1. Siptrunk'v helped me already. It workaround for me BUT there is a bug that create complex INVITE. Unfortunately I cant distinguish is it 3.10.2, 4.2.1 or (hardly) linksys bug.
2. I get your "more info" but - I prefer fishing-rod not fish itself. And I get it! And I really thank you. - Developers are in other maillist I suppose. - 0,5 Mb XML hardly welcome in hundreds mailboxes. I prefer a little less them much more. 7 сентября 2010 г. 14:48 пользователь Nikolay Kondratyev <[email protected]>написал: > Alexander, > > looks like you identified your problem yourself. That's great. > I think it's a good idea to use siptrunk to shorten sip messages. > > And let me focus your attention on the following: please provide more info, > when asking for help. > Don't make developers to ask you for the info. Try to provide it right > away. > As far as i understand merged xml file, suitable for sipviewer, is good > starting point for understanding what is happening. > > Rgds, > Nikolay. > > > ------------------------------ > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Александр Горбунов > *Sent:* Tuesday, September 07, 2010 10:09 AM > > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] Can not call thought sipXecs to sipXecs trunk > ifCaller ID assigned > > Thax Nikolay for pushing me to see log deeper. Let me describe the problem > in other way. > > First, i dont use trunk. Two systems connected by this article > http://wiki.sipfoundry.org/display/xecsuserV4r0/sipXecs+to+sipXecs+Calling > . > > Second,I investigate, that I can call to cisco 7940 phone and sip plugin > for miranda softphone, but cannot call to cisco 7911, linksys SPA922 and > PAP2T devices connected to 3.10.2 system. > > At last, I see in sipviewer, that 10.0.15.2 sends INVITE to linksys SPA922 > phone 4 times without any answer AND! linksys SPA922 say in debug log > "<151>TP Parser error: 6" on each INVITE. See INVITE packet late. > > Honestly to say I'm shocked by "<151>TP Parser error: 6". I'm planning to > use siptrunk to make INVITE simple after a cup of coffee. I'll be glad any > other advices and any suggestion why INVITE too complex for the phone. > > > > +++++++++++++++++++++++++++++++++++++ > Time: 2010-09-07T04:47:36.530860Z > Frame: 292 /tmp/trace.JVfz6454/_.sipXproxy.trace.xml:2660 > Source: atc.sibdistillery.local-SipXProxy > Dest: 10.0.15.233:5060 > > INVITE sip:[email protected]:5060 SIP/2.0 > Record-Route: <sip:10.0.15.2:5060 > ;lr;sipXecs-rs=%2Afrom%7EMDAxNDFjOGRkZjY2MDIwYzM0ZGNlOWVhLTYzMGY2ZWNj.400_authrules%2Aauth%7E%2126266882c9bb7f691d8243a5f7b2f64b> > Record-Route: <sip:10.0.20.2:5060 > ;lr;sipXecs-CallDest=STS;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMDAxNDFjOGRkZjY2MDIwYzM0ZGNlOWVhLTYzMGY2ZWNj.700_fromalias%2Aa%7EIkY0MCBMNDAiPHNpcDozODMzNjMzMzk5QGF0Yy5zbHZ6LnJ1c2FsY29ob2wubG9jYWw%27O3RhZz0%60.700_fromalias%2Aao%7ENTY%60.700_fromalias%2Ac%7EIkY0MCBMNDAiIDxzaXA6OTA0MEBhdGMuc2x2ei5ydXNhbGNvaG9sLmxvY2FsPjt0YWc9.700_fromalias%2Aco%7ENTE%60%21482e9895a44d81546b8af04aa5e69d28> > Via: SIP/2.0/UDP > 10.0.15.2;branch=z9hG4bK-sipXecs-045be37981176d6659b79eca04d107f2bfd9 > Via: SIP/2.0/TCP > 10.0.15.2;branch=z9hG4bK-sipXecs-0457cf18f53bdbff2dcb21627f1fece959d1~2d380cfb19086c4aa57a6b03eedb5ae3 > Via: SIP/2.0/UDP 10.0.20.2;branch=z9hG4bK-XX-64bf_nn8bGCZUsyevgN6krBWew > Via: SIP/2.0/TCP > 10.0.20.2;branch=z9hG4bK-XX-64bcKkZEU38Qd1`2bJDSq`i_Bw~l55cjTcuGEiwbJAWiIOaFQ > Via: SIP/2.0/UDP 10.0.20.253:5060;branch=z9hG4bK79f5ad0a > From: "F40 L40"<sip:[email protected] > >;tag=00141c8ddf66020c34dce9ea-630f6ecc > To: <sip:[email protected];user=phone> > Call-Id: [email protected] > Max-Forwards: 15 > Date: Tue, 07 Sep 2010 04:47:32 GMT > Cseq: 102 