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<body class='hmmessage'><div dir='ltr'>hello,<BR> <BR>201.xxx.xxx.xxx = SIP Softphone which originates the call<BR>xxx.xxx.xxx.xxx = Asterisk server<BR>yyy.yyy.yyy.yyy = ITSP<BR> <BR><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>INVITE sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx><br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=c4446262<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport<br>Call-ID: 052fcf17df558f7b<br>CSeq: 1 INVITE<br>Contact: <sip:100@201.xxx.xxx.xxx:5060><br>Max-Forwards: 70<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>Content-Type: application/sdp<br>User-Agent: eyeBeam release 3007n stamp 17816<br>Content-Length: 233<BR>v=0<br>o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx<br>s=eyeBeam<br>c=IN IP4 201.xxx.xxx.xxx<br>t=0 0<br>m=audio 8612 RTP/AVP 18 101<br>a=alt:1 1 : 88385B47 00000038 201.xxx.xxx.xxx 8612<br>a=fmtp:101 0-15<br>a=rtpmap:101 telephone-event/8000<br>a=sendrecv<br><-------------><BR>[May 28 11:51:34] --- (12 headers 10 lines) ---<BR>[May 28 11:51:34] == Using SIP RTP CoS mark 5<BR>[May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (no NAT)<BR>[May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b<BR>[May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060<BR>[0K[May 28 11:51:34] <br><--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 401 Unauthorized<BR>v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<BR>f: 100<sip:100@xxx.xxx.xxx.xxx>;tag=c4446262<BR>t: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as22c91f20<BR>i: 052fcf17df558f7b<BR>CSeq: 1 INVITE<BR>Server: PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>k: replaces, timer<BR>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="75cff649"<BR>l: 0<BR> <BR><br><------------><BR>[May 28 11:51:34] Scheduling destruction of SIP dialog '052fcf17df558f7b' in 32000 ms (Method: INVITE)<br>[May 28 11:51:34] <br><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>ACK sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as22c91f20<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=c4446262<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport<br>Call-ID: 052fcf17df558f7b<br>CSeq: 1 ACK<br>Content-Length: 0<BR><-------------><br>[May 28 11:51:34] --- (7 headers 0 lines) ---<BR>[May 28 11:51:34] <br><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>INVITE sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx><br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=c4446262<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport<br>Call-ID: 052fcf17df558f7b<br>CSeq: 2 INVITE<br>Contact: <sip:100@201.xxx.xxx.xxx:5060><br>Max-Forwards: 70<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>Content-Type: application/sdp<br>User-Agent: eyeBeam release 3007n stamp 17816<br>Authorization: Digest username="100",realm="asterisk",nonce="75cff649",uri="sip:12127773456@xxx.xxx.xxx.xxx",response="ffeaf69c547dd3d1252e1bc7ab614fea",algorithm=MD5<br>Content-Length: 233<BR>v=0<br>o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx<br>s=eyeBeam<br>c=IN IP4 201.xxx.xxx.xxx<br>t=0 0<br>m=audio 8612 RTP/AVP 18 101<br>a=alt:1 1 : 88385B47 00000038 201.xxx.xxx.xxx 8612<br>a=fmtp:101 0-15<br>a=rtpmap:101 telephone-event/8000<br>a=sendrecv<br><-------------><br>[May 28 11:51:34] --- (13 headers 10 lines) ---<br>[May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (NAT)<br>[May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b<BR>[May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060<BR>[May 28 11:51:34] Found RTP audio format 18<BR>[May 28 11:51:34] Found RTP audio format 101<BR>[May 28 11:51:34] Found audio description format telephone-event for ID 101<BR>[May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)<BR>[May 28 11:51:34] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<BR>[May 28 11:51:34] Peer audio RTP is at port 201.xxx.xxx.xxx:8612<BR>[May 28 11:51:34] Looking for 12127773456 