I have a Vonage Softphone line, and I got instructions on how to get it to work as a SIP Trunk in FreePBX/Asterisk. However, when I attempt to call out, I get a “All Circuits are Busy Now” message. It will not even try to place the call. I got the code from http://www.pbxinfo.com/asterisk-systems/21509-vonage-softphone-line-trunk-asterisk.html . I also saw http://www.voip-info.org/wiki/view/Asterisk+and+Vonage but I did not know what the author meant by his “sip.conf”. Can anyone help me? It will be much appreciated!
Logs:
root@pbx:~ $ asterisk -r
Asterisk 1.4.21.2, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for detail
s.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
Connected to Asterisk 1.4.21.2 currently running on pbx (pid = 3940)
Verbosity is at least 3
pbx*CLI> sip show registry
Host Username Refresh State Reg.
Time
sphone.vopr.vonage.net:5061 18433144226 120 Unregistered
pbxCLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
4357 (Unspecified) D N 0 UNKNOWN
2002 (Unspecified) D N 0 UNKNOWN
2001/2001 192.168.0.199 D N 46434 OK (13 ms)
3 sip peers [Monitored: 1 online, 2 offline Unmonitored: 0 online, 0 offline]
pbxCLI>
root@pbx:~ $