Are your sip phones capable of auto-answer?<br><br>I can imagine you can terminate the incoming call into a meet-me conference (no pass code) and then trigger a script that creates a call file for each of the other participating phones. The auto-answer part seems like the sticky part.
<br><br><div><span class="gmail_quote">On 10/15/06, <b class="gmail_sendername">Marc Heckmann</b> <<a href="mailto:mh@nadir.org">mh@nadir.org</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
On Sun, 2006-15-10 at 05:09 -0400, Henry.L.Coleman wrote:<br>> The quirk of your old PBX is in fact exactly what happens when you put any<br>> two analog phones on the same line. The easiest way to duplicate this is
<br>> to connect another analog phone to your ATA. Some analog phones can<br>> indicate when the other is on the line and can put a call on hold locally.<br><br>In fact no, I should have explained better, but in the old system one
<br>phone was analogue and the other was a multi-line digital Nortel<br>Meridian phone. The one phone has to be analogue because it interfaces<br>with a radio broadcast phone patch.<br><br>-m<br><br>><br>> > Hi,<br>
> ><br>> > I am looking to replace a quirk of our old PBX system functionality with<br>> > asterisk but after searching, archives, wiki, etc.. I cannot figure out<br>> > how.<br>> ><br>> > Here is what I would like to do:
<br>> ><br>> > PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a<br>> > SIP ATA. When an incoming call comes in, I would like to ring both<br>> > phones, but if phoneA is answered first, I would like phoneB to be
<br>> > answered as well and left in a "off hook" state so that when someone<br>> > picks up the receiver of phoneB, they can hear and participate in the<br>> > conversation between the calling party and phoneA.
<br>> ><br>> > I believe I would have to put both phones in a MeetMe conference, but<br>> > how to I "auto-answer" phoneB when phoneA has answered the call?<br>> ><br>> > I suspect that this may not be possible with asterisk, but would like
<br>> > confirmation of that.<br>> ><br>> > Thanks in advance.<br>> ><br>> > -m<br>> ><br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by
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