daniel Posted October 2, 2008 Report Share Posted October 2, 2008 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 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted October 3, 2008 Report Share Posted October 3, 2008 Do we need to manipulate dialplan in OCS or PBXNSIP or what else is needed to do this? The only think that comes to my mind is to use the ANI field. ANI is for outbound, tel:alias (or just a regular alias) for inbound. But I am not the big OCS expert, maybe it is easier to manipulate the caller-ID in OCS. And don't forget the gateway. Most gateways have great flexibility rewriting the caller-ID. Quote Link to comment Share on other sites More sharing options...
daniel Posted October 3, 2008 Author Report Share Posted October 3, 2008 The only think that comes to my mind is to use the ANI field. ANI is for outbound, tel:alias (or just a regular alias) for inbound. But I am not the big OCS expert, maybe it is easier to manipulate the caller-ID in OCS. And don't forget the gateway. Most gateways have great flexibility rewriting the caller-ID. 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? Quote Link to comment Share on other sites More sharing options...
Pradeep Posted October 3, 2008 Report Share Posted October 3, 2008 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? ANI field is available both on the trunk and the extension level. (Note: on the trunk page look for "Trunk DID", if you are using the official 3.1 version) Quote Link to comment Share on other sites More sharing options...
daniel Posted October 3, 2008 Author Report Share Posted October 3, 2008 ANI field is available both on the trunk and the extension level. (Note: on the trunk page look for "Trunk DID", if you are using the official 3.1 version) I dont' see the ANI field, do you need a specific version? We are running 2.1.14.2498 Quote Link to comment Share on other sites More sharing options...
Jan Boguslawski Posted October 16, 2008 Report Share Posted October 16, 2008 Hi Daniel, if you simply want to show the number of the auto-attendant then you should set the Assume that call comes from user: on the Mediations Server Trunk in pbxnsip to the account number of your AA. Next change: Remote Party/Privacy Indication to RFC3325 (P-Asserted Identity) note: Normally, if you have a direct call number for every user you would use RFC3325 (P-Preferred Identity) to show the direct number to PSTN. Did you checked out the pbxnsip Wiki?: http://wiki.pbxnsip.com/index.php/Office_C...ications_Server Best regards Jan Quote Link to comment Share on other sites More sharing options...
daniel Posted October 16, 2008 Author Report Share Posted October 16, 2008 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 Quote Link to comment Share on other sites More sharing options...
Jan Boguslawski Posted October 18, 2008 Report Share Posted October 18, 2008 Hi Daniel, ok just for clarification: 1) you want to show the AA number to whom? To your internal user or to the PSTN? 2) are you using a VoIP-PSTN gateway or a VoIP-Provider? 3) are you using the ANI field, that pbxnsip recommended simultaneously to my recommandation (Remote Party/Privacy Indication to RFC3325 (P-Asserted Identity))? Please change log-level to 8 and activate log other messages! You should see something like this in traces of your OC outbound calls if your MediationsServer Trunk is correctly configured: [8] 2008/10/18 02:54:46: Trunk: Changing the user to 59 This is the Account charged for this trunk to trunk call - in your case this should be the AA number [5] 2008/10/18 02:54:46: Using <sip:+493039978418@OCS-Mediation.domain.de;user=phone>;tag=8f4ce1dc71;epid=618BBC3B7E as redirect from Best regards and a nice weekend, Jan Quote Link to comment Share on other sites More sharing options...
daniel Posted October 20, 2008 Author Report Share Posted October 20, 2008 Hi Daniel, ok just for clarification: 1) you want to show the AA number to whom? To your internal user or to the PSTN? 2) are you using a VoIP-PSTN gateway or a VoIP-Provider? 3) are you using the ANI field, that pbxnsip recommended simultaneously to my recommandation (Remote Party/Privacy Indication to RFC3325 (P-Asserted Identity))? Please change log-level to 8 and activate log other messages! You should see something like this in traces of your OC outbound calls if your MediationsServer Trunk is correctly configured: [8] 2008/10/18 02:54:46: Trunk: Changing the user to 59 This is the Account charged for this trunk to trunk call - in your case this should be the AA number [5] 2008/10/18 02:54:46: Using <sip:+493039978418@OCS-Mediation.domain.de;user=phone>;tag=8f4ce1dc71;epid=618BBC3B7E as redirect from Best regards and a nice weekend, Jan 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 Quote Link to comment Share on other sites More sharing options...
