for some reason (someone would have to look deeper) your SIP peer sends ACK to 200 OK and Asterisk doesn't "get it"
so it retransmits 200 OK a couple times and then assumes there's noone there Martin On Fri, May 22, 2009 at 12:36 PM, James Lamanna <[email protected]> wrote: > 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 > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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