Sip read:
INVITE sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=413d98edc38224cfo0
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 69
Proxy-Authorization: Digest username="[EMAIL PROTECTED]",realm="sip.shelcomm.com",nonce="/O1uQjFNZZm0iZxLoUHo3USzsbg=",uri="sip:[EMAIL PROTECTED]",algorithm=MD5,response="06cba48108291c51a2ba5859e5458135",qop=auth,nc=00000001,cnonce="5756cc01"
Contact: 6262769000 <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=413d98edc38224cfo0
To: <sip:[EMAIL PROTECTED]>;tag=as350e5228
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]>;tag=413d98edc38224cfo0
To: <sip:[EMAIL PROTECTED]>;tag=as350e5228
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=413d98edc38224cfo0
To: <sip:[EMAIL PROTECTED]>;tag=as350e5228
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
Max-Forwards: 69
Proxy-Authorization: Digest username="[EMAIL PROTECTED]",realm="sip.shelcomm.com",nonce="/O1uQjFNZZm0iZxLoUHo3USzsbg=",uri="sip:[EMAIL PROTECTED]",algorithm=MD5,response="63f09665aa7af25c0df0b7a6db4e8545",qop=auth,nc=00000001,cnonce="5756cc01"
Contact: 6262769000 <sip:[EMAIL PROTECTED]:53996>
User-Agent: Sipura/SPA3000-2.0.13(GWg)
Content-Length: 0
12 headers, 0 lines asterisk1*CLI>
Sip read:
BYE sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=413d98edc38224cfo0
To: <sip:[EMAIL PROTECTED]>;tag=as350e5228
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
Max-Forwards: 69
Proxy-Authorization: Digest username="[EMAIL PROTECTED]",realm="sip.shelcomm.com",nonce="/O1uQjFNZZm0iZxLoUHo3USzsbg=",uri="sip:[EMAIL PROTECTED]",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:[EMAIL PROTECTED]>;tag=413d98edc38224cfo0
To: <sip:[EMAIL PROTECTED]>;tag=as350e5228
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
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 '[EMAIL PROTECTED]'
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" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]>
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 _ _______________________________________________ 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
_______________________________________________ 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
_______________________________________________ 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
