<br><br><div class="gmail_quote">On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton <span dir="ltr"><<a href="mailto:dfullertasterisk@shorelinecontainer.com">dfullertasterisk@shorelinecontainer.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">Tzafrir Cohen wrote:<br>
> On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:<br>
><br>
> You have a small typo:<br>
><br>
>> exten => _.,1,Dial(Zap,g1,${EXTEN})<br>
>> exten => _.,2,Dial(SIP,Provider,${EXTEN})<br>
><br>
> exten => _.,1,Dial(Zap/g1/${EXTEN})<br>
> exten => _.,2,Dial(SIP/Provider/${EXTEN})<br>
><br>
> ('/' instead of ',')<br>
><br>
<br>
</div>While this will work, be aware that there are circumstances where you<br>
may end up calling the number twice, once through each provider. One<br>
example is if the number you dial is busy, that progress will be passed<br>
via the PRI to asterisk and the dialplan will continue to the next<br>
priority. In this case, dialing the number again through the SIP<br>
provider. To avoid this you will need to use some dialplan logic and<br>
check the result of the DIALSTATUS variable. See this page for examples:<br>
<br>
<a href="http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS</a><br>
<font color="#888888"><br>
-Dave<br>
</font><div><div></div><div class="h5"><br></div></div></blockquote><div><br>Good point.<br><br>I was unaware that "busy" back from a TDM circuit would progress in the dialplan rather than going to the h exten.<br>
<br>What other cases are there like that? <br></div></div><br>-- <br>Thanks,<br>Steve Totaro <br>+18887771888 (Toll Free)<br>+12409381212 (Cell)<br>+12024369784 (Skype)<br>