27 Mar
2006
27 Mar
'06
7:42 p.m.
Hello
I have this setup, but with Asterisk listening on port 1720.
To forward calls to Asteirks I have this:
#calls with 4 digits go to route 12
if (uri=~"^sip:[0-9]{4}@ourdomain.pt") {
record_route();
route(12);
break;
};
# route 12 sends them to asterisk
route[12] {
rewritehostport("Asterisk_IP:1720");
if (!t_relay()) {
sl_reply_error();
};
}
Do you have something like this?
Joao
Corey S. McFadden wrote:
Hi,
I understand this is a fairly typical setup for Voicemail and media but
we're running OpenSER and Asterisk on the same box. (OpenSER on 5060 and
Asterisk on 5061.) We also use rtpproxy.
Here are the scenarios:
UA -> OpenSER -> UA (no problems)
UA -> OpenSER -> External Asterisk PSTN Gateway (no problems)
UA -> OpenSER -> External Asterisk Voicemail (no problems)
UA -> OpenSER -> Same box Asterisk Voicemail - NO AUDIO
I need to get this running on the same box and am not sure what to do
next. This is probably something silly / obvious, so if anyone can point
me in the right direction it would be greatly appreciated!
Thanks in advance,
-Corey
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