<br><br><div class="gmail_quote">2008/12/17 Artifex Maximus <span dir="ltr"><<a href="mailto:artifexor@gmail.com">artifexor@gmail.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;">
On Wed, Dec 17, 2008 at 11:52 AM, Olivier <<a href="mailto:oza-4h07@myamail.com">oza-4h07@myamail.com</a>> wrote:<br>
> 2008/12/17 Artifex Maximus <<a href="mailto:artifexor@gmail.com">artifexor@gmail.com</a>><br>
>> Is anyone using the $subject setup?<br>
>><br>
>> What I would like to do the following setup:<br>
>> 1. OXE is setup for receiving calls, handling Agents<br>
>> 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931)<br>
>> PRI<br>
>><br>
>> I've talked with support person at Alcatel and he said that Q.931<br>
>> cannot handle this situation because after calls "leave" OXE it does<br>
>> not know anything so I cannot hangup in Asterisk and call will use two<br>
>> channel. Is it right? He said that ABCF2 or Q.SIG is able handling<br>
>> this situation because Q.SIG is an extension to Q.931. I take some<br>
>> search on topic and find out that Asterisk's Q.SIG not fully<br>
>> implemented. Is Asterisk implementation enough for this kind of setup?<br>
> What is needed is that the Asterisk box should either :<br>
> - forward incoming call to the right endpoint, using a single channel,<br>
> - open a second channel and remain in media path till it ends.<br>
Thanks for your answer! You are right and first option what I am<br>
looking for. I have asked support staff and sending back DTMF on open<br>
channel does not help.</blockquote><div><br>True ! <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>
> I'm not an authority on this topic, but I would say that, as OXE and<br>
> asterisk are connected through an E1/T1 link,<br>
> - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and<br>
> check asterisk's QSIG supports Call Deflection),<br>
> - casual PRI is enough if you stick with 2 channels option.<br>
Unfortunately I am not expert on this topic as well but second option<br>
is not good for us. The question is how good Asterisk's Q.SIG<br>
implementation for this task.</blockquote><div><br>That's the question !<br>Maybe someone else could help on this as I don't have much experience to share.<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>
<br>
> If you don't expect to get more than 15 (or 12) calls at a time, I don't see<br>
> any real downside to use option 2.<br>
Often we have more than 15 calls at same time and that is why first<br>
option is not acceptable.</blockquote><div>you mean "second option is not acceptable", don't you ? <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>
Bye,<br>
Zsolt<br>
<br>
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