<div class="gmail_quote">2009/4/18 Tamer Higazi <span dir="ltr"><<a href="mailto:th982a@googlemail.com">th982a@googlemail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
D Tucny schrieb:<br>
<div class="im">> 2009/4/18 Tamer Higazi <<a href="mailto:th982a@googlemail.com">th982a@googlemail.com</a><br>
</div>> <mailto:<a href="mailto:th982a@googlemail.com">th982a@googlemail.com</a>>><br>
<div><div></div><div class="h5">><br>
> Scenario:<br>
> I have a Asterisk PBX with a cologne chipset ISDN BRI card on it a DSP<br>
> cpu to take out the echo cancellation.<br>
><br>
> Communication is done through the chan_capi interface module.<br>
><br>
> If a call comes inside, and I forward it to the SIP account that is<br>
> registered in the module, then all DECT phone do ring. But DECT / GAP<br>
> phones are not designed for these issues.<br>
><br>
> Scenario what a commercial PBX system does which has a ISDN board.<br>
><br>
> Set up the phones:<br>
> 1 - queues through system messages the dect man station on which the<br>
> cordless devices are registered to. the main station tells him the<br>
> ID of<br>
> the devices and I assign through the webinterface the numbers (DDI or<br>
> MSN) to the devices.<br>
><br>
> 2 - set up is done!<br>
><br>
> Call routine:<br>
><br>
><br>
> Call in!<br>
> 1 - from the NT unit of my home line comes a call that goes to the<br>
> PBX.<br>
> 2 - The PBX which receives the call extract the number (DDI or<br>
> MSN) and<br>
> compare it in the list of which phone it is (from step one)!<br>
> 3 - The PBX send a message queue to the base station to check if the<br>
> phone is busy, if yes forget it. If no pass the call through. Done<br>
> with<br>
> sending a message to the base that the call is passed to this device,<br>
> for that the other devices won't ring.<br>
><br>
> Call out!<br>
> 1 - from the handset I make a call<br>
> 2- the PBX, sends a message to the base station asking who dialed the<br>
> number.<br>
> 3 - the base station gives back the id, the outgoing number is set for<br>
> that the call is passed through with the desired outgoing number.<br>
><br>
><br>
> Now Asterisk, if SIP supports it receiving and placing several calls<br>
> through one FXS port:<br>
><br>
> the agi script:<br>
> <a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText</a><br>
><br>
> 1 - a call is placed<br>
> 2 - the agi script sends a message through the sip channel and the<br>
> anser<br>
> comes back, the answer is held in a variable<br>
> 3 - the variable had been worked out, and the MSN or DDI is set<br>
><br>
> ---<br>
><br>
> 1 a call is received through the chan_capi interface<br>
> 2. the dialplan knows which id belongs to the DDI or MSN number and<br>
> calls the AGI script, which sends the message to the base station<br>
> asking<br>
> if the handset is available, busy or ready to receive calls.<br>
> 3. the script returns a value that is being worked out and the agi<br>
> script is called again to tell the base station that the incoming call<br>
> is for the handset id (let us say number 5), that not all phones<br>
> do ring.<br>
> 4. the call is forwarded to the FXS port and that's it.<br>
><br>
><br>
> This is how a usual PBX System in Germany and across europe do<br>
> work. But<br>
> if SIP or Asterisk do not support receiving and placing more calls<br>
> through one FXS port and channel at the same time, then the DECT<br>
> sollution can be dropped at all for me, and I shouldn't lose more time<br>
> in this issue.<br>
><br>
><br>
> DECT itself, is a well worked out technologie that gives you the<br>
> chance<br>
> to make a lot! It is programming work, not more then that.<br>
><br>
> I hope all questions are being answered.<br>
><br>
><br>
> You are confused...<br>
</div></div>no I am not</blockquote><div><br><br> </div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
<div class="im">><br>
> While DECT may well be capable of this sort of functionality and while<br>
> asterisk, and SIP are capable of this sort of functionality, you are<br>
> using an intermediate technology, a single POTS analogue connection,<br>
> that isn't capable...<br>
><br>
</div>read the DECT specification from A-Z. In Germany we have digital (isdn<br>
analog adapter that do it). By the way, in Germany and many other<br>
European countries, BRI ISDN connections for household and companies are<br>
widespread.<br>
</blockquote><div><br>ISDN is not analogue... you are confused... I know BRI ISDN connections are widespread, I've used them extensively before... Yes, a BRI ISDN connection can support two concurrent voice calls... Yes, a BRI ISDN connection can pass all sorts of additional signalling information... but... you are talking about a single analogue connection provided by an ATA device and a DECT base that only has a single rj-11 analogue connection...<br>
<br>Show me the section of the DECT specifications where they give details on how to run multiple analogue voice streams over a single standard analogue circuit without special hardware and I'll admit defeat right now and go to the pub to toast all the engineers involved in writing the specs as they will have acheived something miraculous... <br>
</div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
commercial PBX systems are BRI ISDN and analogue FXS related. Call comes<br>
in, and call is going out.</blockquote><div><br>And the same is true with asterisk, call comes in, processing happens, call goes out... that is not what is at issue here... it's the basic technological issue that the analogue interface will take a single call and you seem to be hoping to send multiple concurrent calls over it, and that problem exists with any pbx you use...<br>
</div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
<br>
If you read the DECT specification completly! you know how to set up<br>
before you route the call the DECT / GAP system.</blockquote><div><br>I've read the section 5 of the specs that you mentioned... It only talks about DECT communication, it's not at all relevant here because a PBX will not talk DECT unless it has DECT modules, from what you've told us you are using an ATA and a base with only an analogue line interface... <br>
</div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
<div class="im"><br>
> You'll need a DECT base that either directly supports SIP for<br>
> communicating with Asterisk, or, with a more capable interface, such<br>
> as ISDN, that allows for more advanced communication and multiple<br>
> channels...<br>
><br>
</div>not true, read the specification from A-Z, i't only about SIP, to<br>
receive several calls at the same time, as well placing.</blockquote><div><br>If I have a DECT base that has a single analogue line connection, I can have maximum 1 concurrent call, no matter what the specs of the base or the handsets.<br>
If I have a DECT base that has a single ISDN2/BRI digital line connection, I can have a maximum of 2 concurrent calls, no matter what the specs of the base or the handsets<br>If I have a DECT base that has a single ethernet connection and supports SIP, I can in theory have as many concurrent calls as the hardware can handle, even with a single SIP account.<br>
If I have a DECT base that has a proprietary digital connection to a proprietary commercial PBX, I can in theory have as many concurrent calls as the hardware can handle...<br><br>So, you have an Asterisk PBX set up, massive numbers of concurrent calls supported...<br>
You have an ISDN BRI line connection, 2 concurrent calls supported...<br>You are using SIP for client connections to your asterisk PBX, massive numbers of concurrent calls supported...<br>You are using DECT for last hop connection to your handsets, potential for quite a few concurrent calls supported...<br>
But, and this is a big but... you have a single analogue connection between the SIP parts and the DECT parts... So, whatever massive numbers of concurrent calls are supported with SIP or DECT, your single analogue connection is the limiting factor here... There is no SIP running over the analogue circuit, there is no DECT running over the analogue circuit, the best signalling you get is variable ring timing, i.e. distinctive ring, caller ID and progress tones... if you're really lucky, your ATA may support polarity reversal too... but... that's it... no digital signalling channel, this is standard analogue, no multiple analogue voice paths, this is standard analogue... no magic...<br>
</div></div>d<br>