Here's what I ultimately got to work (in case it helps someone): Name your trunk Enter your outgoing CID
Under Outgoing settings- Trunk name - whatever you choose to name it PEER Details- host=IP address of SIP gateway type=friend context=from-trunk disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite Incoming settings - none Registration string - username:[email protected] James Cass <http://goog_987864563> [email protected] On Tue, Feb 23, 2016 at 12:20 PM, Rodrigo Ramírez Norambuena < [email protected]> wrote: > February 23 2016 9:37 AM, "James Cass" <[email protected]> wrote: > > Thanks everyone, all sound advice. Still can't even get the calls to > show up on the console at all > > - I suspect the issue is on the WS side, as I'm not having any issues > with other carriers with > > similar settings. > > You can debug SIP to detect the problem. May be exists some cause tell you > more information in the > trace SIP. > -- > Rodrigo Ramírez Norambuena > http://www.rodrigoramirez.com > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
