Hi All, I have experiancing strenge issue with my production Asterisk system. I'm using asterisk vertion 1.4.28 installed cent OS 5. Issue decription. I have SIP trunk from local carrier to their hosted PBX( broadsoft). Out going calls over this trunk working fine and I can make a conversation with landlines and mobiles but incomming not working. I can see calls are hitting my IVR but no audio. This system worked till yesterday without any issue. Please find attached SIP traces from received from my carrier and what they are saying is they not receiving proper information on SIP 200 message. I'm attached my asterisk system traces also named asterisk SIP log. Please looking to this and provide me help ASAP. Thanks & Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110623/0451b418/attachment.htm> -------------- next part -------------- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.1.36:5061;branch=z9hG4bKooi3r32663oonttllrzts6lio Call-ID: 63kothjzr4m4ks2jsimktm64ihrsrhlk at SoftX3000 From: <sip:717747766 at 10.10.1.36;user=phone>;tag=t4h4mk46-CC-24 To: <sip:600 at 10.20.1.66;user=phone>;tag=aprqtqg8kq3-mt4au32000020 CSeq: 1 CANCEL -------------- next part -------------- --- Retransmitting #6 (NAT) to 10.8.55.194:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.8.55.194:5060;branch=z9hG4bKrjogsq209000rjk8s-3.1;received=10.8.55.194 From: <sip:773208775 at 10.8.55.194;user=phone>;tag=SD5j4u701-or643njn-CC-23 To: <sip:600 at 10.94.0.45;user=phone>;tag=as18171f50 Call-ID: SD5j4u701-8b8b9bbad4be07ce8971c1f3532859ab-ag220u0 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:600 at 10.94.0.45> ontent-Type: application/sdp Content-Length: 304 v=0 o=root 29336 29336 IN IP4 10.94.0.45 s=session c=IN IP4 10.94.0.45 t=0 0 m=audio 10802 RTP/AVP 8 0 18 97 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- elastix*CLI> <--- SIP read from 194.78.20.125:1092 ---> REGISTER sip:202.124.179.130 SIP/2.0 From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: "Tweco Wim"<sip:1103 at 202.124.179.130> Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127 CSeq: 1860 REGISTER Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295 Max-Forwards: 70 Supported: replaces,net2phone User-Agent: LGE LIP 6812 v1.1.31s SN/00405A181B6A Contact: <sip:1103 at 194.78.20.125:1093> Expires: 60 Authorization: Digest username="1103",realm="asterisk",nonce="0dcbdd61",uri="sip:202.124.179.130",response="8ff1c4bfbd6ad4a9dc5db70640dfc431",algorithm=MD5 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 194.78.20.125 : 1092 (NAT) <--- Transmitting (NAT) to 194.78.20.125:1092 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295;received=194.78.20.125 From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: "Tweco Wim"<sip:1103 at 202.124.179.130> Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127 CSeq: 1860 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to 194.78.20.125:1092 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbe-212749c0-7fc98295;received=194.78.20.125 From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=as5c46c474 Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127 CSeq: 1860 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b3b16d9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127' in 32000 ms (Method: REGISTER) elastix*CLI> <--- SIP read from 194.78.20.125:1092 ---> REGISTER sip:202.124.179.130 SIP/2.0 From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: "Tweco Wim"<sip:1103 at 202.124.179.130> Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127 CSeq: 1861 REGISTER Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2 Max-Forwards: 70 Supported: replaces,net2phone User-Agent: LGE LIP 6812 v1.1.31s SN/00405A181B6A Contact: <sip:1103 at 194.78.20.125:1093> Expires: 60 Authorization: Digest username="1103",realm="asterisk",nonce="3b3b16d9",uri="sip:202.124.179.130",response="1d39bc50536a47031997a5c40cd65402",algorithm=MD5 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 