Andrep

Hi all, is it possible to disable the PRACK support? I have a gateway for OCS connectivity (OfficeMaster Gate) and i need to disable the PRACK support on the GW (their support asked me to do that for testing) because i have no audio between the GW and PBXNSIP with their new release. If I do so, the phone rings (a normal SIP-Phone connected to pbxnsip) only one time and then the call disappears. I just want to test what happens if i could disable the PRACK support on the pbxnsip... Thanks

Hi Jan, It works now. I had to replace the ip-address of the mediation server with the fqdn. Regarding the OfficeMaster Gateway I have still problems (no audio --> I guess codec mismatch). I'm in contact with their support and I will post the config as soon as the problems are solved.

Hi all, I configured PBXNSIP for OCS. I can dial in from a PSTN device to an extension and/or to the OCS, but i cannot dial out from OCS to a PSTN device (It works from an extension which is directly connected to PBXNSIP) I have two trunks configured (1 to OCS and 1 to the GW for PSTN connectivity) Here is the log (Communicator is +41434435683 and i want to dial +41763801234 which is a mobile phone): [5] 2008/06/03 15:23:13: SIP port accept from 172.25.20.66:4749 [9] 2008/06/03 15:23:13: SIP Rx tcp:172.25.20.66:4749: INVITE sip:+41763801234@172.25.20.110;user=phone SIP/2.0 FROM: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31 TO: <sip:+41763801234@172.25.20.110;user=phone> CSEQ: 13 INVITE CALL-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8 CONTACT: <sip:MEDIATION.collabcom.ch:5060;transport=Tcp;maddr=172.25.20.66;ms-opaque=3d8ccc5b0ddbda89> CONTENT-LENGTH: 302 SUPPORTED: 100rel USER-AGENT: RTCC/3.0.0.0 MediationServer CONTENT-TYPE: application/sdp; charset=utf-8 ALLOW: UPDATE ALLOW: Ack, Cancel, Bye,Invite v=0 o=- 0 0 IN IP4 172.25.20.66 s=session c=IN IP4 172.25.20.66 b=CT:1000 t=0 0 m=audio 55808 RTP/AVP 97 101 0 8 c=IN IP4 172.25.20.66 a=rtcp:55809 a=label:Audio a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 [9] 2008/06/03 15:23:13: Resolve 5679: tcp 172.25.20.66 4749 [9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749: SIP/2.0 100 Trying Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8 From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31 To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094 CSeq: 13 INVITE Content-Length: 0 [9] 2008/06/03 15:23:13: Resolve 5680: tcp 172.25.20.66 4749 [9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749: SIP/2.0 404 Not Found Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8 From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31 To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094 CSeq: 13 INVITE Contact: <sip:+41763801234@172.25.20.110:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.10.2474 Content-Length: 0 [9] 2008/06/03 15:23:13: Resolve 5681: tcp 172.25.20.66 4749 [9] 2008/06/03 15:23:13: SIP Tx tcp:172.25.20.66:4749: SIP/2.0 404 Not Found Via: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8 From: <sip:+41434435683@mediation.collabcom.ch;user=phone>;epid=FF7797D3EA;tag=b42ca2ac31 To: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e Call-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094 CSeq: 13 INVITE Contact: <sip:+41763801234@172.25.20.110:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.10.2474 Content-Length: 0 [9] 2008/06/03 15:23:13: SIP Rx tcp:172.25.20.66:4749: ACK sip:+41763801234@172.25.20.110;user=phone SIP/2.0 FROM: <sip:+41434435683@mediation.collabcom.ch;user=phone>;tag=b42ca2ac31;epid=FF7797D3EA TO: <sip:+41763801234@172.25.20.110;user=phone>;tag=3b5230389e CSEQ: 13 ACK CALL-ID: 5b978b49-1fdf-4745-94ba-ce4206d8d094 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 172.25.20.66:4749;branch=z9hG4bK7315eaa8 CONTENT-LENGTH: 0 Thanks for your help. Could it be that my Dial Plan is not correct? I also need to replace the + to 00!

Thanks a lot for the answers. Sounds great --> I will plan to deply a test installation of the pbxnsip. Just another question: Is it true that I will need to configure 2 trunks (1 to gateway and 1 to OCS)? In case I handle outgoing calls with OCS: do I need to create a trunk to the gateway for incoming calls?

Hi all, does somebody now if the Gateway "OfficeMaster Gate/Box" from Ferrari Electronics is supported? Thanks

Hi all, We have an OCS and Exchange 2007 UC Setup. For PSTN Connectivity we use a Ferrari Electronic OfficeMaster Box. We are missing the hunt group feature on the Microsoft side and so we are in need of a PBX, which we not have at this moment... Does it work if i setup a hunt group in pbxnsip and add extensions of some OC-Clients? The goal is that our main phone number rings on multiple Office Communicator Clients Also i wanted to know if the ferrari officemaster gateway is supported with pbxnsip. If you are confident I will ask for an evaluation license and setup pbxnsip. Thanks a lot for your help