<div>What does your dial command look like? </div>
<div> </div>
<div>If had DID working before, all you have to do is send the same information on to your pbx. For example, if the telco was sending the full number dialed on to your pbx, then you need to add to you dial command. If the telco was only sending, say, the last three digits, then that is all you send to your pbx.
</div>
<div> </div>
<div>Basically, you need to know what the telco was sending your old pbx, and then make sure you are sending that to *.</div>
<div> </div>
<div>For example, for mine, this is what i have in me extensions.conf file:</div>
<div> </div>
<div>exten => 2816040532,1,Dial(Zap/g3/210)</div>
<div> </div>
<div>This will take a call from my main number and send it to my auto attendant. My pbx picks up the 210 as being the DID number for my automated attendant. In your case, since you already had DID functioning, you would replace the 210 with whatever your telco would have sent. In other words, you need to pass on whatever the telco sends to you. You need to pass that on to your pbx. If before, your fax number was, say, 281-456-7890, and you receive all ten digits from the telco, then your dial command would like this:
</div>
<div> </div>
<div>exten => 2814567890,1,DIal(Zap/g3/2814567890), assuming g3 is the group assigned to your PRI, this may be different in your case.</div>
<div> </div>
<div>Does this make sense? You can use variables to expand on this so that you don't have to have a statement for each DID, but this should at least get you started.<br><br> </div>
<div><span class="gmail_quote">On 4/30/06, <b class="gmail_sendername">Remco Barende</b> <<a href="mailto:asterisk@barendse.to">asterisk@barendse.to</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Yes indeed I suspect that * is not passing any DID information on the<br>call. This could be because my Dial command is wrong or I may need to use
<br>different signalling settings. (Is there any other setting with pri_net?)<br><br>When doing pri debug I noticed a line that * was thinking that the other<br>side was not ISDN equipment (but there is a lot of output and I never read
<br>debug output before)<br><br>DID was working, the PRI was passing it on to the Alcatel.<br><br>I'll have a look on the forum you mentioned.<br><br>Thanks!<br><br><br>On Sun, 30 Apr 2006, Lacy Moore - Aspendora wrote:<br>
<br>> Hmm... In my case, it could be just dumb luck. I found some instructions<br>> on setting up DID on my pbx, and started that. Part way through, I wasn't<br>> sure what the rest of the instructions were talking about and felt I was
<br>> getting in too deep. So, I decided to see what would happen if I just tried<br>> it. It worked. Since this is only a temporary solution until we move<br>> completely off the pbx to Asterisk, I felt like I didn't need to find out
<br>> why it worked, just be thankful that it worked.<br>><br>> It sounds like your system is just answering the line and not paying<br>> attention to any DID information. I was able to find a lot of information
<br>> on the <a href="http://tek-tips.com">tek-tips.com</a> forums for the Merlin Legend. You may try a search on<br>> there and see.<br>><br>> Was DID working in the past, or have you just added it with the addition of
<br>> the Asterisk system?<br>><br>><br>> On 4/30/06, Remco Barende <<a href="mailto:asterisk@barendse.to">asterisk@barendse.to</a>> wrote:<br>>><br>>><br>>> Thanks! The PBX is a Alcatel Novo Supreme. All calls go straight into the
<br>>> auto attendant no matter whoch extension I dial on the Zap group the PBX<br>>> is connected to. I tried dialling in by hand using several combinations<br>>> but I always get the auto attendant.<br>
>><br>>> How do you transfer the call straight to the extension the fax is on? I<br>>> guess using a Dial command fro *?<br>>><br>>> I suspect that the PBX is missing some signalling.<br>>>
<br>>> Thanks!<br>>><br>>> On Sun, 30 Apr 2006, Lacy Moore - Aspendora wrote:<br>>><br>>> > Also, what is the legacy PBX? On the Merlin Legend, for instance, there<br>>> are<br>>> > special Class of Services that can be setup to go straight to the auto
<br>>> > attendant. I'm not sure if that's what you need or not. The other<br>>> question<br>>> > is, why can't you transfer the call straight to the extension the fax is<br>>> > on? Again, on the Legend, it defaults (I guess) to receving the
<br>>> extension<br>>> > number. For example, if my fax machine is located on ext. 170, then I<br>>> just<br>>> > dial 170 from Asterisk on the PRI that is connected to the Legend. I<br>>> didn't
<br>>> > have to do half of what the manual says you have to do, because Asterisk<br>>> > takes care of all the translations.<br>>> ><br>>> > On 4/30/06, Jerry Jones <<a href="mailto:jjones@danrj.com">
jjones@danrj.com</a>> wrote:<br>>> >><br>>> >> You do not say how you have the two connected/<br>>> >><br>>> >> Are you connecting the * to stations via fxo or to lines via fxs on
<br>>> >> the legacy?<br>>> >><br>>> >><br>>> >> On Apr 30, 2006, at 11:22 AM, Remco Barende wrote:<br>>> >><br>>> >> > Hi list!<br>>> >> >
<br>>> >> > I managed to come reasonably far (farther than I thought I would)<br>>> >> > but have two problems.<br>>> >> ><br>>> >> > I still need to pass calls to the Legacy PBX for Fax (I need it as
<br>>> >> > a channel bank).<br>>> >> ><br>>> >> > I have calls coming in into asterisk, that works fine. Based on the<br>>> >> > DID I can route calls to the Legacy PBX but I'm puzzled how.
<br>>> >> ><br>>> >> > I guess I need a new dial command for that? All fax calls are now<br>>> >> > coming in a new context which I called topbx. If I issue a dial<br>>> >> > command there the legacy PBX treats it as a local extension call
<br>>> >> > and not a call from the outside.<br>>> >> ><br>>> >> > Which dial command do I need to use to make the old PBX believe the<br>>> >> > call came from outside?
<br>>> >> ><br>>> >> > (All the pages I found on this subject mention something about<br>>> >> > retaining caller ID which is nice but now I need to retain DID info<br>>> >> > on the call I guess?)
<br>>> >> ><br>>> >> > Thanks for any help!<br>>> >> ><br>>> >> ><br>>> >> > _______________________________________________<br>>> >> > --Bandwidth and Colocation provided by
<a href="http://Easynews.com">Easynews.com</a> --<br>>> >> ><br>>> >> > Asterisk-Users mailing list<br>>> >> > To UNSUBSCRIBE or update options visit:<br>>> >> >
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>>> >><br>>> >> _______________________________________________<br>>> >> --Bandwidth and Colocation provided by
<a href="http://Easynews.com">Easynews.com</a> --<br>>> >><br>>> >> Asterisk-Users mailing list<br>>> >> To UNSUBSCRIBE or update options visit:<br>>> >> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">
http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>>> >><br>>> ><br>>> ><br>>> ><br>>> > --<br>>> > Lacy Moore<br>>> > Aspendora, Inc.<br>>> >
<br>>> _______________________________________________<br>>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>>><br>>> Asterisk-Users mailing list<br>>> To UNSUBSCRIBE or update options visit:
<br>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>>><br>><br>><br>><br>> --<br>> Lacy Moore<br>> Aspendora, Inc.
<br>><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:
<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>-- <br>Lacy Moore<br>Aspendora, Inc.