INVITE > User-Agent: Cisco-CP7940G/8.0 > Contact: <sip:[email protected]:5060;transport=udp;x-sipX-nonat> > Proxy-Authorization: Digest > username="9040",realm="atc.slvz.rusalcohol.local",uri="sip:[email protected] > ;user=phone",response="0efea332aeec31ecae1c12d9f8f75b12",nonce="4e3c1f4e5eff782288d684ff3686b2af4c85c3e4",cnonce="2bf78bbf",qop=auth,nc=00000001,algorithm=md5 > Expires: 7200 > Accept: application/sdp > Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE > Supported: replaces,join,norefersub > Content-Length: 274 > Content-Type: application/sdp > Content-Disposition: session;handling=optional > > v=0 > o=Cisco-SIPUA 27540 0 IN IP4 10.0.20.253 > s=SIP Call > t=0 0 > m=audio 28514 RTP/AVP 0 8 18 101 > c=IN IP4 10.0.20.253 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > > +++++++++++++++++++++++++++++++++++++ > > 2010/9/6 Nikolay Kondratyev <[email protected]> > >> Alexander, >> the info, you provided is not enough. >> The following, i guess, will be ok: >> Set at least INFO log level for all the processes. >> Provide two xml traces of the same failed call from both systems (see wiki >> how to trace call using sipviewer, these traces will contain sipx internal >> interprocess message flow). >> And two system 'snapshots' containing the same period of time. Use >> sipx-snapshot utility or GUI >> to make them. Don't forget to limit snapshot by time. >> Snapshot files are large and you'll not be able to sent them to the list, >> so make them available for downloading on your ftp or elsewhere. >> Regards, >> Nikolay. >> >> >> >> ------------------------------ >> *From:* [email protected] [mailto: >> [email protected]] *On Behalf Of *Александр Горбунов >> *Sent:* Monday, September 06, 2010 1:29 PM >> *To:* sipx-users >> *Subject:* [sipx-users] Can not call thought sipXecs to sipXecs trunk >> ifCaller ID assigned >> >> I have 3.10.2 (10.0.15.2) and 4.2.1(10.0.20.2) sipxecs systems. Sip >> trunk configured and works well. If I configure any caller ID for the >> extension (or for the trunking gateway) on 4.2.1 system I can not call to >> 3.10.2 system. >> >> >> Let's see 4.2.1 trace. >> >> (1) is usual INVITE from 9011 extension (4.2.1) to 2201(3.10.2). It has >> Ringing answer. >> >> (2) is INVITE from 9040 extension (4.2.1) with 3833633399 caller ID to >> 2201(3.10.2). It has Request timeout (3) answer after about 5 seconds. >> >> >> Please hint me how to resolve it. >> >> Alexander >> >> (1) >> >> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ >> >> >> Time: 2010-09-06T08:24:59.366776Z >> Frame: 294 /tmp/trace.vCh13192/_.sipXproxy.trace.xml:387946 >> Source: atc.slvz.rusalcohol.local-SipXProxy >> Dest: 10.0.15.2:5060 >> >> INVITE >> sip:[email protected];user=phone;transport=udp;sipxecs-lineid=1 >> SIP/2.0 >> Record-Route: <sip:10.0.20.2:5060 >> ;lr;sipXecs-CallDest=STS;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMDAxNDFjOGRkZjY2MDFhZDBmZDY5ZjU2LTQ5YTc3N2Y3%21eed16b1c3a14a00c8694b737dc8886ca> >> Via: SIP/2.0/UDP 10.0.20.2;branch=z9hG4bK-XX-7c382LGJh2JeqIqG3lxccfxrKA >> Via: SIP/2.0/TCP >> 10.0.20.2;branch=z9hG4bK-XX-7c35Al81`DrL4EEQ3RIamUYOlQ~l55cjTcuGEiwbJAWiIOaFQ >> Via: SIP/2.0/UDP 10.0.20.253:5060;branch=z9hG4bK0238a549 >> From: "First9011 Last9011" <sip:[email protected] >> >;tag=00141c8ddf6601ad0fd69f56-49a777f7 >> To: <sip:[email protected];user=phone> >> Call-Id: [email protected] >> Max-Forwards: 18 >> Date: Mon, 06 Sep 2010 08:24:58 GMT >> Cseq: 102 INVITE >> User-Agent: Cisco-CP7940G/8.0 >> Contact: <sip:[email protected]:5060;transport=udp;x-sipX-nonat> >> Proxy-Authorization: Digest >> username="9011",realm="atc.slvz.rusalcohol.local",uri="sip:[email