Confirmed, I removed the r from my dial command and it connects now.
Thank you for the tip!
Luki wrote:
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 [EMAIL PROTECTED]
[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
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--
Mojo <[EMAIL PROTECTED]>
Office Manager, Horan & Company, LLC
(907) 747-6666 x112
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