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 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 >-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671
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
----- Original Message ----> From: Daniel Cole <dcole at hcit.com.au> > To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> > Sent: Tuesday, 29 January, 2008 10:31:55 PM > Subject: [asterisk-users] Source Based Call Routing > > 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.>Hi Daniel, 3 different contexts in your dial plan would work. Assign each block of accounts (rather than extensions) to the context with the routes that they should use. To change an account from using one trunk to another it would be as simple as changing its context. Regards, Greyman. Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail
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 plan....hmmmm..... PaulH On Wed, 2008-01-30 at 09:31 +1100, 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
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 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 > plan....hmmmm..... > > PaulH > > > On Wed, 2008-01-30 at 09:31 +1100, 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 >-- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671
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
-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/x-pkcs7-signature
Size: 3295 bytes
Desc: S/MIME Cryptographic Signature
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20080130/9a5bd7fa/attachment.bin