[Asterisk-Users] Help with incoming SIP routing

[El Flynn]

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