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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
