[Asterisk-Users] No Audio sent using playback cmd

Tue Apr 26 18:39:42 MST 2005

No errors at all.  Here is a sip debug.  what in the config files could 
prevent playback from working.  All sound files are GSM

Sip read:
INVITE sip:9009 at 208.41.254.125 SIP/2.0
Via: SIP/2.0/UDP 208.41.254.119:5060; 
branch=z9hG4bKAClkQjtHCQArDAAAAAAAAEpVGXuz0b7qRnBunQL+fOs_
Via: SIP/2.0/UDP 68.68.11.31:53996;branch=z9hG4bK-f4ea6ce7
From: 6262769000 <sip:6262769000 at 208.41.254.119>;tag=413d98edc38224cfo0
To: <sip:9009 at 208.41.254.119>
Call-ID: 50a569c5-a784c9d7 at 192.168.1.161
CSeq: 102 INVITE
Max-Forwards: 69
Proxy-Authorization: Digest 
username="6262769000 at sip.shelcomm.com",realm="sip.shelcomm.com",nonce="/O1uQjFNZZm0iZxLoUHo3USzsbg=",uri="sip:9009 at 208.41.254.119",algorithm=MD5,response="06cba48108291c51a2ba5859e5458135",qop=auth,nc=00000001,cnonce="5756cc01"
Contact: 6262769000 <sip:6262769000 at 68.68.11.31:53996>
Expires: 240
User-Agent: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 282
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Record-Route: <sip:208.41.254.119;lr;hash=sipd-0-2-2>

v=0
o=- 130649 130649 IN IP4 68.68.11.31
s=-
c=IN IP4 68.68.11.31
t=0 0
m=audio 54872 RTP/AVP 0 8 18 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

17 headers, 14 lines
Using latest request as basis request
Sending to 208.41.254.119 : 5060 (NAT)
Found no matching peer or user for '208.41.254.119:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 68.68.11.31:54872
Found description format PCMU
Found description format PCMA
Found description format G729a
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c 
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Looking for 9009 in default
list_route: hop: <sip:208.41.254.119;lr;hash=sipd-0-2-2>
list_route: hop: <sip:6262769000 at 68.68.11.31:53996>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.41.254.119:5060; 
branch=z9hG4bKAClkQjtHCQArDAAAAAAAAEpVGXuz0b7qRnBunQL+fOs_;received=208.41.254.119;rport=5060
Via: SIP/2.0/UDP 68.68.11.31:53996;branch=z9hG4bK-f4ea6ce7
From: 6262769000 <sip:6262769000 at 208.41.254.119>;tag=413d98edc38224cfo0
To: <sip:9009 at 208.41.254.119>;tag=as350e5228
Call-ID: 50a569c5-a784c9d7 at 192.168.1.161
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9009 at 208.41.254.125>
Content-Length: 0

 to 208.41.254.119:5060
    -- Executing Answer("SIP/208.41.254.119-089b2aa8", "") in new stack
We're at 208.41.254.125 port 41528
Answering with preferred capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.41.254.119:5060; 
branch=z9hG4bKAClkQjtHCQArDAAAAAAAAEpVGXuz0b7qRnBunQL+fOs_;received=208.41.254.119;rport=5060
Via: SIP/2.0/UDP 68.68.11.31:53996;branch=z9hG4bK-f4ea6ce7
Record-Route: <sip:208.41.254.119;lr;hash=sipd-0-2-2>
From: 6262769000 <sip:6262769000 at 208.41.254.119>;tag=413d98edc38224cfo0
To: <sip:9009 at 208.41.254.119>;tag=as350e5228
Call-ID: 50a569c5-a784c9d7 at 192.168.1.161
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9009 at 208.41.254.125>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 2330 2330 IN IP4 208.41.254.125
s=session
c=IN IP4 208.41.254.125
t=0 0
m=audio 41528 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 208.41.254.119:5060
    -- Executing Playback("SIP/208.41.254.119-089b2aa8", 
"telephone-in-your-pocket") in new stack
    -- Playing 'telephone-in-your-pocket' (language 'en')
asterisk1*CLI>

