Thanks for your response.
What I found is:
1. If call is from phone registered to IP (external or internal) - then I
do not need any of my modifications - ACK goes through loose_route,
or t_check_trans() is OK and ACK is also OK.
2. If call is from phone registered to name (sip.mycompany.com) - th
Hi Mihaylov,
If your Asterisk servers add a Record-Route header to the initial
Invite, for in-dialog requests ( ACK, BYE) you should use *loose_route()
*function to do the routing. This will make sure the requests go the
same path as the initial Invite. It is not a good practice to manually
r
My whole configuration is:
[Sip clients] < = > Kamailio 3.2 <=> Asterisk servers (behind Kamailio)
Asterisk servers have only local IP addresses, and I use t_relay instead of
forward.
Kamailio runs on same server as rtpproxy.
Everything is fine if clients connect to Kamailio with its IP address -
g
