[asterisk-users] Converting GSM calls to SIP
Jeff Brower
jbrower at signalogic.com
Wed Apr 14 11:52:19 CDT 2010
Tonty-
> This is more or less the idea. I was not thinking about the E3 then break
> it down, because I am not sure they provide E3s, they suggest me invest into
> multiple E1 cards to support as many call as I can
Ok but how do you get the data? 35 E1s is a lot of cabling for an external connection.
-Jeff
> On Wed, Apr 14, 2010 at 11:56 AM, Jeff Brower <jbrower at signalogic.com>wrote:
>
>> Tonty-
>>
>> > This is a solution they proposed, using GSM gateways, but it wont let me
>> > handle 1000 simultaneous calls, the other option was using an E1 but the
>> > cost would be too much to deploy 35 E1s to support that many calls.
>> There
>> > might be a better way of doing it.
>>
>> Can you explain the "multiple E1" approach? Are you saying you would
>> connect to your GSM provider using an E3 line
>> and then break that out into multiple E1s that can be used with
>> Asterisk-compatible PCI/PCIe cards?
>>
>> If that's not accurate, please clarify.
>>
>> -Jeff
>>
>> > On Wed, Apr 14, 2010 at 11:08 AM, William Stillwell (Lists) <
>> > william.stillwell-lists at ablebody.net> wrote:
>> >
>> >>
>> http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Pascal Bruno
>> >> *Sent:* Wednesday, April 14, 2010 10:52 AM
>> >> *To:* asterisk-users at lists.digium.com
>> >> *Subject:* [asterisk-users] Converting GSM calls to SIP
>> >>
>> >>
>> >>
>> >> I have asked a GSM operator in my country if he can route a number or a
>> >> short code to my asterisk server via SIP (since they dont give DIDs in
>> my
>> >> country) the operator said they do not support SIP, they have no way of
>> >> converting GSM calls to SIP to then send them to me. I would like to
>> know
>> >> what is needed from the operator side to do this, what kind of material
>> is
>> >> needed, or what can be done from their side to send SIP calls to my
>> server.
>> >>
>> >> Thank you
>>
>>
>
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