[asterisk-users] IAX or SIP termination provider that reaches6421xxxxxxx?

[Luki]

> I'm interested if anyone else in the Asterisk list can get
> through to +1-907-747-8633 via voip

Sure, no problem. A nice friendly female voice tells you the time and
temp, indeed. The thing is that the call never connects -- that info
is sent via call progress, so a misconfigured server (i.e. one that
uses the "r" option in dial() or equivalent) would just give you
ringing and ringing...

[Sep 21 17:49:45]     -- Called 9077478633 at trunks
[Sep 21 17:49:45]     -- SIP/trunks-094da090 is making progress
passing it to SIP/1001-b7a030f8
[Sep 21 17:49:48]     -- Ringing
[Sep 21 17:49:48]     -- Progress
[Sep 21 17:49:48]     -- Peer audio RTP is at port 1.2.3.4:12345

etc.

--Luki