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@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
