Hi all, Let me first describe to you our environment -
Asterisk Server (xxxxxx.jacksongrp.com) --------------------------------------- in non-natted environment w/ public IP Office 1 - Millersville, MD --------------------------- - Natted environment - Cisco 7960 telephone, registered with asterisk successfully as '200' Office 2 - Annapolis, MD ------------------------ - Natted Environment - Cisco ATA-186 bridge, both lines on device registered as '203' & '204' I can successfully place a call to extension 203, from Office 1 (200). I cannot place a call to extension 200, from Office 2 (203) - this call is routed directly to vmail. I can, however, successfully dial out Office 2 (203) to our SIP provider, and make a phone call to a regular telephone. Running SIP SHOW PEERS at CLI> yields: 204/204 162.33.157.88 (D) 255.255.255.255 55275 Unmonitored 203/203 162.33.157.88 (D) 255.255.255.255 55275 Unmonitored 202 (Unspecified) (D) 255.255.255.255 0 Unmonitored 201 (Unspecified) (D) 255.255.255.255 0 Unmonitored 200/200 68.50.239.4 (D) 255.255.255.255 58031 Unmonitored sipdemo/1637 66.159.92.6 255.255.255.255 5060 Unmonitored 1637/1637 66.159.92.6 255.255.255.255 5060 Unmonitored Regards, Phillip _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
