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

Sat Apr 18 05:16:37 CDT 2009

D Tucny schrieb:
> 2009/4/18 Tamer Higazi <th982a at googlemail.com
> <mailto: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...
no I am not
>
> 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...
>
read the DECT specification from A-Z. In Germany we have digital (isdn
analog adapter that do it). By the way, in Germany and many other
European countries, BRI ISDN connections for household and companies are
widespread.

commercial PBX systems are BRI ISDN and analogue FXS related. Call comes
in, and call is going out.

If you read the DECT specification completly! you know how to set up
before you route the call the DECT / GAP system.

> 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...
>
not true, read the specification from A-Z, i't only about SIP, to
receive several calls at the same time, as well placing.

> 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
>

Tamer
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