PJSIP: the new connected number is not updated to the SIP endpoints

Asterisk Asterisk SIP
1

Hello,

In my configuration i have some SIP endpoints and one SIP trunk.
The normal call handling is fine.
Nevertheless, I have a problem if the phone number of the peer has changed during call forwarding or explicit call transfer.

In the SIP trace, I see new phone number in a re-invite message to the SIP-trunk. However, the re-invite message is not pass through to the SIP endpoint.

In the re-invite message the new phone number is in a PAI header. The re-invite message contains also a PRIVACY header with the value “none”.
I think that could not be a problem.

In the pjsip.conf, the option “send_pai” has the value “yes” and the option is set for the SIP trunk and the SIP endpoints.

Do I need a additional configuration?

I use asterisk version 13.9.1.

Thanks.

2

Asterisk is a back to back user agent, so INVITE’s are never passed on.

Connected line update processing in mid-call requires that:

the source side has trustrpid set.

the destination has sendrpid set.

There are connected line update handler macros in the dialplan to to filter and relay; the default is not to relay. https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information

3

Thank you for this information.
Independent of the back to back user agent, I think it is necessary that the new information (new connected number) is passed to the endpoint.

I will check the usage of trustrpid and send_rpid.
Do you mean the trust_id_inbound or trust_id_outtbound?

4

The equivalent for trustrpid would be “trust_id_inbound” and the equivalent for sendrpid would be “send_rpid”. There is also “trust_id_outbound” for sending private identification details to the endpoint and “send_pai” for sending a P-Asserted-Identity header.