Update callerID on trasfer

Asterisk Asterisk General
1

I have experienced this problem.
when I make a trip on the caller does not update the field called id, so if the 200 called the 201 which transfers the call to the 202, the 200 I keep seeing the number 201

Any idea for this problem

thanks

2

Connected Line Updates.

See the Asterisk Wiki for more information.

Short story:
sendrpid
trustrpid

parameters of sip.conf.

And, your SIP phone’s going to have to support displaying the updates as they’re sent.

Cheers

3

Thank you for your answer.

In sip.conf file I am already using the parameters you mentioned, but I continue to have problems updating the callerdID.
You happen to have any other suggestions.

INVITE sip:8134@10.28.131.109 SIP/2.0.
Via: SIP/2.0/UDP 10.28.131.83:5060;branch=z9hG4bK4a8d5dd6;rport.
Max-Forwards: 70.
From: sip:8133@10.28.131.83;tag=as4dd8eee9.
To: 8134sip:8134@10.28.131.83;tag=3635553478.
Contact: sip:8133@10.28.131.83:5060.
Call-ID: 476400207@10.28.131.109.
CSeq: 102 INVITE.
User-Agent: VitroVoip.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
Supported: replaces, timer.
Remote-Party-ID: “-> (via 8133@10.20.64.67:49782)” sip:8135@10.28.131.83;party=called;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 247.

Above you can find the SIP invite the track, in the field of Remote-Party-ID you see the correct number of the called party, m ail field of the caller continues to see the number of the person making the Transfer.
may depend on the point in qyesto Sip Client?

Greetings and thanks
Silvio