hello, 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> INVITE sip:[email protected] SIP/2.0 To: <sip:[email protected]> From: 100<sip:[email protected]>;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 1 INVITE Contact: <sip:[email protected]:5060> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Content-Length: 233 v=0 o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx s=eyeBeam c=IN IP4 201.xxx.xxx.xxx t=0 0 m=audio 8612 RTP/AVP 18 101 a=alt:1 1 : 88385B47 00000038 201.xxx.xxx.xxx 8612 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [May 28 11:51:34] --- (12 headers 10 lines) --- [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (no NAT) [May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b [May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060 [0K[May 28 11:51:34] <--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 401 Unauthorized v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 f: 100<sip:[email protected]>;tag=c4446262 t: <sip:[email protected]>;tag=as22c91f20 i: 052fcf17df558f7b CSeq: 1 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75cff649" l: 0
<------------> [May 28 11:51:34] Scheduling destruction of SIP dialog '052fcf17df558f7b' in 32000 ms (Method: INVITE) [May 28 11:51:34] <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> ACK sip:[email protected] SIP/2.0 To: <sip:[email protected]>;tag=as22c91f20 From: 100<sip:[email protected]>;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 1 ACK Content-Length: 0 <-------------> [May 28 11:51:34] --- (7 headers 0 lines) --- [May 28 11:51:34] <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> INVITE sip:[email protected] SIP/2.0 To: <sip:[email protected]> From: 100<sip:[email protected]>;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 2 INVITE Contact: <sip:[email protected]:5060> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="100",realm="asterisk",nonce="75cff649",uri="sip:[email protected]",response="ffeaf69c547dd3d1252e1bc7ab614fea",algorithm=MD5 Content-Length: 233 v=0 o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx s=eyeBeam c=IN IP4 201.xxx.xxx.xxx t=0 0 m=audio 8612 RTP/AVP 18 101 a=alt:1 1 : 88385B47 00000038 201.xxx.xxx.xxx 8612 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> [May 28 11:51:34] --- (13 headers 10 lines) --- [May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (NAT) [May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b [May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060 [May 28 11:51:34] Found RTP audio format 18 [May 28 11:51:34] Found RTP audio format 101 [May 28 11:51:34] Found audio description format telephone-event for ID 101 [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) [May 28 11:51:34] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 28 11:51:34] Peer audio RTP is at port 201.xxx.xxx.xxx:8612 [May 28 11:51:34] Looking for 12127773456 in asterisk (domain xxx.xxx.xxx.xxx) [May 28 11:51:34] list_route: hop: <sip:[email protected]:5060> [May 28 11:51:34] <--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 100 Trying v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 f: 100<sip:[email protected]>;tag=c4446262 t: <sip:[email protected]> i: 052fcf17df558f7b CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer m: <sip:[email protected]> l: 0 <------------> [May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "call.php") [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [May 28 11:51:34] Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE) [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) [May 28 11:51:34] -- <SIP/100-0000115f>AGI Script call.php completed, returning 0 [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-0000115f' status is 'CHANUNAVAIL' [May 28 11:51:34] <--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---> SIP/2.0 503 Service Unavailable v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 f: 100<sip:[email protected]>;tag=c4446262 t: <sip:[email protected]>;tag=as4e329d09 i: 052fcf17df558f7b CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer l: 0 <------------> [May 28 11:51:34] <--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---> ACK sip:[email protected] SIP/2.0 To: <sip:[email protected]>;tag=as4e329d09 From: 100<sip:[email protected]>;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 2 ACK Content-Length: 0 <-------------> [May 28 11:51:34] --- (7 headers 0 lines) --- [May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "hangup.php") [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php [May 28 11:51:34] -- <SIP/100-0000115f>AGI Script hangup.php completed, returning 0 Thanks, Kamlesh > From: [email protected] > To: [email protected] > Date: Mon, 27 May 2013 11:51:53 -0400 > Subject: Re: [asterisk-users] G.729 codec in pass-thru mode > > Show us the sip debug for a failed call. > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of Kamlesh Kumar > Sent: Monday, May 27, 2013 2:20 AM > To: [email protected] > Subject: [asterisk-users] G.729 codec in pass-thru mode > > Hello, > Trying to use g729 in pass-thru mode. > Call flow: > SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When > using G.729, call is not getting connected. Below is the extract from CLI. > == Using SIP RTP CoS mark 5 > -- Executing [12127773456@default:1] AGI("SIP/100-00000000", "call.php") in > new stack > -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php > -- AGI Script Executing Application: (Dial) Options: > (SIP/xxx.xxx.xxx.xxx/12127773456) > -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at > this time (0:0/0/0) > -- <SIP/100-00000000>AGI Script call.php completed, returning 0 > -- Auto fallthrough, channel 'SIP/100-00000000' status is 'CHANUNAVAIL' > > If I use, ulaw, call works fine. > > Thanks, > Kamlesh > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
