[asterisk-biz] SIP Trunking Question

[Michael Young]

I compete with bandwidth.com, so I hate to help ;-) ... but make sure
you are looking for the incoming call as +14075173015 (for instance,
based on your signature) -- I believe the plus gets sent to you and that
usually catches people.

Michael Young
NetLogic

-----Original Message-----
From: asterisk-biz-bounces at lists.digium.com
[mailto:asterisk-biz-bounces at lists.digium.com] On Behalf Of Ryan M.
Colbert
Sent: Tuesday, July 03, 2007 5:04 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: [asterisk-biz] SIP Trunking Question

This may be the wrong forum for my question, and if so, please forgive
my error.

I have been working on setting up a SIP trunk from Bandwidth.com for
almost a week now. Outgoing calls are working fine but I can't seem to
get the inbound calls to process.  Would anyone be willing to share a
working extensions.conf file?  I set the context ok in sip.conf and can
see the initial connection come in. I think my trouble is in parsing and
passing the DID string.

Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue & McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando, FL 32801
(407) 517-3105 - Direct Telephone
(407) 839-0120 - Main Office
(407) 841-9726 - Fax
www.rissman.com

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