in asterisk (domain xxx.xxx.xxx.xxx)<BR>[May 28 11:51:34] list_route: hop: <sip:100@201.xxx.xxx.xxx:5060><BR>[May 28 11:51:34] <br><--- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 100 Trying<BR>v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<BR>f: 100<sip:100@xxx.xxx.xxx.xxx>;tag=c4446262<BR>t: <sip:12127773456@xxx.xxx.xxx.xxx><BR>i: 052fcf17df558f7b<BR>CSeq: 2 INVITE<BR>Server: PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>k: replaces, timer<BR>m: <sip:12127773456@xxx.xxx.xxx.xxx><BR>l: 0<BR> <BR><br><------------><BR>[May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "call.php")<BR>[May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php<BR>[May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456)<BR>[May 28 11:51:34] == Using SIP RTP CoS mark 5<BR><strong>[May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456</strong><br>[May 28 11:51:34] Scheduling destruction of SIP dialog <a href="mailto:'142182ef20750fda512f8d2b0b071ad6@xxx.xxx.xxx.xxx'">'142182ef20750fda512f8d2b0b071ad6@xxx.xxx.xxx.xxx'</a> in 32000 ms (Method: INVITE)<BR>[May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0)<BR>[May 28 11:51:34] -- <SIP/100-0000115f>AGI Script call.php completed, returning 0<BR>[May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-0000115f' status is 'CHANUNAVAIL'<br>[May 28 11:51:34] <br><--- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 ---><br>SIP/2.0 503 Service Unavailable<BR>v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060<BR>f: 100<sip:100@xxx.xxx.xxx.xxx>;tag=c4446262<BR>t: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as4e329d09<BR>i: 052fcf17df558f7b<BR>CSeq: 2 INVITE<BR>Server: PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>k: replaces, timer<BR>l: 0<BR> <BR><br><------------><BR>[May 28 11:51:34] <br><--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---><br>ACK sip:12127773456@xxx.xxx.xxx.xxx SIP/2.0<br>To: <sip:12127773456@xxx.xxx.xxx.xxx>;tag=as4e329d09<br>From: 100<sip:100@xxx.xxx.xxx.xxx>;tag=c4446262<br>Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport<br>Call-ID: 052fcf17df558f7b<br>CSeq: 2 ACK<br>Content-Length: 0<BR><-------------><BR>[May 28 11:51:34] --- (7 headers 0 lines) ---<BR>[May 28 11:51:34] -- Executing AGI("SIP/100-0000115f", "hangup.php")<br>[May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.php<BR>[May 28 11:51:34] -- <SIP/100-0000115f>AGI Script hangup.php completed, returning 0<BR> <BR>Thanks,<BR>Kamlesh<br> <BR><div>> From: EWieling@nyigc.com<br>> To: asterisk-users@lists.digium.com<br>> Date: Mon, 27 May 2013 11:51:53 -0400<br>> Subject: Re: [asterisk-users] G.729 codec in pass-thru mode<br>> <br>> Show us the sip debug for a failed call.<br>> <br>> -----Original Message-----<br>> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kamlesh Kumar<br>> Sent: Monday, May 27, 2013 2:20 AM<br>> To: asterisk-users@lists.digium.com<br>> Subject: [asterisk-users] G.729 codec in pass-thru mode<br>> <br>> Hello,<br>> Trying to use g729 in pass-thru mode.<br>> Call flow:<br>> 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.<br>> == Using SIP RTP CoS mark 5<br>> -- Executing [12127773456@default:1] AGI("SIP/100-00000000", "call.php") in new stack<br>> -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php<br>> -- AGI Script Executing Application: (Dial) Options: (SIP/xxx.xxx.xxx.xxx/12127773456)<br>> -- Couldn't call xxx.xxx.xxx.xxx/12127773456 == Everyone is busy/congested at this time (0:0/0/0)<br>> -- <SIP/100-00000000>AGI Script call.php completed, returning 0<br>> -- Auto fallthrough, channel 'SIP/100-00000000' status is 'CHANUNAVAIL'<br>> <br>> If I use, ulaw, call works fine.<br>> <br>> Thanks,<br>> Kamlesh<br>> <br>> <br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> http://www.asterisk.org/hello<br>> <br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> http://lists.digium.com/mailman/listinfo/asterisk-users<br></div> </div></body>
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