Hi all,
I had setup an Opensips (3.0) instance (as Registrar) on an Amazon EC2
instance and freeswitch (media server) on another instance. I had tested it
thoroughly and it seems to work great on the following code (Opensips.cfg):
if (!is_method("INVITE")) {
return;
}
# if the called number begins with "star" (*) then strip it and redirect to
freeswitch
# (if it begins with two stars, eg: **, then one will be passed to FS)
if ($rU=~"^\*") {
strip(1);
$du = "sip:172.31.23.7:5090";
route(relay);
}
}
I dial the IVR - everything works ok for MicroSIP (laptop) and Linphone
(Android phone).
However, in the following snippet - MicroSIP (call disconnects immediately)
and Linphone has no audio (disconnects after 32 seconds). Pcap shows no
connection on MicroSIP but on Linphone I can play the audio stream.
if(is_method("INVITE")) {
if (t_check_status("408|486")) {
$du = "sip:172.31.23.7:5090";
# do not set the missed call flag again
route(relay);
}
Is there a difference in the headers/routing sent out in
"if(is_method("INVITE"))" vs "if(!is_method("INVITE"))". And how to fix
please?
Please let me know if any other information is required.
Thanks again.
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
