Christopher Aloi wrote:
Hello -
I currently have 10 DID's coming into one Asterisk server, I seem to be
having some difficulty routing based on the DID dialed and am hoping
someone
on the list can assist me.
<snip>
Unless I'm misunderstanding you, how about trying this:
1. In your sip.conf:
[general]
useragent=Asterisk
port=5060
context=default
tos=lowdelay
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtptimeout=300
rtpholdtimeout=600
2. In your extensions.conf:
[default]
exten => s,1,Goto(${CALLERIDNUM},s,1)
[123456789]
exten => s,1,Answer()
exten => s,2,Playback(beep)
exten => s,3,GoTo(queue-test,s,1)
So if you get an incoming SIP call from 123456789, it enters the "default"
context and is then routed to the "123456789" context.
Flynn
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