[asterisk-users] opening 2 and more channels on 1 SIP account

[D Tucny]

2009/4/18 Tamer Higazi <th982a at googlemail.com>

> Scenario:
> I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP
> cpu to take out the echo cancellation.
>
> Communication is done through the chan_capi interface module.
>
> If a call comes inside, and I forward it to the SIP account that is
> registered in the module, then all DECT phone do ring. But DECT / GAP
> phones are not designed for these issues.
>
> Scenario what a commercial PBX system does which has a ISDN board.
>
> Set up the phones:
> 1 - queues through system messages the dect man station on which the
> cordless devices are registered to. the main station tells him the ID of
> the devices and I assign through the webinterface the numbers (DDI or
> MSN) to the devices.
>
> 2 - set up is done!
>
> Call routine:
>
>
> Call in!
> 1 - from the NT unit of my home line comes a call that goes to the PBX.
> 2 - The PBX which receives the call extract the number (DDI or MSN) and
> compare it in the list of which phone it is (from step one)!
> 3 - The PBX send a message queue to the base station to check if the
> phone is busy, if yes forget it. If no pass the call through. Done with
> sending a message to the base that the call is passed to this device,
> for that the other devices won't ring.
>
> Call out!
> 1 - from the handset I make a call
> 2- the PBX, sends a message to the base station asking who dialed the
> number.
> 3 - the base station gives back the id, the outgoing number is set for
> that the call is passed through with the desired outgoing number.
>
>
> Now Asterisk, if SIP supports it receiving and placing several calls
> through one FXS port:
>
> the agi script:
> http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText
>
> 1 - a call is placed
> 2 - the agi script sends a message through the sip channel and the anser
> comes back, the answer is held in a variable
> 3 - the variable had been worked out, and the MSN or DDI is set
>
> ---
>
> 1  a call is received through the chan_capi interface
> 2.  the dialplan knows which id belongs to the DDI or MSN number and
> calls the AGI script, which sends the message to the base station asking
> if the handset is available, busy or ready to receive calls.
> 3. the script returns a value that is being worked out and the agi
> script is called again to tell the base station that the incoming call
> is for the handset id (let us say number 5), that not all phones do ring.
> 4. the call is forwarded to the FXS port and that's it.
>
>
> This is how a usual PBX System in Germany and across europe do work. But
> if SIP or Asterisk do not support receiving and placing more calls
> through one FXS port and channel at the same time, then the DECT
> sollution can be dropped at all for me, and I shouldn't lose more time
> in this issue.
>
>
> DECT itself, is a well worked out technologie that gives you the chance
> to make a lot! It is programming work, not more then that.
>
> I hope all questions are being answered.
>
>
You are confused...

While DECT may well be capable of this sort of functionality and while
asterisk, and SIP are capable of this sort of functionality, you are using
an intermediate technology, a single POTS analogue connection, that isn't
capable...

You'll need a DECT base that either directly supports SIP for communicating
with Asterisk, or, with a more capable interface, such as ISDN, that allows
for more advanced communication and multiple channels...

The best you could probably hope to get using an FXS connection is that a
single inbound call could be routed to one of the handsets by using
distinctive ring, if the base supports it... However, you can not have more
than one call over one analogue FXS connection, this isn't an Asterisk or
SIP limitation, this is a limitation of the analogue connection...

Example SIP DECT devices:
http://www.snom.com/en/products/snom-m3-voip-phone/
http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-1814D885/03/hs.xsl/30395.htm
Multiple Siemens devices

d
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