What I am trying to do is the following: A call is sent to the * box
via a SIP invite. The * box answers via an IVR menu system with "
Enter the extension you want to dial" so I enter in my 5 digit
extension and get the below message.
Jan 18 10:10:03 WARNING[-1380238416]: channel.c:1860 ast_request: No
channel type registered for 'SIP)'
Jan 18 10:10:03 NOTICE[-1380238416]: app_dial.c:696 dial_exec: Unable
to create channel of type 'SIP)'
Jan 18 10:10:05 WARNING[-1115923536]: chan_sip.c:673 retrans_pkt:
Maximum retries exceeded on call
[EMAIL PROTECTED]
for seqno 1 (Non-critical Response)
My extension.conf outbound dial peer:
[outbound]
exten => _124XX,1,Dial(SIP)/${EXTEN:[EMAIL PROTECTED])
exten => _124XX,2,Playback(invalid)
exten => _124XX,3,Hangup
My sip.conf
[outbound]
type=peer
host=192.168.1.1
What the * needs to do is receive the call via SIP and then send it
out dialed extension via SIP to an another IP PBX. SO the * does not
need to register to a server just blindly send a SIP invite to the ip
address in the SIP.CONF file: 192.168.1.1
Any help would be appricated
Kurt
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
