[SR-Users] Call transfer between pbx servers behind kamailio proxy

[Iskren Hadzhinedev]

Hi everyone,
I'm having a hard time transferring calls when the users are on 
different FreeSWITCH servers behind kamailio.

The users are distributed as follows:
user1 at sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1
user2 at sip.test.com-> Kamailio - ds_select_dst(1,1) -> FreeSWITCH1
user3 at sip.test.com -> Kamailio - ds_select_dst(1,1) -> FreeSWITCH2

When user1 calls user2 and user2 does an attended transfer to user3, the 
following happens:
285  67.220453  USER2_IP -> KAMAILIO_IP SIP 726 Request: REFER 
sip:mod_sofia at FreeSWITCH1_IP:5060, in-dialog |
286  67.220674 KAMAILIO_IP -> FreeSWITCH1_IP SIP 877 Request: REFER 
sip:mod_sofia at FreeSWITCH1_IP:5060, in-dialog |
287  67.223651 FreeSWITCH1_IP -> KAMAILIO_IP SIP 925 Status: 202 Accepted |
288  67.223753 KAMAILIO_IP -> USER2_IP  SIP 838 Status: 202 Accepted |
289  67.225502 FreeSWITCH1_IP -> KAMAILIO_IP SIP/SDP 1225 Request: 
INVITE sip:user3 at sip.test.com |
290  67.387323 KAMAILIO_IP -> 69.172.200.109 SIP/SDP 1420 Request: 
INVITE sip:user3 at sip.test.com |

Turns out 69.172.200.109 is the actual IP address of sip.test.com (I've 
probably should have gone with sip.example.com or something else).
What is the proper way to route that INVITE back to the other FreeSWITCH 
server so that the transfer would work?
Thanks!

Kind regards,
-- 
/Iskren Hadzhinedev/

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