[pjsip] Caller ID?

Rajkumar S rajkumars at gmail.com
Thu Nov 1 06:28:35 EDT 2007


Hi,

I am using pjsua for testing asterisk. We are using an AudioCodes
device to connect asterisk to PSTN. When a call lands in Audiocodes,
it gets forwarded to asterisk. The call gets landed in * with
following headers.

INVITE sip:2753009 at 192.168.9.210;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.9.230;branch=z9hG4bKac1714857730
Max-Forwards: 70
From: <sip:1445169631 at 192.168.9.230>;tag=1c1714851315
To: <sip:2753009 at 192.168.9.210;user=phone>
Call-ID: 171485058731200032953 at 192.168.9.230
CSeq: 1 INVITE
Contact: <sip:1445169631 at 192.168.9.230>
Supported: em,100rel,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,
UPDATE
Remote-Party-ID: <sip:2753009 at 192.168.9.210>;party=called;npi=1;ton=2
Remote-Party-ID: <sip:1445169631 at 192.168.9.210>;party=calling;privacy=off;screen
=yes;screen-ind=3;npi=1;ton=0
User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.4.60A.016.003
Content-Type: application/sdp
Content-Length: 350

Here I am registering with username 3911700, calling to 2753009, from 1445169631

As you can see, the From: is the telephone number of the caller. When
I normally connect with pjsua, the From: header is the login id, and
To: is the number to call. Is there any way to change the From: so
that I can write a script to do a regression test of asterisk
configuration when I make changes?

I am reasonably proficient with python, so I can use the python
bindings and write a program if required.

raj




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