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<font face="Courier New, Courier, monospace">Hello,<br>
<br>
After watching Matt Jordan's presentation at Kamailio World Conf
in 2016 , I decided to switch architecture chan_sip to pjsip.<br>
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. <br>
First of all , i realized that to header manipulation is removed
with exclamation mark in Pjsip.<br>
I tried to some configuration with outbound parameters.But i
failed. Maybe , i couldnt find way to change it. <br>
<br>
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!<br>
<br>
<br>
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?<br>
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?<br>
<br>
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.<br>
<br>
<br>
Thanks for Helps.<br>
<br>
Yasin CANER<br>
<br>
<br>
<br>
<br>
Here is Flow;<br>
<br>
AsteriskIP:5060 ---> KamailioIP:5060<br>
x INVITE (SDP) x<br>
<br>
<br>
xINVITE sip:102105066109057atKamailioIP:5060 SIP/2.0 <br>
xVia: SIP/2.0/UDP AsteriskIP:5060;rport;branch=z9hG4b <br>
xFrom: "8503023423" <sip:8503023423atAsteriskIP>;tag=a21fc
<br>
xTo: <sip:102105066109057atKamailioIP><br>
xContact: <sip:asteriskatAsteriskIP:5060><br>
xCall-ID: d170e273-6bcc-474b-9816-a6b884419ff2<br>
xCSeq: 23680 INVITE<br>
xRoute: <sip:KamailioIP;lr><br>
xAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE<br>
xUser-Agent: Asterisk PBX 14.0.1<br>
xContent-Type: application/sdp<br>
xContent-Length: 259<br>
<br>
<br>
KamailioIP:5060 --> AsteriskIP:5060<br>
x x<br>
<br>
xINVITE sip:102105066109057atAsteriskIP:5060 SIP/2.0<br>
xRecord-Route: <sip:KamailioIP;lr;ftag=c963d657><br>
Via: SIP/2.0/UDP KamailioIP;branch=z9hG4bKcdef.f<br>
xVia: SIP/2.0/UDP 192.168.0.223:64556<br>
xContact: <sip:8503023423atUac1IP:3321;transport=UDP><br>
xTo: <sip:05066109057atKamailioIP;transport=UDP><br>
xFrom:
"8503023423"<sip:8503023423atKamailioIP;transport=UDP>;tag=c963d6<br>
xCall-ID: 2tiU_X3S8_qbX3K0nOFYeQ..<br>
xCSeq: 2 INVITE<br>
xAllow: INVITE, ACK, CANCEL, BYE<br>
<br>
<br>
<br>
<br>
<br>
Here is architecture;<br>
Kamailio -> registrar server , location server , edge server
...<br>
Asterisk -> Application server , RTP server ...<br>
<br>
<br>
|||||||<br>
| UAC1|<br>
|||||||<br>
^<br>
|<br>
|<br>
|||||||||||| ||||||||||||||<br>
| |<---------- | |<br>
| Kamailio | | Asterisk* |<br>
| |----------> | |<br>
|||||||||||| ||||||||||||||<br>
^<br>
|<br>
|<br>
|<br>
|||||||<br>
| UAC2|<br>
|||||||<br>
<br>
<br>
<br>
<br>
Here is my pjsip.conf<br>
<br>
[global]<br>
max_forwards=30<br>
user_agent=TEST<br>
keep_alive_interval=60<br>
<br>
[simpletrans]<br>
type=transport<br>
protocol=udp<br>
bind=AsteriskIP<br>
<br>
[kamailio]<br>
type=endpoint<br>
transport=simpletrans<br>
context=netgsm<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
;outbound_proxy=sip:kamailioIP<br>
outbound_proxy=sip:kamailioIP\;lr<br>
<br>
[kamailio]<br>
type=identify<br>
endpoint=kamailio<br>
match=kamailioIP<br>
</font><br>
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