James Lamanna
2009-May-22 17:36 UTC
[asterisk-users] No response to our critical packet problem
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,responseContact: "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,responseUser-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,responseContact: "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,responseContact: "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 <------------>
Mark Michelson
2009-May-22 18:16 UTC
[asterisk-users] No response to our critical packet problem
James Lamanna 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 >I think I know what the problem is here. It's not the fault of the phone, but of Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#', specifically) to Asterisk. Asterisk only keeps track of the last incoming Cseq in a dialog, so once the INFO arrives, we no longer have any memory of the Cseq of the INVITE that the phone sent. Later, we send a 200 OK response for the INVITE. Then, when we receive the ACK from the phone, we drop it since it's Cseq is less than the latest Cseq we received in this dialog. As a result, Asterisk never realizes that it has received the ACK. Asterisk continues retransmitting a 200 OK to the phone and the phone dutifully keeps sending an ACK in response until Asterisk has retransmitted the maximum amount of times. There are a couple of potential ways of solving this issue. One is to add an Answer to your dialplan as the first priority. This way, the INVITE is completely answered before the phone ever sends any INFO requests. Another is to switch the phone away from using INFO to transmit DTMF. I would be willing to bet that the other phones on your network are not using INFO for transmission of DTMF, and so they are not experiencing the same issue. Mark Michelson
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 > <------------> > > _______________________________________________ > -- 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 >
James Lamanna
2009-May-22 20:49 UTC
[asterisk-users] No response to our critical packet problem
Hi Guys, I just wanted to let you all know that you were indeed correct, it was the SIP INFO '#' that was causing the problem. You'll pardon me, but I find this problem _utterly ridiculous_. I am running asterisk v1.4.18. Are there any asterisk versions that this is fixed on? Thanks. (Oh and please CC me, I'm reading in digest mode..) -- James On Fri, May 22, 2009 at 10:36 AM, 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 > <------------> >