On Thu, Nov 19, 2009 at 1:47 AM, tetto Kalle <tettoka...@hotmail.com> wrote: > Hi, > > I have a problem using my Siemens Openstage phones on sipxecs, latest 4.1 > version. > > Calls coming in to the sipxbridge and destined for my internal phones work > very well. Pickup and Hold/Resume works on SNOM and Polycom. As we are in > Germany and our guys are used to Siemens, they want to use the new Openstage > phones. I have upgraded to the latest SIP firmware. > > When a call comes to my Siemens phone, I cannot put the caller on hold. I > can give the SIP trace for my the Siemens Phone, which does not work and a > Snom Phone where it works. > It works also for internal hold/resume with Siemens phones, but I'm not able > to hear MOH on the phone, where I try to put the caller on hold. > > Are there any know Bugs with Siemens? The reinvite from the Siemens seems > not to be totally wrong, as I it works internally to put the caller on
We do not normally test with Siemens phone so I cannot be certain of the reason. It is possible that Siemens is sending Re-INVITE when another INVITE is pending and hence the problem. Please send a sipx-snapshot (see http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration#Problem_reporting) for further information. Regards, Ranga. > > SNOM > Time: 2009-11-18T14:27:00.236417Z > Frame: 10 /tmp/trace.FZd14188/_.sipXproxy.trace.xml:233533 > Source: 10.100.120.254:1024 > Dest: obelix.voip.lab-SipXProxy > > INVITE sip:1...@10.100.120.11:5090;x-sipX-nonat SIP/2.0 > Via: SIP/2.0/UDP 10.100.120.254:1024;branch=z9hG4bK-efmcu2t1bzzj;rport > Route: > <sip:10.100.120.10:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2Aauth%7E.%2Afrom%7EQUFDRkYyQy1GMUM%60%21d76fd415ae8397857ee85c13cd4e7ae5> > From: <sip:1...@voip.lab>;tag=gkmrf7lce0 > To: "SLT Gesamt Brake" <sip:93...@10.100.120.1>;tag=AACFF2C-F1C > Call-ID: 41c1fa30-d38511de-89e0aef2-e00c6...@10.100.120.1.0 > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: <sip:1...@10.100.120.254:1024;line=vhk5ttq1>;reg-id=1 > P-Key-Flags: resolution="31x13", keys="4" > User-Agent: snom370/7.3.27 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, > MESSAGE, INFO > Allow-Events: talk, hold, refer, call-info > Supported: timer, replaces, from-change > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 365 > > v=0 > o=root 753282881 753282883 IN IP4 10.100.120.254 > s=call > c=IN IP4 0.0.0.0 > t=0 0 > m=audio 61042 RTP/AVP 9 0 8 99 3 18 4 101 > a=rtpmap:9 g722/8000 > a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:99 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > > SIEMENS > Time: 2009-11-18T14:24:09.743482Z > Frame: 18 /tmp/trace.DvU14082/_.sipXproxy.trace.xml:233299 > Source: 10.100.120.214:5060 > Dest: obelix.voip.lab-SipXProxy > > INVITE sip:1...@10.100.120.11:5090;x-sipX-nonat SIP/2.0 > Via: SIP/2.0/UDP 10.100.120.214;branch=z9hG4bK9b4d40000c0ddf928 > Route: > <sip:10.100.120.10:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=*auth~.*from~QUFBNEVEMC0xOTk3!8d1932ebe4a1babd0e8d8a5824cd067f> > Max-Forwards: 70 > From: <sip:1...@voip.lab>;tag=763820167 > To: "SLT Gesamt Brake" <sip:93...@10.100.120.1>;tag=AAA4ED0-1997 > Call-ID: d8b8f16e-d38411de-89d2aef2-e00c6...@10.100.120.1.0 > CSeq: 1124947490 INVITE > Contact: Openstage 40 <sip:1...@10.100.120.214:5060;transport=udp> > Min-SE: 90 > Supported: replaces, 100rel, timer > User-Agent: OpenStage_40_V2 R0.16.1 SIP 090709 > X-Siemens-Call-Type: ST-insecure > Content-Type: application/sdp > Content-Length: 328 > > v=0 > o=MxSIP 0 240481723 IN IP4 10.100.120.214 > s=SIP Call > c=IN IP4 10.100.120.214 > t=0 0 > m=audio 5010 RTP/AVP 9 8 0 18 101 > a=rtpmap:9 G722/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > a=silenceSupp:off - - - - > a=fmtp:18 annexb=no > a=fmtp:101 0-15 > a=inactive > a=sendonly > > From the sipxbridge I get this: > Time: 2009-11-18T14:24:09.753377Z > Frame: 23 /tmp/trace.DvU14082/_.sipXproxy.trace.xml:233304 > Source: obelix.voip.lab-SipXProxy > Dest: 10.100.120.214:5060 > > SIP/2.0 491 Request Pending > Via: SIP/2.0/UDP 10.100.120.214;branch=z9hG4bK9b4d40000c0ddf928 > From: <sip:1...@voip.lab>;tag=763820167 > To: "SLT Gesamt Brake" <sip:93...@10.100.120.1>;tag=AAA4ED0-1997 > Call-Id: d8b8f16e-d38411de-89d2aef2-e00c6...@10.100.120.1.0 > Cseq: 1124947490 INVITE > Server: sipXecs/4.1.0 sipXecs/sipxbridge (Linux) > Retry-After: 1 > Content-Length: 0 > Date: Wed, 18 Nov 2009 14:24:09 GMT > > > Any help is very much appreciated, if you need more logs please let me know. > > With Best Regards > ________________________________ > Große Daten übertragen? Ganz einfach - mit dem Messenger! > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ > -- M. Ranganathan _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
