Andrep Posted June 3, 2008 Report Share Posted June 3, 2008 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! Quote Link to comment Share on other sites More sharing options...
Jan Boguslawski Posted June 4, 2008 Report Share Posted June 4, 2008 Hi Andrep, I assume that you have read my wiki page for Basic Setup for pbxnsip / Office Communications Server 2007 Interoperability OCS. If not, please read it very careful. I can see that the NOT FOUND is coming prompt from the pbxnsip. So I guess another trunk (outbound) is not involved, you need to check the trunk, configured to the OCS Meditiation Server again. Please compare your configuration with the one in the wiki. Like described, make sure that the Account under "Assume that calls come from user:" is existing. It must only exist, a registration is not necessary. btw.: I was trying to call your +41434435683 and it was succesfully ringing , but did no redirect to an Exchange 2007 Unified Messaging Server Simply Decline after some rings. Are you using the Ferrari Office Master now with pbxnsip for PSTN? You can check your call log, find my Berlin number and call me back if you like or simply look at my signature her Best regards, Jan Quote Link to comment Share on other sites More sharing options...
Andrep Posted July 16, 2008 Author Report Share Posted July 16, 2008 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 Andrep, I assume that you have read my wiki page for Basic Setup for pbxnsip / Office Communications Server 2007 Interoperability OCS. If not, please read it very careful. I can see that the NOT FOUND is coming prompt from the pbxnsip. So I guess another trunk (outbound) is not involved, you need to check the trunk, configured to the OCS Meditiation Server again. Please compare your configuration with the one in the wiki. Like described, make sure that the Account under "Assume that calls come from user:" is existing. It must only exist, a registration is not necessary. btw.: I was trying to call your +41434435683 and it was succesfully ringing , but did no redirect to an Exchange 2007 Unified Messaging Server Simply Decline after some rings. Are you using the Ferrari Office Master now with pbxnsip for PSTN? You can check your call log, find my Berlin number and call me back if you like or simply look at my signature her Best regards, Jan Quote Link to comment Share on other sites More sharing options...
paco81 Posted August 29, 2008 Report Share Posted August 29, 2008 Hi all, I have a problem...when I make a call from MOC, if the called doesn't answer, after 180+100rel I can not send the message "480 Temporally not Available": the connection to Mediation server is refused after 1 minute of inactivity. Someone can help me? Thanks Quote Link to comment Share on other sites More sharing options...
Guest Pats Posted August 29, 2008 Report Share Posted August 29, 2008 Hi all,I have a problem...when I make a call from MOC, if the called doesn't answer, after 180+100rel I can not send the message "480 Temporally not Available": the connection to Mediation server is refused after 1 minute of inactivity. Someone can help me? Thanks One thing you should check in your trunk configuration "Remote Party/Privacy Indication", make sure it is setting to "No Indication". Quote Link to comment Share on other sites More sharing options...
paco81 Posted September 1, 2008 Report Share Posted September 1, 2008 One thing you should check in your trunk configuration "Remote Party/Privacy Indication", make sure it is setting to "No Indication". Hi Pats, thanks for answer...how can I check it from the SIP traffic? Thank you. Quote Link to comment Share on other sites More sharing options...
Pradeep Posted September 2, 2008 Report Share Posted September 2, 2008 Hi Pats,thanks for answer...how can I check it from the SIP traffic? Thank you. He might have suggested to check it under the trunk configuration. Quote Link to comment Share on other sites More sharing options...
safiman Posted April 28, 2009 Report Share Posted April 28, 2009 Hi, i have a same problem, i can't call out from communicator to iP-Phone, i can call from iP-Phone to communicator and communicator to communicator, i don't have a media gateway just mediation server with direct sip trunk to Mitel 3300. do you have some suggestions ? thank's Quote Link to comment Share on other sites More sharing options...
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