<br><br><div class="gmail_quote">On Thu, Apr 23, 2009 at 6:11 PM, Matt Riddell <span dir="ltr"><<a href="mailto:lists@venturevoip.com">lists@venturevoip.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">On 24/04/2009 2:24 a.m., Nhadie wrote:<br>
> Hi,<br>
><br>
> i'm currently using Originate command on AMI, i can call a certain<br>
> channel like a SIP user SIP/1000 then once 1000 is answered it dials out<br>
> to amobile or landline.<br>
><br>
> Would just like to know if i can use AMI to dialout to a mobile or<br>
> landline first (instead of SIP user) and once answered, dial another<br>
> mobile or landline again.<br>
><br>
> If not is it possible to call a macro from the AMI? i think i can<br>
> probably use AGI for this, but i don't know if i can call a macro from<br>
> the AMI command.<br>
<br>
</div>Asterisk will always dial the channel first then the<br>
context/extension/application etc.<br>
<br>
Couple of things to bear in mind - you won't be able to tell if the call<br>
is answered if you are using analogue lines.<br>
<br>
Action: Originate<br>
Channel: Zap/g1/12345<br>
Context: extensions<br>
Extension: 1000<br>
Priority: 1<br>
<br>
The above will call 12345 and when connected (either when the call<br>
starts with analogue or when it is connected with digital) it will go to<br>
extension 1000 in the context extensions, where you would have something<br>
like:<br>
<br>
[extensions]<br>
exten => _1XXX,1,Dial(SIP/${EXTEN})<br>
<br>
--<br>
Kind Regards,<br>
<br>
Matt Riddell<br>
Director<br>
_______________________________________________<br>
<br>
<a href="http://www.venturevoip.com" target="_blank">http://www.venturevoip.com</a> (Great new VoIP end to end solution)<br>
<a href="http://www.venturevoip.com/news.php" target="_blank">http://www.venturevoip.com/news.php</a> (Daily Asterisk News - html)<br>
<a href="http://www.venturevoip.com/newrssfeed.php" target="_blank">http://www.venturevoip.com/newrssfeed.php</a> (Daily Asterisk News - rss)<br>
<div><div></div><div class="h5"><br></div></div></blockquote><div> </div></div><br>A much more scalable way to do this is to create and then FTP or move .call files to the proper directory. Depends how much you plan on banging on the AMI.<br clear="all">
<br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br><br>