Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect calls to internal office extensions (which still go through asterisk) OR voicemail 2) The other 20+ phones in the same office on the same network have 0 problems. Here's a SIP trace of the problem. yyy.yyy.yyy.yyy is the outside NAT IP xxx.xxx.xxx.xxx is the IP of my PBX dddddddddd is the dialed phone number sssssssssss is the source phone number The peculiar thing is that asterisk sends an OK in response to an INVITE, then the phone sends back an ACK, which asterisk seems to ignore because it retransmits the OK message again Then eventually the phone gives up and sends a BYE message. -- James <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> INVITE sip:[email protected] SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: [email protected]^m CSeq: 101 INVITE^M Max-Forwards: 70^M Contact: "sss-sss-ssss" ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 6363534 6363534 IN IP4 10.1.24.145^M s=-^M c=IN IP4 10.1.24.145^M t=0 0^M m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:2 G726-32/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:18 G729a/8000^M a=rtpmap:96 G726-40/8000^M a=rtpmap:97 G726-24/8000^M a=rtpmap:98 G726-16/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:20^M a=sendrecv^M <-------------> <--- Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 407 Proxy Authentication Required^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as70a8455c^M Call-ID: [email protected]^m CSeq: 101 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6d2db4b7"^M Content-Length: 0^M <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> ACK sip:[email protected] SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as70a8455c^M Call-ID: [email protected]^m CSeq: 101 ACK^M Max-Forwards: 70^M Contact: "sss-sss-ssss" ^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 0^G ^M <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> INVITE sip:[email protected] SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: [email protected]^m CSeq: 102 INVITE^M Max-Forwards: 70^M Proxy-Authorization: Digest username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:[email protected]",algorithm=MD5,response= Contact: "sss-sss-ssss" ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M Content-Type: application/sdp^M ^M v=0^M o=- 6363534 6363534 IN IP4 10.1.24.145^M s=-^M c=IN IP4 10.1.24.145^M t=0 0^M m=audio 16458 RTP/AVP 0 2 4 8 18 96 97 98 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:2 G726-32/8000^M a=rtpmap:4 G723/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:18 G729a/8000^M a=rtpmap:96 G726-40/8000^M a=rtpmap:97 G726-24/8000^M a=rtpmap:98 G726-16/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=ptime:20^M a=sendrecv^M <-------------> <--- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 100 Trying^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: [email protected]^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Length: 0^M ^M <------------> <--- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 183 Session Progress^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Type: application/sdp^M Content-Length: 264^M ^M v=0^M o=root 32147 32147 IN IP4 xxx.xxx.xxx.xxx^M s=session^M c=IN IP4 xxx.xxx.xxx.xxx^M t=0 0^M m=audio 19536 RTP/AVP 0 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M a=ptime:20^M a=sendrecv^M <------------> <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> INFO sip:[email protected] SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-234dc2a4^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: [email protected]^m CSeq: 103 INFO^M Max-Forwards: 70^M Proxy-Authorization: Digest username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:[email protected]",algorithm=MD5,response= User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 24^M Content-Type: application/dtmf-relay^M ^M Signal=#^M Duration=100^M <-------------> <--- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-234dc2a4;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 103 INFO^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Length: 0^M ^M <------------> <--- Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 180 Ringing^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Length: 0^M ^M <------------> OPTIONS sip:[email protected]:7388 SIP/2.0^M Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK481375ee;rport^M From: "Unknown" ;tag=as1e5e0912^M To: ^M Contact: ^M Call-ID: [email protected]^m CSeq: 102 OPTIONS^M User-Agent: Asterisk PBX^M Max-Forwards: 70^M Date: Fri, 22 May 2009 16:49:47 GMT^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Content-Length: 0^M ^M --- [May 22 09:49:47] VERBOSE[32177] logger.c: <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 200 OK^M To: ;tag=6bb2ad0e65f932fi0^M From: "Unknown" ;tag=as1e5e0912^M Call-ID: [email protected]^m CSeq: 102 OPTIONS^M Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK481375ee^M Server: Linksys/SPA942-6.1.3(a)^M Content-Length: 0^M Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER^M Supported: replaces^M ^M <-------------> <--- Reliably Transmitting (NAT) to yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Type: application/sdp^M Content-Length: 264^M ^M v=0^M o=root 32147 32148 IN IP4 xxx.xxx.xxx.xxx^M s=session^M c=IN IP4 xxx.xxx.xxx.xxx^M t=0 0^M m=audio 19536 RTP/AVP 0 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M a=ptime:20^M a=sendrecv^M <------------> [May 22 09:49:52] VERBOSE[32177] logger.c: <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> ACK sip:[email protected] SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-21970f9d^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 102 ACK^M Max-Forwards: 70^M Proxy-Authorization: Digest username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:[email protected]",algorithm=MD5,response= Contact: "sss-sss-ssss" ^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 0^M ^M <-------------> Retransmitting #1 (NAT) to yyy.yyy.yyy.yyy:24050: SIP/2.0 200 OK^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-3d87585d;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 102 INVITE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Contact: ^M Content-Type: application/sdp^M Content-Length: 264^M ^M v=0^M o=root 32147 32148 IN IP4 xxx.xxx.xxx.xxx^M s=session^M c=IN IP4 xxx.xxx.xxx.xxx^M t=0 0^M m=audio 19536 RTP/AVP 0 8 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:8 PCMA/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M a=ptime:20^M a=sendrecv^M <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> ACK sip:[email protected] SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-21970f9d^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 102 ACK^M Max-Forwards: 70^M Proxy-Authorization: Digest username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:[email protected]",algorithm=MD5,response= Contact: "sss-sss-ssss" ^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 0^M ^M [ RETRANSMIT ABOVE 6 TIMES ] <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> BYE sip:[email protected] SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-18e57808^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 104 BYE^M Max-Forwards: 70^M Proxy-Authorization: Digest username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:[email protected]",algorithm=MD5,response="5090 User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 0^M ^M <-------------> <--- Transmitting (no NAT) to yyy.yyy.yyy.yyy:24050 ---> SIP/2.0 481 Call leg/transaction does not exist^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-18e57808;received=yyy.yyy.yyy.yyy^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ;tag=as30846812^M Call-ID: [email protected]^m CSeq: 104 BYE^M User-Agent: Asterisk PBX^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY^M Supported: replaces^M Content-Length: 0^M ^M <------------> _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
