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Overriding CallerID


daniel
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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

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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.

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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?

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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)

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  • 2 weeks later...

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

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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

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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

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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

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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

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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?

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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

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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

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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

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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

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