Daniel,
attach a dialplan variable to each extension using setvar
in sip.conf:
[6318]
type=friend
username=6318
secret=xx
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 ${use
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Source Based Call Routing
- Original Message
> From: Daniel Cole <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Sent: Tuesday, 29 January, 2008 10:31:55 PM
> Sub
I would still say the easiest thing by far is to introduce a mediator
in the dial plan that is far more intelligent and extensible than the
dial plan logic itself. Enter FastAGI. Then you can just do it ...
however you want.
On Wed, 30 Jan 2008, Paul Hales wrote:
>
> You can also look at routi
You can also look at routing based on number ranges (if you keep the
separate numbers in separate number ranges) but I would guess that this
is not going to suit your needs.
Maybe storing all the accounts in mysql (realtime) would also be a good
planh.
PaulH
On Wed, 2008-01-30 at 0
- Original Message
> From: Daniel Cole <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Sent: Tuesday, 29 January, 2008 10:31:55 PM
> Subject: [asterisk-users] Source Based Call Routing
>
> Hi List,
>
> I have a sc
I would broker the dial-out requests through FastAGI and put the logic
that examines extensions and implements the load balancing / distribution
in there.
On Wed, 30 Jan 2008, Daniel Cole wrote:
> Hi List,
>
> I have a scenario that I want to try out (we potential have a client who
> would need
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 s
