[asterisk-users] No response to our critical packet problem

Martin asterisklist at callthem.info
Fri May 22 13:25:24 CDT 2009


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 <jlamanna at gmail.com> 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:dddddddddd at xxx.xxx.xxx.xxx 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: c4560330-de7ca29d at 10.1.24.145^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: c4560330-de7ca29d at 10.1.24.145^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:dddddddddd at xxx.xxx.xxx.xxx 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: c4560330-de7ca29d at 10.1.24.145^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:dddddddddd at xxx.xxx.xxx.xxx 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: c4560330-de7ca29d at 10.1.24.145^M
> CSeq: 102 INVITE^M
> Max-Forwards: 70^M
> Proxy-Authorization: Digest
> username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:dddddddddd at xxx.xxx.xxx.xxx",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: c4560330-de7ca29d at 10.1.24.145^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: c4560330-de7ca29d at 10.1.24.145^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:dddddddddd at xxx.xxx.xxx.xxx 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: c4560330-de7ca29d at 10.1.24.145^M
> CSeq: 103 INFO^M
> Max-Forwards: 70^M
> Proxy-Authorization: Digest
> username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:dddddddddd at xxx.xxx.xxx.xxx",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: c4560330-de7ca29d at 10.1.24.145^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: c4560330-de7ca29d at 10.1.24.145^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:ssssssssss at 10.1.24.145:7388 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: 47649b454714f359238cb6bb41eb75dd at xxx.xxx.xxx.xxx^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: 47649b454714f359238cb6bb41eb75dd at xxx.xxx.xxx.xxx^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: c4560330-de7ca29d at 10.1.24.145^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:dddddddddd at xxx.xxx.xxx.xxx 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: c4560330-de7ca29d at 10.1.24.145^M
> CSeq: 102 ACK^M
> Max-Forwards: 70^M
> Proxy-Authorization: Digest
> username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:dddddddddd at xxx.xxx.xxx.xxx",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: c4560330-de7ca29d at 10.1.24.145^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:dddddddddd at xxx.xxx.xxx.xxx 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: c4560330-de7ca29d at 10.1.24.145^M
> CSeq: 102 ACK^M
> Max-Forwards: 70^M
> Proxy-Authorization: Digest
> username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:dddddddddd at xxx.xxx.xxx.xxx",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:dddddddddd at xxx.xxx.xxx.xxx 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: c4560330-de7ca29d at 10.1.24.145^M
> CSeq: 104 BYE^M
> Max-Forwards: 70^M
> Proxy-Authorization: Digest
> username="ssssssssss",realm="asterisk",nonce="6d2db4b7",uri="sip:dddddddddd at xxx.xxx.xxx.xxx",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: c4560330-de7ca29d at 10.1.24.145^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
> <------------>
>
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