194.78.20.125 : 1092 (NAT) <--- Transmitting (NAT) to 194.78.20.125:1092 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2;received=194.78.20.125 From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: "Tweco Wim"<sip:1103 at 202.124.179.130> Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127 CSeq: 1861 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <------------> Reliably Transmitting (NAT) to 194.78.20.125:1092: OPTIONS sip:1103 at 194.78.20.125:1093 SIP/2.0 Via: SIP/2.0/UDP 202.124.179.130:5060;branch=z9hG4bK70e928aa;rport From: "Unknown" <sip:Unknown at 202.124.179.130>;tag=as4531a54a To: <sip:1103 at 194.78.20.125:1093> Contact: <sip:Unknown at 202.124.179.130> Call-ID: 64502bb31876dd3858d19c3f5470f727 at 202.124.179.130 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 22 Jun 2011 22:50:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- elastix*CLI> <--- Transmitting (NAT) to 194.78.20.125:1092 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 194.78.20.125:1093;branch=z9hG4bK-87cbf-21274a88-2be0b1a2;received=194.78.20.125 From: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=94cc0830-a20007f-13c4-7cc81-4636a78b-7cc81 To: "Tweco Wim"<sip:1103 at 202.124.179.130>;tag=as5c46c474 Call-ID: 3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127 CSeq: 1861 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 60 Contact: <sip:1103 at 194.78.20.125:1093>;expires=60 Date: Wed, 22 Jun 2011 22:50:33 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3131303300-aabb-5b49-0405a181b6a-0-3b at 10.32.0.127' in 32000 ms (Method: REGISTER) elastix*CLI> <--- SIP read from 194.78.20.125:1092 ---> SIP/2.0 200 OK From: "Unknown"<sip:Unknown at 202.124.179.130>;tag=as4531a54a To: <sip:1103 at 194.78.20.125:1093>;tag=94cbf448-a20007f-13c4-87cbf-186a602e-87cbf Call-ID: 64502bb31876dd3858d19c3f5470f727 at 202.124.179.130 CSeq: 102 OPTIONS Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,SUBSCRIBE,OPTIONS Via: SIP/2.0/UDP 202.124.179.130:5060;rport=5060;branch=z9hG4bK70e928aa Supported: replaces Content-Type: application/sdp Content-Length: 258 v=0 o=LGEIPP 0 0 IN IP4 10.32.0.127 s=SIP Call c=IN IP4 194.78.20.125 t=0 0 m=audio 23000 RTP/AVP 18 4 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (10 headers 12 lines) --- Really destroying SIP dialog '64502bb31876dd3858d19c3f5470f727 at 202.124.179.130' Method: OPTIONS elastix*CLI> <--- SIP read from 10.94.0.18:5974 ---> <-------------> -- Executing [h at ivr-3:1] Hangup("SIP/Suntel OUT-0000002c", "") in new stack == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/Suntel OUT-0000002c' Really destroying SIP dialog 'SD5j4u701-8b8b9bbad4be07ce8971c1f3532859ab-ag220u0' Method: INVITE <--- SIP read from 10.8.55.194:5060 ---> INVITE sip:600 at 10.94.0.45;user=phone SIP/2.0^M Via: SIP/2.0/UDP 10.8.55.194:5060;branch=z9hG4bKrjogsq209000rjk8s-3.1^M Call-ID: SD5j4u701-8b8b9bbad4be07ce8971c1f3532859ab-ag220u0^M From: <sip:773208775 at 10.8.55.194;user=phone>;tag=SD5j4u701-or643njn-CC-23^M To: <sip:600 at 10.94.0.45;user=phone>^M CSeq: 1 INVITE^M Contact: <sip:773208775 at 10.8.55.194:5060;user=phone;transport=udp>^M Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER^M User-Agent: Huawei SoftX3000 V300R010^M Supported: 100rel^M Max-Forwards: 69^M Content-Length: 274^M Content-Type: application/sdp^M ^M v=0^M o=HuaweiSoftX3000 11035821 11035821 IN IP4 10.8.55.194^M s=Sip Call^M c=IN IP4 10.8.55.194^M t=0 0^M m=audio 39858 RTP/AVP 8 0 18 97^M a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:18 G729/8000^M a=rtpmap:97 telephone-event/8000^M a=fmtp:97 0-15^M a=fmtp:18 annexb=yes^M <-------------> [Jun 23 04:20:11] VERBOSE[29370] logger.c: --- (13 headers 12 lines) --- [Jun 23 04:20:11] VERBOSE[29370] logger.c: Ignoring this INVITE request [Jun 23 04:20:11] VERBOSE[29370] logger.c: <--- Transmitting (NAT) to 10.8.55.194:5060 ---> SIP/2.0 100 Trying^M Via: SIP/2.0/UDP 10.8.55.194:5060;branch=z9hG4bKrjogsq209000rjk8s-3.1;received=10.8.55.194^M