protected] >> ;user=phone",response="1c2ed59909713e08e55f7f2de5d1f15f",nonce="29eaf48c191a5cff5eb29c945e53a9274c84a55b",cnonce="126330ce",qop=auth,nc=00000001,algorithm=md5 >> Expires: 7200 >> Accept: application/sdp >> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE >> Supported: replaces,join,norefersub >> Content-Length: 273 >> Content-Type: application/sdp >> Content-Disposition: session;handling=optional >> >> v=0 >> o=Cisco-SIPUA 4460 0 IN IP4 10.0.20.253 >> s=SIP Call >> t=0 0 >> m=audio 16924 RTP/AVP 0 8 18 101 >> c=IN IP4 10.0.20.253 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=sendrecv >> >> (2) >> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ >> >> Time: 2010-09-06T08:25:03.598006Z >> Frame: 323 /tmp/trace.vCh13192/_.sipXproxy.trace.xml:387998 >> Source: atc.slvz.rusalcohol.local-SipXProxy >> Dest: 10.0.15.2:5060 >> >> INVITE >> sip:[email protected];user=phone;transport=udp;sipxecs-lineid=1 >> SIP/2.0 >> Record-Route: <sip:10.0.20.2:5060 >> ;lr;sipXecs-CallDest=STS;sipXecs-rs=%2Aauth%7E.%2Afrom%7EMDAxNDFjOGRkZjY2MDFhZTYxZmY5MzA5LTU2MTg1N2Ri.700_fromalias%2Aa%7EIkY0MCBMNDAiPHNpcDozODMzNjMzMzk5QGF0Yy5zbHZ6LnJ1c2FsY29ob2wubG9jYWw%27O3RhZz0%60.700_fromalias%2Aao%7ENTY%60.700_fromalias%2Ac%7EIkY0MCBMNDAiIDxzaXA6OTA0MEBhdGMuc2x2ei5ydXNhbGNvaG9sLmxvY2FsPjt0YWc9.700_fromalias%2Aco%7ENTE%60%219d9651226c94d84689723fe1d4f5cba7> >> Via: SIP/2.0/UDP 10.0.20.2;branch=z9hG4bK-XX-7c3f_vlRj1Z8FlrMWoAi0p6dVQ >> Via: SIP/2.0/UDP >> 10.0.20.2;branch=z9hG4bK-XX-7c3cgfUsnSpyJ8YhSS3ScAG2bw~l55cjTcuGEiwbJAWiIOaFQ >> Via: SIP/2.0/UDP 10.0.20.253:5060;branch=z9hG4bK7d82b768 >> From: "F40 L40"<sip:[email protected] >> >;tag=00141c8ddf6601ae61ff9309-561857db >> To: <sip:[email protected];user=phone> >> Call-Id: [email protected] >> Max-Forwards: 18 >> Date: Mon, 06 Sep 2010 08:25:02 GMT >> Cseq: 102 INVITE >> User-Agent: Cisco-CP7940G/8.0 >> Contact: <sip:[email protected]:5060;transport=udp;x-sipX-nonat> >> Proxy-Authorization: Digest >> username="9040",realm="atc.slvz.rusalcohol.local",uri="sip:[email protected] >> ;user=phone",response="bd28fdce9e610bad801a67506dc2cdbd",nonce="df6115b2e7c6287a0156df7f5f6cab064c84a55f",cnonce="15c0dd66",qop=auth,nc=00000001,algorithm=md5 >> Expires: 7200 >> Accept: application/sdp >> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE >> Supported: replaces,join,norefersub >> Content-Length: 273 >> Content-Type: application/sdp >> Content-Disposition: session;handling=optional >> >> v=0 >> o=Cisco-SIPUA 1739 0 IN IP4 10.0.20.253 >> s=SIP Call >> t=0 0 >> m=audio 19326 RTP/AVP 0 8 18 101 >> c=IN IP4 10.0.20.253 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=sendrecv >> >> (3) >> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ >> >> Time: 2010-09-06T08:25:07.639170Z >> Frame: 325 /tmp/trace.vCh13192/_.sipXproxy.trace.xml:388001 >> Source: 10.0.15.2:5060 >> Dest: atc.slvz.rusalcohol.local-SipXProxy >> >> SIP/2.0 408 Request timeout >> From: "F40 L40"<sip:[email protected] >> >;tag=00141c8ddf6601ae61ff9309-561857db >> To: <sip:[email protected];user=phone> >> Call-Id: [email protected] >> Cseq: 102 INVITE >> Via: SIP/2.0/UDP 10.0.20.2;branch=z9hG4bK-XX-7c3f_vlRj1Z8FlrMWoAi0p6dVQ >> Via: SIP/2.0/UDP >> 10.0.20.2;branch=z9hG4bK-XX-7c3cgfUsnSpyJ8YhSS3ScAG2bw~l55cjTcuGEiwbJAWiIOaFQ >> Via: SIP/2.0/UDP 10.0.20.253:5060;branch=z9hG4bK7d82b768 >> Server: sipXecs/3.10.2 sipXecs/sipXproxy (Linux) >> Content-Length: 0 >> Date: Mon, 06 Sep 2010 08:25:07 GMT >> >> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ >> >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > > > > -- > Александр Горбунов > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Александр Горбунов
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