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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