I have a question on Asterisk and whether it will work with the following design. Install ASTERISK on the external side of the Network. Purchase an AudioCodes 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here is the twist. The company currently has Cisco Call Manager 3.3 which does not support SIP Trunking. But it does have a VG248. I would like to place 4 lines through the Cisco Call Manager. I want to set up a dial plan where 7 would grab the fx0 line for internal and the users would be able to place internal calls through the Cisco Call Manager. I envision people dialing 7xxxx (4 digit extension.) This would call internally. I then envision setting up a calling plan where 7 would grab the trunk and 8 would grab an outside line from the Cisco Call Manager and then dial the 10 digit telephone number. 78xxxxxxxxxx. This would allow them to place external calls through the call manager. Is this something that would be feasible? Since the company is not looking to invest a lot in upgrading the Cisco yet they want to allow external sales reps to work from home. Would there be a way through Asterisk where I can then program the FX0 extension coming in from the Cisco Call Manager to ring into the Audiocodes and be dialed directly to an extension in the Asterisk server? Example ----- 1300-------200 on the Asterisk. This would allow people calling the company to directly dial their sales people and be forwarded to the extension attached to the audiocodes. If this is feasible please let me know as I would like to propose this solution to the company. Thanks.
Gary Richardson
2006-Jan-26 11:58 UTC
[Asterisk-Users] Asterisk Setup Question -- Please Help
I've had success using oh323 to create a trunk between a CCM3.2 and
asterisk.. I wouldn't put any analog devices in your asterisk box if
you already have CM.
You could also move completely to Asterisk and flash your 79xx's into
SIP mode and turn your Cisco boxes off.
BTW, you could also have asterisk just do something like:
exten => _9XXXXXXXXXX,1,Dial(OH323/${EXTEN}@yourcallmanager)
So you don't have to do fancy 78 stuff to get an outside line. I'm
also playing around with something like:
exten => 1234,1,Dial(SIP/SomeUser)
exten => 1235,1,Dial(SIP/SomeOtherUser)
; we didn't find any users here .. let's try Cisco
exten => _XXXX,1,Dial(OH323/${EXTEN}@yourcallmanager)
On this Cisco side, when I move people to Asterisk, I'll just hard
code a dial plan to the h323 gate on my asterisk box. Then, I can
flash hardphone users one at a time and get them using asterisk as
their call server at my leisure (or not move people at all).
It's mostly working now, I'm just having problems with including
contexts and precedence -- I haven't quite figured it all out yet. I
hope that's a source of inspiration :)
On 1/25/06, Goran Donev <gorand@dvvti.com> wrote:> I have a question on Asterisk and whether it will work with the following
> design.
>
>
> Install ASTERISK on the external side of the Network. Purchase an
AudioCodes
> 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here
> is the twist.
>
> The company currently has Cisco Call Manager 3.3 which does not support SIP
> Trunking. But it does have a VG248. I would like to place 4 lines through
> the Cisco Call Manager.
>
> I want to set up a dial plan where 7 would grab the fx0 line for internal
> and the users would be able to place internal calls through the Cisco Call
> Manager. I envision people dialing 7xxxx (4 digit extension.) This would
> call internally.
>
> I then envision setting up a calling plan where 7 would grab the trunk and
8
> would grab an outside line from the Cisco Call Manager and then dial the 10
> digit telephone number.
>
> 78xxxxxxxxxx. This would allow them to place external calls through the
call
> manager. Is this something that would be feasible?
> Since the company is not looking to invest a lot in upgrading the Cisco yet
> they want to allow external sales reps to work from home.
>
> Would there be a way through Asterisk where I can then program the FX0
> extension coming in from the Cisco Call Manager to ring into the Audiocodes
> and be dialed directly to an extension in the Asterisk server?
>
> Example ----- 1300-------200 on the Asterisk.
> This would allow people calling the company to directly dial their sales
> people and be forwarded to the extension attached to the audiocodes.
> If this is feasible please let me know as I would like to propose this
> solution to the company.
>
> Thanks.
>
>
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