On Tue, Mar 29, 2011 at 3:38 PM, Douglas Hubler <dhub...@ezuce.com> wrote: > On Tue, Mar 29, 2011 at 3:16 PM, Matthew Kitchin (public/usenet) > <mkitchin.pub...@gmail.com> wrote: >> Sure. Merged logs are attached. This is a call from 6159253938 (sipx 4.4 >> test) to 6154670142 (sipx 4.2.1 production). The 2 systems are on seperate >> Verizon sip trunks.
I'm no expert so here are the final messages, maybe someone has an idea. Also: It occurred to me we really need one more trace: call forwarding working to verizon from your 4.2.1 system. DOESN'T WORK (far-end ultimately send CANCEL after 20 seconds. Generated from call forwarding) ==================================================== INVITE sip:6154670142@172.30.216.62;sipx-noroute=Voicemail;user=phone SIP/2.0 Call-ID: BW102147333250311467067431@63.77.76.250-1 CSeq: 1 INVITE From: "WIRELESS CALLER" <sip:6155008073@63.77.76.250>;tag=1035674599848618864 To: <sip:6154670142@172.30.216.62;user=phone> Via: SIP/2.0/UDP 10.81.3.5:5080;branch=z9hG4bKaeb1e386274092e542c00ac082f20b35333432 Max-Forwards: 70 User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux) P-Asserted-Identity: "WIRELESS CALLER" <sip:6155008073@63.77.76.250;user=phone> Contact: <sip:6155008073@10.81.3.5:5080;transport=udp> Route: <sip:172.30.216.62:5070;transport=udp;lr> Session-Expires: 1800;refresher=uac References: BW102147333250311467067431@63.77.76.250-0;rel=chain;sipxecs-tag=request-invite-z9hg4bk-xx-a7c7hxqzqrk5`emcdlkjkxchrq Allow: INVITE,BYE,ACK,CANCEL,OPTIONS Supported: timer Content-Type: application/sdp Content-Length: 185 v=0 o=sipxbridge 4725027341631093188 1 IN IP4 10.81.3.5 s=- c=IN IP4 10.81.3.5 t=0 0 m=audio 30502 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 WORKS (Enters ringing state. Straight INVITE thru sipXbridge) ==================================================== INVITE sip:6154670142@4.sipxt.voipt;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.81.3.254;branch=z9hG4bK661f71454BF4BD84 From: "ID: 272" <sip:~~id~sipXprovision@4.sipxt.voipt>;tag=D012051C-AE0C1E7 To: <sip:6154670142@4.sipxt.voipt;user=phone> CSeq: 2 INVITE Call-ID: 5c69e470-9d6b90cb-328ec102@10.81.3.254 Contact: <sip:~~id~sipXprovision@10.81.3.254> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/3.2.4.0244 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="~~id~sipXprovision/0004f22cea3b", realm="4.sipxt.voipt", nonce="ab6954891c162b30f6caa5632465a72f4d92389c", qop=auth, cnonce="vRSCyQc4L5i5Pgy", nc=00000001, uri="sip:6154670142@4.sipxt.voipt;user=phone", response="fe2f380662f28c73bcf87149c81d7be4", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 292 v=0 o=- 1301426889 1301426889 IN IP4 10.81.3.254 s=Polycom IP Phone c=IN IP4 10.81.3.254 t=0 0 a=sendrecv m=audio 2258 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 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 _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
