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Caller-id issues


daniel
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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

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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?

 

The PBX is a B2BUA. That means that different Call-ID may have different From-header. Within a call, the From is not allowed to changed (unless the connected UA indicates it supports from-change, see http://www.ietf.org/rfc/rfc4916.txt).

 

What the PBX puts into the other leg of a call is a complex topic. "It depends". Trunks have their own rules, and the address book plays a major role for inbound calls.

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The PBX is a B2BUA. That means that different Call-ID may have different From-header. Within a call, the From is not allowed to changed (unless the connected UA indicates it supports from-change, see http://www.ietf.org/rfc/rfc4916.txt).

 

What the PBX puts into the other leg of a call is a complex topic. "It depends". Trunks have their own rules, and the address book plays a major role for inbound calls.

 

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

 

Daniel

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