A bit offtopic: regarding siptrunk between two sipx systems. I think, that sometime it is convinient to use siptrunk. The thing is that, when two sipx systems are "routable" but have different "intranet subnets" one has to relay media via sipxbridge, albeit via internal ip address. Rgds, Nikolay.
_____ From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: Monday, September 06, 2010 4:28 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Can not call thought sipXecs to sipXecs trunk ifCaller ID assigned Can you also explain how you configured your dialplan to call between the two systems? A siptrunk would be unneeded in any version, so it would be helpful to know what the dialplan looks like. On Mon, Sep 6, 2010 at 6:13 AM, Nikolay Kondratyev <[email protected]> wrote: 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/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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