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Hello, After watching Matt Jordan's presentation at Kamailio World Conf in 2016 , I decided to switch architecture chan_sip to pjsip. So i started to testing Pjsip that is suitable for our system because There is always a feature that can be forgetten , missing or has a bug. First of all , i realized that to header manipulation is removed with exclamation mark in Pjsip. I tried to some configuration with outbound parameters.But i failed. Maybe , i couldnt find way to change it. In Addition , i realized that it changes to header with the same as Request Uri Number and adding "asterisk" to Contact header instead of Number! In conclusion , I already sent a topic to Forum and then couldn't find solution with jcolp So is it possible to Add a Function about Setting To header Number like CallerId In Pjsip? How can we solve this problem? I dont want to add a purge about To header in Kamailio because it can be breakable on ACK ,200-OK or other transacations. Asterisk is so good about dialog transacations.If i try to add a function it , which ways is acceptable? Why i am trying to do that? Because some kind of FXS devices need to waits Request Uri Number and To header Number and Contact Header Name must be same ,if not it declines the calls.Kamailio Modules can only remove /add prefix on Request Uri Number. Thanks for Helps. Yasin CANER Here is Flow; AsteriskIP:5060 ---> KamailioIP:5060 x INVITE (SDP) x xINVITE sip:102105066109057atKamailioIP:5060 SIP/2.0 xVia: SIP/2.0/UDP AsteriskIP:5060;rport;branch=z9hG4b xFrom: "8503023423" <sip:8503023423atAsteriskIP>;tag=a21fc xTo: <sip:102105066109057atKamailioIP> xContact: <sip:asteriskatAsteriskIP:5060> xCall-ID: d170e273-6bcc-474b-9816-a6b884419ff2 xCSeq: 23680 INVITE xRoute: <sip:KamailioIP;lr> xAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE xUser-Agent: Asterisk PBX 14.0.1 xContent-Type: application/sdp xContent-Length: 259 KamailioIP:5060 --> AsteriskIP:5060 x x xINVITE sip:102105066109057atAsteriskIP:5060 SIP/2.0 xRecord-Route: <sip:KamailioIP;lr;ftag=c963d657> Via: SIP/2.0/UDP KamailioIP;branch=z9hG4bKcdef.f xVia: SIP/2.0/UDP 192.168.0.223:64556 xContact: <sip:8503023423atUac1IP:3321;transport=UDP> xTo: <sip:05066109057atKamailioIP;transport=UDP> xFrom: "8503023423"<sip:8503023423atKamailioIP;transport=UDP>;tag=c963d6 xCall-ID: 2tiU_X3S8_qbX3K0nOFYeQ.. xCSeq: 2 INVITE xAllow: INVITE, ACK, CANCEL, BYE Here is architecture; Kamailio -> registrar server , location server , edge server ... Asterisk -> Application server , RTP server ... ||||||| | UAC1| ||||||| ^ | | |||||||||||| |||||||||||||| | |<---------- | | | Kamailio | | Asterisk* | | |----------> | | |||||||||||| |||||||||||||| ^ | | | ||||||| | UAC2| ||||||| Here is my pjsip.conf [global] max_forwards=30 user_agent=TEST keep_alive_interval=60 [simpletrans] type=transport protocol=udp bind=AsteriskIP [kamailio] type=endpoint transport=simpletrans context=netgsm disallow=all allow=ulaw allow=alaw ;outbound_proxy=sip:kamailioIP outbound_proxy=sip:kamailioIP\;lr [kamailio] type=identify endpoint=kamailio match=kamailioIP --
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