asterisk users - Jun 2006 - Help with incoming SIP routing

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.

Here's the relevant info:

Ingress SIP trunk:
IP: 123.45.45.3456
DID's XXX-XXX-XX00-XX10

sip.conf:

[general]
useragent=Asterisk
port=5060
context=default
tos=lowdelay
disallow=all
allow=ulaw
allow=alaw
allow=gsm
rtptimeout=300
rtpholdtimeout=600

// // My thought in this context is I will grab any incoming SIP call from
the IP address of my SIP trunk and pass it to my sip-defaul-in context // //
in extensions.conf

[sip-default-in]
type=friend
defaultip=123.45.3456
host=123.45.3456
nat=no
insecure=very
context=sip-default-in
canreinvite=no
dtmfmode=rfc2833

// //  My thought here is I will grab any incoming SIP call form the IP
address of my SIP trunk that matches XXX-XXX-XX00 and pass it to my
XXX-XXX-XX00 context in extensions.conf

[XXXXXXXX00]
type=friend
defaultip=69.67.248.51
host=69.67.248.51
fromuser=XXXXXXXX00
nat=no
context=XXXXXXXX00
insecure=very

And a look at extesions.conf:

// // My thought is here I will route my incoming calls to a DID i haven't
specifically routed to my default context (GoTo(XXX))

[sip-default-in]
exten => s,1,Answer()
exten => s,2,Playback(beep)
exten => s,2,Ringing
exten => s,3,Wait,1
exten => s,4,GoTo(XXX)

// // My thought here is I will handle my incoming calls to XXX-XXX-XX00 and
pass it to a specific context, say a queue

[XXXXXXXX00]
exten => _XXXXXXXX00,1,Answer()
exten => _XXXXXXXX00,2,Playback(beep)
exten => _XXXXXXXX00,3,GoTo(queue-test,s,1)

What am I doing wrong??

I can receive calls fine, but they aren't routing properly....I think I
overlooked something.

Thanks list!!

/Chris
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/9500d22e/attachment.htm

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