialities. I just noticed in Larry's PRI debug info that he received a STATUS message during the waiting, so I thought that the waiting could lead to some kind of timeout at the telco. In EuroISDN the callerid always come in first SETUP message and so it did in Larry's pri debug. The calling number _is_ delivered in the SETUP message; what is not delivered (in National ISDN-2) is the calling name. That comes later in a FACILITY message, and if you Dial() an extension before it has arrived, the destination phone won't see the calling name. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio on PRI channel answered by Playback()orMeetMe()
On Fri, 27 Aug 2004, Larry Shields wrote:
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.
-- 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
Why do you start with a Wait statement? Just answer the line immediately
if you want to do that, or you should at least put a Ringing before the
first wait statement if you want the caller to hear a ringing tone before
you answer.
--
Regards,
Tobias Jönsson, Lund SE___
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RE: [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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Jackson
Sent: Friday, August 27, 2004 11:31 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No audio on PRI channel answered by
Playback()orMeetMe()
>-Original Message-
>From: Larry Shields [mailto:[EMAIL PROTECTED]
>Sent: Friday, August 27, 2004 12:20 PM
>To: [EMAIL PROTECTED]
>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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RE: [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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert Jackson
Sent: Friday, August 27, 2004 11:31 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] No audio on PRI channel answered by
Playback()orMeetMe()
>-Original Message-
>From: Larry Shields [mailto:[EMAIL PROTECTED]
>Sent: Friday, August 27, 2004 12:20 PM
>To: [EMAIL PROTECTED]
>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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RE: [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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Larry Shields
Sent: Friday, August 27, 2004 9:20 AM
To: [EMAIL PROTECTED]
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
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RE: [Asterisk-Users] No audio on PRI channel answered by Playback() orMeetMe()
>-Original Message- >From: Larry Shields [mailto:[EMAIL PROTECTED] >Sent: Friday, August 27, 2004 12:20 PM >To: [EMAIL PROTECTED] >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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
