[SR-Users] SIP-Trunk Kamailio and Asterisk

[Володимир Іванець]

Hello everyone,

I'm doing some custom Kamailio configuration to achieve few things but seem
to be stuck with SIP trunks. I'm currently using Kamailio 4.4.5 and here is
what I have at the moment:

*1.* Asterisk 11.25.1 servers behing Kamailio using RealTime for SIP peers.

*2.* Route that checks if packet was sent from one of configured SIP
trunks. This part works correctly:

*route[FROMPSTN] {*
*        xlog("== FROMPSTN ==\n");*
*        $var(socket) = $si + ":" + $sp;*
*        xlog("**** $var(socket)\n");*
*        xlog("**** $(sht(trunks_kamailio=>$var(socket)))\n");*
*        if ($(sht(trunks_kamailio=>$var(socket))) != $null) {*
*                xlog("**** From $(sht(trunks_kamailio=>$var(socket)))
PSTN\n");*
*                return 1;*
*                }*
*        xlog("**** Not from PSTN\n");*
*        return -1;*
*}*


*3.* Here is a fragment that sends INVITE packet to *TOASTERISK* route if
it was received from one of trunks. I have 4444444444 at XXX.XXX.XXX.XXX
hardcoded for now, and in my configuration 4444444444 and XXX.XXX.XXX.XXX
represent my real caller id number and carrier's IP address:

*if(route(FROMPSTN)) {*
*        uac_replace_from("sip:4444444444 at XXX.XXX.XXX.XXX");*
*        route(TOASTERISK);*
*}*


*4.* TOASTERISK basically has folloving fragment that load balances traffic
between Asterisk nodes:

*if(!ds_select_dst("0", "4")) {*
*        xlog("**** Call was not sent to none of Asterisk hosts\n");*
*        if(!is_method("REGISTER")) {*
*                xlog("**** Method is not REGISTER\n");*
*                send_reply("404", "No destination");*
*        }*
*        xlog("**** Dispatcher fail\n");*
*        exit;*
*}*


So when I'm making a call to my test number, I can see following incomming
packet in Asterisk from Kamailio. From header has correct caller id number
and carrier's IP address (192.168.88.5 is Kamailio's internal IP address):
[image: Вбудоване зображення 1]

And when Asterisk tries to find a maching peer for this incoming call, it
is using Kamailio's instead of Carrier's IP address. So it is actually
finds one of peers (101-XXXXXXX) that is being authenticated on Kamailio as
many others:
[image: Вбудоване зображення 2]

I'm not sure if Asterisk is just using packet's source IP to lookup for
corresponding peer or something else from it's content. I will really
appreciate any help on this.

Thanks a lot!
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