On Thu, Nov 19, 2009 at 9:41 AM, M. Ranganathan <mra...@gmail.com> wrote: > 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. >
It should also be stated that some things are done a bit differently between 4.0.x and 4.1 with respect to how MOH is handled so as to enable MOH to be on simultaneously on the phone as well as the bridge. It is possible that a scenario involving MOH which works in 4.0.4 may be broken in 4.1. If you decide to send a snapshot, please ensure that logging levels are correctly set. You can mail it to me directly. 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 > -- 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/
