[Asterisk-Users] PRI for Data and Voice
Sergey Kuznetsov
asterisk_biz at deeptown.org
Sat Jan 29 12:58:58 MST 2005
I met Sangoma guys this Tuesday, and got a AFT102 for evaluation.
Right now I am in progress to develop a * channel driver for AFT10* devices.
In that case you will have much more flexibility and to use all their API.
Steven Critchfield wrote:
>On Sat, 2005-01-29 at 11:45 -0500, Jim Van Meggelen wrote:
>
>
>>David Norton wrote:
>>
>>
>>>Hi,
>>>
>>>Currently I only have 1 PRI which I am using for dial-in customers.
>>>The line is connected to a Portmaster3. I have never used more than
>>>10 concurrent channels. The calls can be both analog or ISDN. It
>>>would be a waste to order another PRI for my Asterisk box. Is there
>>>any way of splitting a PRI into 2 PRI’s of 15 channels each, or
>>>plugging the PRI into the * box and it send the data calls to the
>>>portmaster, or handles them itself?
>>>
>>>Any advice would be much appreciated
>>>
>>>
>>I betcha Sangoma has something that'd do this for you. They've been
>>supporting T1 data on Linux for years, and they're recently added zapata
>>to their list of open-source drivers.
>>
>>Give them a shout, they love this kind of stuff.
>>
>>
>
>Of course when you go to using the Sangoma cards with asterisk, it
>appears you lose any extra functionality Sangoma built into the card.
>That isn't a bad thing, but it negates any benefit of longevity.
>
>As for the original posters question. The TE cards from Digium can take
>care of your ISDN dial ups by itself. Asterisk can't take care of your
>analog dialups yet.
>
>The first thing to know is that you are not splitting the PRI, you are
>routing calls. Until you get the setup messages, you don't know what is
>what. Then when you get it, the call could be on any of the B channels.
>But once you get it, you can determine by the phone number that was
>dialed how to route the call. You can assign a DID for your dialups and
>route it all to your portmaster through a separate span or assign
>different numbers for ISDN and analog dialups so only the modem users go
>to the portmaster while your ISDN users are handled on the asterisk
>machine. All others are voice and dealt with from inside asterisk.
>
>
--
All the Best!
Sergey.
=========================
Sergey Kuznetsov
President/CEO
High Intellectual Technologies, Inc.
Web: http://www.hitcalls.com
E-mail: sergey.kuznetsov at highintellect.com
Business phone: (416) 548-9700
Mobile phone: (647) 287-8448
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