daniel

Upgrading to the latest 3.1 version seems to have solve the problem. Outbound caller id seems to be working now Daniel

I am getting similar bizarre behaviour on caller id, the From header is not what is expected Anyone see this?

I am struggling with some caller-id issue Sicne upgrading to 3.0, the caller id don't show correctly. I tryied to play with some of the options but can't make sense of it HEre are some logs Initial FROM HEader is Daniel/101 but subseuqent headers shows the real phone number? Anybody has some insight? Daniel [8] 2008/12/16 15:08:57: SIP Tx udp:10.1.10.50:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.1.10.50;branch=z9hG4bK251bbdf391F404A2 From: "Daniel/101" <sip:101@x.x.x.x>;tag=D3E2B506-B7E82D99 To: <sip:912403557269@x.x.x.x;user=phone>;tag=f4bfbaad3a Call-ID: 45395c85-e44ca00c-d2fd76e7@x.x.x.x CSeq: 1 CANCEL Contact: <sip:101@10.1.10.10:5060> User-Agent: pbxnsip-PBX/3.1.0.3043 Content-Length: 0 [8] 2008/12/16 15:08:57: SIP Tx udp:x.x.x.x:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK251bbdf391F404A2 From: "Daniel/101" <sip:101@x.x.x.x>;tag=D3E2B506-B7E82D99 To: <sip:912403557269@x.x.x.x;user=phone>;tag=f4bfbaad3a Call-ID: 45395c85-e44ca00c-d2fd76e7@x.x.x.x CSeq: 1 INVITE Contact: <sip:101@10.1.10.10:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.1.0.3043 Content-Length: 0 [8] 2008/12/16 15:08:57: SIP Tx udp:66.23.129.253:5060: CANCEL sip:1240yyyyyyyy@nextvortex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-fa92c65b9a0ebf6636fe6856d54fc379;rport From: "2404502166" <sip:portalsolutions@nextvortex.com>;tag=3913 To: <sip:1240yyyyyyyy@nextvortex.com;user=phone> Call-ID: d4f5e38b@pbx CSeq: 4384 CANCEL Max-Forwards: 70 Remote-Party-ID: "Daniel Cohen-Dumani" <sip:101@pbx.portalsolutions.net>;party=calling;screen=yes;privacy=full Content-Length: 0 [8] 2008/12/16 15:08:57: SIP Rx udp:x.x.x.x:5060: ACK sip:101@x.x.x.x:5060 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK251bbdf391F404A2 From: "Daniel/101" <sip:101@x.x.x.x>;tag=D3E2B506-B7E82D99 To: <sip:912403557269@x.x.x.x;user=phone>;tag=f4bfbaad3a CSeq: 1 ACK Call-ID: 45395c85-e44ca00c-d2fd76e7@10.1.10.50 Contact: <sip:101@x.x.x.x> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 Max-Forwards: 70 Content-Length: 0

Jan You are correct In other words the 24045021xx is the extension of our AA, the one we would like to present to the PSTN Do you see any other workaround? Daniel

Join I noticed that the trunk was not setup with Asserted Identity, I made the changes on the VOIP trunk and now I can see in the log INVITE sip:1301xxxxx50@nextvortex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 70.xx.ddd.99:5060;branch=z9hG4bK-e3cf7134a04f3fbb27bf132195aa69c4;rport From: <sip:+1240ddd2101@PSLCS3.portalsolutions.local;user=phone>;tag=54393 To: <sip:1301990yyyy@nextvortex.com;user=phone> Call-ID: 286de05d@pbx CSeq: 2929 INVITE Max-Forwards: 70 Contact: <sip:portalsolutions@70.88.xxx.99:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Diversion: <tel:500>;reason=unconditional;screen=no;privacy=off P-Asserted-Identity: "24045021xx" <sip:portalsolutions@nextvortex.com> Content-Type: application/sdp Content-Length: 216 Now I am wondering if our VOIP provider support asserted identity because although it shows

I change the log level 5] 2008/10/20 21:42:46: Using <sip:+1240xxx2101@PSLCS3.portalsolutions.local;user=phone>;tag=e3ae2e24f;epid=1E5D3CFF4A as redirect from [8] 2008/10/20 21:42:46: SIP Tx udp:66.23.129.xxx:5060: I could not find any message [8] 2008/10/18 02:54:46: Trunk: Changing the user to Should I be seeing this?

