Hi Misme,
From the simptoms that you describe, I don't think it can be something
wrong with the opensips configuration ( since it works from xlite and
not from Grandstream). Maybe there is something wrong with the phones
configuration. Have you set correctly the outbound proxy in Grandstream?
I suggest you to monitor the traffic at the server and check which are
the SIP messages received from Grandstream.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 09/26/2010 11:30 PM, misme Gazeta.pl wrote:
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?
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