Jan Boguslawski Posted October 20, 2008 Report Share Posted October 20, 2008 Hi Daniel, please can you ask your VoIP-Provider if he supports: Remote Party/Privacy Indication - RFC3325 (P-Asserted Identity) ? If he doesn't, we should look for another workaround. When you change log-level to 8 and activate log other messages, you should see the P-Asserted Identity = Your AA extension if configured correctly. btw.: We are using a VoIP Gateway Audiocodes Mediant 1000. It supports the RFC3325 (P-Asserted Identity) and passing it correctly to PSTN. best regards, Jan Quote Link to comment Share on other sites More sharing options...
daniel Posted October 21, 2008 Author Report Share Posted October 21, 2008 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 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? Quote Link to comment Share on other sites More sharing options...
daniel Posted October 21, 2008 Author Report Share Posted October 21, 2008 Hi Daniel, please can you ask your VoIP-Provider if he supports: Remote Party/Privacy Indication - RFC3325 (P-Asserted Identity) ? If he doesn't, we should look for another workaround. When you change log-level to 8 and activate log other messages, you should see the P-Asserted Identity = Your AA extension if configured correctly. btw.: We are using a VoIP Gateway Audiocodes Mediant 1000. It supports the RFC3325 (P-Asserted Identity) and passing it correctly to PSTN. best regards, Jan 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 Quote Link to comment Share on other sites More sharing options...
Jan Boguslawski Posted October 21, 2008 Report Share Posted October 21, 2008 Ok. Is the P-Asserted-Identitiy now correct? P-Asserted-Identity: "24045021xx" <sip:portalsolutions@nextvortex.com> In other words the 24045021xx is the extension of your AA, the one you like to present to the PSTN? If your answer is yes, fine Regarding the support of RFC3325 (P-Asserted Identity) by your VoIP-Provider NexVortex, please take a look at this thread Caller ID From Address It seems that NexVortex do not support RFC3325 . Maybe they should stop selling "Business Grade VoIP Trunks" until they support RFC3325 and become "Enterprise Ready" If you think about buying a PSTN-VoIP-Gateway, I can recommend the AudioCodes Mediant 1000. Best Regards, Jan Quote Link to comment Share on other sites More sharing options...
daniel Posted October 21, 2008 Author Report Share Posted October 21, 2008 Ok. Is the P-Asserted-Identitiy now correct? P-Asserted-Identity: "24045021xx" <sip:portalsolutions@nextvortex.com> In other words the 24045021xx is the extension of your AA, the one you like to present to the PSTN? If your answer is yes, fine Regarding the support of RFC3325 (P-Asserted Identity) by your VoIP-Provider NexVortex, please take a look at this thread Caller ID From Address It seems that NexVortex do not support RFC3325 . Maybe they should stop selling "Business Grade VoIP Trunks" until they support RFC3325 and become "Enterprise Ready" If you think about buying a PSTN-VoIP-Gateway, I can recommend the AudioCodes Mediant 1000. Best Regards, Jan 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 Quote Link to comment Share on other sites More sharing options...
Jan Boguslawski Posted October 21, 2008 Report Share Posted October 21, 2008 Hi Daniel, I am sorry, but I dont see a workaround at the moment. I am at my wit's end. . Has anybody else an idea? Best regards, Jan Quote Link to comment Share on other sites More sharing options...
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