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abalashov at evaristes...
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 Posted: Tue Jan 29, 2008 5:28 pm Post subject: [asterisk-users] Source Based Call Routing |
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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:
| Quote: | 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
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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
Direct : +1-678-954-0671 |
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dcole at hcit.com.au
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| Posted: Tue Jan 29, 2008 5:31 pm Post subject: [asterisk-users] Source Based Call Routing |
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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 |
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greyvoip at yahoo.com.au
Guest
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| Posted: Tue Jan 29, 2008 5:38 pm Post subject: [asterisk-users] Source Based Call Routing |
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----- Original Message ----
| Quote: | 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
it.
| Quote: |
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
setup back to providers. What we want to do is to be able to
groups of extensions that use specific outbound trunks.
| Quote: |
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
control over assigning with trunks the given extensions will use.
| Quote: |
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 |
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pdhales at optusnet.co...
Guest
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| Posted: Tue Jan 29, 2008 6:14 pm Post subject: [asterisk-users] Source Based Call Routing |
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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:
| Quote: | 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 |
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abalashov at evaristes...
Guest
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| Posted: Tue Jan 29, 2008 6:19 pm Post subject: [asterisk-users] Source Based Call Routing |
|
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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:
| Quote: |
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:
| Quote: | 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 |
|
| Back to top |
|
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dcole at hcit.com.au
Guest
|
| Posted: Tue Jan 29, 2008 6:34 pm Post subject: [asterisk-users] Source Based Call Routing |
|
|
Thank you Greg and Alex for your contribution.
I will use your leads to see what I can get asterisk to do 
Many Thanks,
Daniel
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Grey Man
Sent: Wednesday, 30 January 2008 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Source Based Call Routing
----- Original Message ----
| Quote: | 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
it.
| Quote: |
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
setup back to providers. What we want to do is to be able to
groups of extensions that use specific outbound trunks.
| Quote: |
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
control over assigning with trunks the given extensions will use.
| Quote: |
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
_______________________________________________
-- 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 |
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| Back to top |
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ron.arts at neonova.nl
Guest
|
| Posted: Wed Jan 30, 2008 2:25 am Post subject: [asterisk-users] Source Based Call Routing |
|
|
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:
| Quote: | 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
|
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