It’s been a while since I’ve setup a voip provider, so I’m sure I’m overlooking something obvious here. I have my provisioning information from digium for the sip-trunk. I can either get incoming working, or outgoing, but never both.
The problem lies with the ‘host’ config option in the sip stanza for the service. If I define a host, incoming won’t work but outgoing will ( kinda, the call connects successfully, but no audio is passed in either direction. ). If I leave the host option out ( or set it to dynamic ), incoming works but outgoing doesn’t ( presumably because asterisk doesn’t know where to send the call. Notably, however, audio works perfectly in both directions ). In both instances, I have a
[quote]register => :@sip.digiumcloud.net[/quote] statement.
Working incoming stanza[quote][digium-in]
Working outgoing stanza[quote]
If I pull out the digium-out stanza, incoming starts working. If I put it back in, outgoing works but incoming doesn’t.
Any input would be greatly appreciated.