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

[Tamer Higazi]

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.


Tamer


Steve Edwards schrieb:
> On Sat, 18 Apr 2009, Tamer Higazi wrote:
>
>   
>>> On Fri, 17 Apr 2009, Steve Edwards wrote:
>>>
>>> Impatient are we :)
>>>       
>
>   
>> Yes you are entirely right! :)
>>     
>
> "Patience is a virtue."
>
>   
>>> If so, does it have 2 "rj-11's" or 1? Since the other handset says 
>>> "busy" this implies it is a single line DECT base.
>>>
>>>       
>> it has only 1 rj-11 port.
>>
>> ... with the DECT specification of ETSI I can send System messages where 
>> I can route the call from the base station to the handset direclty.
>>     
>
> Isn't DECT (Digital Enhanced Cordless Telecommunications) the protocol 
> between the handset and the base? From the base on, isn't it just POTS?
>
>   
>> with system messages you can receive device through outgoing calling 
>> number and assign through ISDN a DDI or MSN number, like those 
>> commercial PBX. The messages can be send through a small 2 way AGI 
>> script.
>>     
>
> The base has an rj-11 but is an ISDN device? I thought ISDN was rj-4[58]. 
> If your base is ISDN shouldn't you be connecting it to an ISDN terminal 
> adapter instead of an ATA?
>
> What the heck is a 2 way AGI script? To whom and from whom?
>
>   
>> Now, if you guys tell me that SIP isn't able to make it, to receive more 
>> then one calls at the same time, as well as placing, even if the base 
>> station has only 1 RJ11 port, then cordless phone systems aren't 
>> suitable at all for the asterisk PBX, which is a very sad issue.
>>     
>
> I think I'm confused. If the base has a single 2-wire rj-11, it's a single 
> line POTS device regardless of how many handsets you have or the 
> capabilities of your ATA. The fact that you say the "not in use" handset 
> says "busy" supports my assumptions.
>
>   
>> I hope still, that SIP can manage more then one call through one 
>> account, if not, it is a sad thing and I have to drop the development 
>> for DECT for the PBX system at all.
>>     
>
> SIP can manage multiple simultaneous calls through one account to multiple 
> SIP end points. But that's not what you have. As far as I can see, 
> Asterisk is going to see multiple handsets talking (DECTing?) to a single 
> POTS based base connected to an ATA as a single end point.
>
> Thanks in advance,
> ------------------------------------------------------------------------
> Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                             Fax: +1-760-731-3000
>
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