Sip read:
ACK sip:9009 at 208.41.254.125 SIP/2.0
Via: SIP/2.0/UDP 208.41.254.119:5060; 
branch=z9hG4bKAClkQjtHCQAsDAAAAAAAAO7a31fAtm9YYI3XmpeyH14_
Via: SIP/2.0/UDP 68.68.11.31:53996;branch=z9hG4bK-c795d158
From: 6262769000 <sip:6262769000 at 208.41.254.119>;tag=413d98edc38224cfo0
To: <sip:9009 at 208.41.254.119>;tag=as350e5228
Call-ID: 50a569c5-a784c9d7 at 192.168.1.161
CSeq: 102 ACK
Max-Forwards: 69
Proxy-Authorization: Digest 
username="6262769000 at sip.shelcomm.com",realm="sip.shelcomm.com",nonce="/O1uQjFNZZm0iZxLoUHo3USzsbg=",uri="sip:9009 at 208.41.254.125",algorithm=MD5,response="63f09665aa7af25c0df0b7a6db4e8545",qop=auth,nc=00000001,cnonce="5756cc01"
Contact: 6262769000 <sip:6262769000 at 68.68.11.31:53996>
User-Agent: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0

12 headers, 0 lines
asterisk1*CLI>

Sip read:
BYE sip:9009 at 208.41.254.125 SIP/2.0
Via: SIP/2.0/UDP 208.41.254.119:5060; 
branch=z9hG4bKAClkQjtHCQAtDAAAAAAAADRAijYtzRUbr0h3HNlvezg_
Via: SIP/2.0/UDP 68.68.11.31:53996;branch=z9hG4bK-44e76474
From: 6262769000 <sip:6262769000 at 208.41.254.119>;tag=413d98edc38224cfo0
To: <sip:9009 at 208.41.254.119>;tag=as350e5228
Call-ID: 50a569c5-a784c9d7 at 192.168.1.161
CSeq: 103 BYE
Max-Forwards: 69
Proxy-Authorization: Digest 
username="6262769000 at sip.shelcomm.com",realm="sip.shelcomm.com",nonce="/O1uQjFNZZm0iZxLoUHo3USzsbg=",uri="sip:9009 at 208.41.254.125",algorithm=MD5,response="5c12ad4047730b1e8630f0c4df546f5f",qop=auth,nc=00000002,cnonce="5756cc01"
User-Agent: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0

11 headers, 0 lines
Sending to 208.41.254.119 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.41.254.119:5060; 
branch=z9hG4bKAClkQjtHCQAtDAAAAAAAADRAijYtzRUbr0h3HNlvezg_;received=208.41.254.119;rport=5060
Via: SIP/2.0/UDP 68.68.11.31:53996;branch=z9hG4bK-44e76474
From: 6262769000 <sip:6262769000 at 208.41.254.119>;tag=413d98edc38224cfo0
To: <sip:9009 at 208.41.254.119>;tag=as350e5228
Call-ID: 50a569c5-a784c9d7 at 192.168.1.161
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:9009 at 208.41.254.125>
Content-Length: 0

 to 208.41.254.119:5060
  == Spawn extension (default, 9009, 2) exited non-zero on 
'SIP/208.41.254.119-089b2aa8'
Destroying call '50a569c5-a784c9d7 at 192.168.1.161'

Rod Bacon wrote:

> What errors are you seeing at the console?
>
> The only time I've ever had this problem was because I specified the 
> file extension in the filename.
>
> Eg.
>
> Playback(file.wav) is INCORRECT. Needs to be specified as Playback(file).
>
> Some more info may help to get your question answered!
>
>
>
>
> ----- Original Message ----- From: "Michael D Schelin" 
> <mike at shelcomm.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users at lists.digium.com>
> Sent: Wednesday, April 27, 2005 10:55 AM
> Subject: [Asterisk-Users] No Audio sent using playback cmd
>
>
>> Hi All, I really need help on this. What would keep Asterisk from 
>> playing out audio files using the (Playback command) but I can play 
>> the busy tone . playtone(Congestion)  ??  I have verified this with 
>> ethereal and see the audio only going one way. In to Asterisk bun 
>> nothing coming out.   Because I can hear the audio with the play tone 
>> I know there is something preventing the playback cmd from working.
>>
>> Thanks
>> _
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>
>
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