[OpenSIPS-Users] Internal calls problem

[misme Gazeta.pl]

I just have installed opensips and I'm tring to configure it to make calls
like in this scenario: sips registered user -> sips -> asterisk -> sips ->
sips registered user (I need asterisk to make transcode and bill call). I
have used nathelper.cfg config from example with some modifications: a) I
have add modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock", b)
also every time when in config is "route(1);" i have change it to: if(src_ip
== 'IP_OF_MY_ASTERISK'){ route(1); else{ route(2); } } route(2) is:
force_rtp_proxy();
rewritehostport("IP_OF_MY_ASTERISK:5060"); t_relay(); so I expect that when
I make call to sips registered user from other than asterisk IP, it will be
switched to asterisk (and then asterisk swtich back to sips and then to
user) in other case it will connect to sips registered user, but it not
works every time. I have tested in like this: X-Lite = sips user 1 (my local
IP) Grandstream HT502 gateway = sips user 2 (my local IP - same as X-Lite)
SIPS - on public IP Asterisk - on public IP (diferent than SIPS, but on the
same server) When I make call: X-Lite -> Grandstream (via sips) it works
fine but when I make call: Grandstream -> X-Lite (via sips) it dosnt goes
throu asterisk (in asterisk logs there is no info about this), also there is
one way audio from granstream to x-lite (and no audio from x-lite to
grandstream). Do you have any idea what is the problem?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20100926/1e905e8c/attachment.htm