Jan 1) We are using a VOIP provider 2) I want to show the AA number to external parties being called, for internal parties, showing the extension number would be ideal instead of the full fake DID

Still no luck I made the changes you suggested but still no change It is my understanding that if the Mediation server send the number with the Remote Party ID and user=phone, the number is nver overwritten Thanks for your help Daniel Here is the log from PBXNSIP INVITE sip:12403557269@nextvortex.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 70.88.239.99:5060;branch=z9hG4bK-b50f6d9807bf45e26a5510846782f3a7;rport From: "240xxx2166 - NT" <sip:portalsolutions@nextvortex.com>;tag=53113 To: <sip:1240xxx7269@nextvortex.com;user=phone> Call-ID: 1ea319da@pbx CSeq: 4808 INVITE Max-Forwards: 70 Contact: <sip:portalsolutions@70.88.xxx.yy:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.0.1.3023 Remote-Party-ID: <sip:+1240xxx2101@PSLCS3.portalsolutions.local;user=phone>;party=calling;screen=yes Content-Type: application/sdp Content-Length: 216 v=0 o=- 34703 34703 IN IP4 70.88.239.99 s=- c=IN IP4 70.88.239.99 t=0 0 m=audio 50500 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2008/10/16 15:01:16: Resolve 62954: tcp 10.1.10.22 2583 [9] 2008/10/16 15:01:16: SIP Tx tcp:10.1.10.22:2583: SIP/2.0 183 Ringing Via: SIP/2.0/TCP 10.1.10.22:2583;branch=z9hG4bK6522e822 From: <sip:+1240xxx2101@PSLCS3.portalsolutions.local;user=phone>;tag=79e91e3154;epid=1E5D3CFF4A To: <sip:91240xxx7269@10.1.10.10;user=phone>;tag=15dd9e5311 Call-ID: ba8dd0c8-1173-4df1-941e-04c9e133343f CSeq: 16 INVITE Contact: <sip:anonymous@10.1.10.10:5060;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/3.0.1.3023 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 224 v=0 o=- 42105 42105 IN IP4 10.1.10.10 s=- c=IN IP4 10.1.10.10 t=0 0 m=audio 59994 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

I dont' see the ANI field, do you need a specific version? We are running 2.1.14.2498

where is the ANI field located in OCS or PBXNSIP. We don't use a gateway and connect the OCS Mediation server directly to PBXNSIP?

We had a similar issue. For use the issue related to the way the Mediation Server was setup. The A/V authentication service FQDN was not setup correctly and matching the certificate on the Edge Server

I succesfully setup integration between OCS and PBXNSIP. After much pain, Inbound and outbound calls are working. Our PBXNSIP is setup with an auto-attendant as we don't have DID for each extension xxx-xxx-xxxx. FOR OCS to work, we setup and registered each extension using a "fake" DID xxx-xxx-xyyy where yyy is extension is the extension of the user. We created static registration in PBXNSIp The problem we have is that outbound call shows with the "fake" DID. How can we override the DID to show the number of the auto-attendant for every users. Do we need to manipulate dialplan in OCS or PBXNSIP or what else is needed to do this? Thanks - Daniel

Thanks for the quick response, this seems to work as expected. Now I do I fix the Caller ID, calls are not apparently coming from the "fake" number and not from the main number. Do I need to change something in PBXNSIP or manipulate DialPlan in OCS

Hi everyone I have been trying unusucesffully to register Office Communicator client thru PBXNSIP suing the Mediation Server. I succesfully setup the Mediation server and enabled outgoing phone call from Office Communicator which works fine except for some dialplan issues However I can't seemt to get the Communicator client to register with PBXNSIP. I followed the setup in the wiki page and created static registration but not sure how to setup the users because we don't have DID 1. How should the line URI be setup in OCS User properties, we do not use DID and have an auto-attendant with extension, should the lineURI: be tel:+main phone number; ext=101 2. How to register the static registration should it be sip:+main number;ext=101@pbxnsipip I would appreciate some insight on how to set this up Daniel

Problem resolved, Our ISTP made a chance to override all outgoing to havea our main number, we reverted the changes and handle this in the PBX