Received Caller List - Showing internal caller on attended transfer

[Chappo]

Guys,

 

As the title states I am having some issues with how the internal phones register the incoming calls.

 

 

The phones on an attended transfer or even picking up an external trunk directly will register either the star code or the person who performed the attended transfer.

 

Is there a way to override this on the pbx side of things?

 

callrecord_received_local0!: 45@pbx.domain.com.au

callrecord_received_local1!: 45@pbx.domain.com.au

callrecord_received_local2!: 45@pbx.domain.com.au

callrecord_received_local3!: 45@pbx.domain.com.au

callrecord_received_local4!: 45@pbx.domain.com.au

callrecord_received_remote0!: sip:40@pbx.domain.com.au

callrecord_received_remote1!: sip:47@pbx.domain.com.au

callrecord_received_remote2!: sip:40@pbx.domain.com.au

callrecord_received_remote3!: sip:61@pbx.domain.com.au

callrecord_received_remote4!: sip:*6011273@pbx.domain.com.au

 

Trunk Settings:

Request-URI Let the System Decide

From: Based on trunk account info

To: Same as request-URI

P-Asserted-Identity: Don't use Header

P-Preferred-Identity: Don't use Header

Remote-Party-ID: Don't use Header

Privacy Indication: Other: id

 

The problem I have is modifying any of these values results in a 502 Bad Gateway error as shown below:

 

 

[5] 2013/04/19 12:04:41: SIP Rx udp:125.213.160.81:5060:

SIP/2.0 502 Bad Gateway

Via: SIP/2.0/UDP 172.28.1.5:5060;branch=z9hG4bK-a6f55ac37452f8b6bbc0f28e35179a25;rport

To: <sip:947180000@sip00.mynetfone.com.au>

From: "JR"<sip:OurPhoneNo@pbx.domain.com.au;user=phone>;tag=13990

Call-ID: 08b383d5@pbx

CSeq: 5001 INVITE

User-Agent: ENSR3.0.66.41-IS1

Content-Length: 0

 

[5] 2013/04/19 12:04:41: SIP Tx udp:125.213.160.81:5060:

ACK sip:947180000@sip00.mynetfone.com.au;user=phone SIP/2.0

Via: SIP/2.0/UDP 172.28.1.5:5060;branch=z9hG4bK-a6f55ac37452f8b6bbc0f28e35179a25;rport

From: "JR" <sip:OurPhoneNo@pbx.domain.com.au;user=phone>;tag=13990

To: <sip:947180000@sip00.mynetfone.com.au>

Call-ID: 08b383d5@pbx

CSeq: 5001 ACK

Max-Forwards: 70

Contact: <sip:09406157@172.28.1.5:5060;transport=udp>

Privacy: id

Content-Length: 0

 

[5] 2013/04/19 12:04:41: INVITE Response 502 Bad Gateway: Terminate 08b383d5@pbx

[5] 2013/04/19 12:04:41: SIP Tx tls:172.28.1.84:3629:

SIP/2.0 502 Bad Gateway

Via: SIP/2.0/TLS 172.28.1.84:3629;branch=z9hG4bK-66fvzg68v8or;rport=3629

From: "JR" <sip:48@pbx.domain.com.au>;tag=9q3f1xp4xm

To: <sip:947180000@pbx.domain.com.au;user=phone>;tag=a7fad5b5fd

Call-ID: 49c270511180-6ninip39wyuz

CSeq: 1 INVITE

Contact: <sip:48@172.28.1.5:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snomONE/5.0.8

[Chappo]

Anyone?

[Vodia PBX]

Well the Bad Gateway does come from the service provider (ENSR3.0.66.41-IS1). Probably you have to tweak the SIP trunk header settings to make it work. If the provider has a description where he wants the authentication info and where the caller-ID info, it should speed up the process.

[Chappo]

This is what has been provided by the VSP.

 

INVITE sip:1300887899@sip10.mynetfone.com.au:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.150:5060;branch=z9hG4bK-d8754z-591caa70b8520754-1---d8754z-;rport=61973;received=xxx.xxx.xxx.xxx

Max-Forwards: 70

Contact: <sip:09123456@xxx.xxx.xxx.xxx:5060>

To: <sip:1300887899@sip10.mynetfone.com.au:5060>

From: "0280088000"<sip: 09123456@sip10.mynetfone.com.au:5060>;tag=e53ba10d

Call-ID: MTY4YTMxOTE2MDk4ZTAzMThkYmFlY2QwY2MxZTE4MDE.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE

Content-Type: application/sdp

Supported: replaces

User-Agent:

Content-Length: 332

Remote-Party-ID: "0280088000"<sip: 0280088000@sip10.mynetfone.com.au:5060>;party=calling

 

^ Please ensure that your remote party ID or P asserted identity string is exactly the same as the "from" string – The above example would fail because it is not the same.

 

Typically the P Asserted Identity would be your internal extension number OR your DID.

[Vodia support]

You can choose other under the P-asserted identity and put something like

 

<sip:{0280088000}@{sip10.mynetfone.com.au};user=phone>

 

or

 

<sip:{0280088000}@{trunk-outbound-proxy};user=phone>

 

not sure if 0280088000 is the DID provided to you