Re: Source Based Call Routing

1 msg	Best practice security for internet access toAs...
5 msg	asterisk gateway
1 msg	Queue works with across server agent?
2 msg	Queue - ${ANSWEREDTIME}
6 msg	chanspy does not pull the call back to asterisk...

Source Based Call Routing
\ Daniel Cole (29 Jan 2008)
. \ Alex Balashov (29 Jan 2008)
. \ Grey Man (29 Jan 2008)
. . \ Daniel Cole (29 Jan 2008)
. \ Paul Hales (29 Jan 2008)
. . \ Alex Balashov (29 Jan 2008)
. \ Ron Arts (30 Jan 2008)

1 msg	ShoreTel <-> Asterisk Integration
3 msg	Queue member add
2 msg	speex, ilbc and g729 codecs
3 msg	When does Asterisk 'REFER'?
3 msg	Re: POE draw on Aastra 480i
5 msg	When can I AIG?
2 msg	softmodems bank for ast.
3 msg	Asterisk 1.4.18-rc2 Now Available
2 msg	codec_g729a.so problem...
3 msg	SET with pipe symbol
2 msg	test please ignore
2 msg	PRI Alarms, Comes Back,But Asterisk Won't Touch...
10 msg	transcoder
3 msg	Do Asterisk requires audio codec to be installed?

Subject:	Re: Source Based Call Routing
Group:	Asterisk-users
From:	Ron Arts
Date:	30 Jan 2008

Daniel,

attach a dialplan variable to each extension using setvar
in sip.conf:

[6318]
type=friend
username=6318
secret=xxxxxx
host=dynamic
nat=no
dtmfmode=rfc2833
qualify=0
amaflags=billing
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=phone
setvar=__usetrunk=1

you can use the ${usetrunk} variable in your dialpan.

Ron

Daniel Cole wrote:
> Hi List,
>
> I have a scenario that I want to try out (we potential have a client who would need this), but I am as of yet unable to find much help with it.
>
> What we want to do is have an asterisk box with a large number of extensions (1000+). This asterisk box will have approximately 3 SIP trunks setup back to providers. What we want to do is to be able to define groups of extensions that use specific outbound trunks.
>
> Approximately a third of the extensions will one the first trunk, a third the second trunk, and the rest will use the last trunk. We also need control over assigning with trunks the given extensions will use.
>
> Any suggestions on how to get this to work would be very much appreciated.
>
>
> Many Thanks,
>
> Daniel
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users