<div dir="ltr">I had in a same situation and solved by Background 1 sec. silence.</div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Nov 25, 2015 at 5:45 PM, Brian :: <span dir="ltr"><<a href="mailto:bc@iptel.co" target="_blank">bc@iptel.co</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">add a pause in the dialplan for a second then proceed..<div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield <span dir="ltr"><<a href="mailto:tony@softins.co.uk" target="_blank">tony@softins.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">In article <<a href="mailto:20151125133008.6369360.14455.17239@gmail.com" target="_blank">20151125133008.6369360.14455.17239@gmail.com</a>>,<br>
<span>Israel Gottlieb <<a href="mailto:isrlgb@gmail.com" target="_blank">isrlgb@gmail.com</a>> wrote:<br>
> Try putting progress instead of answer<br>
<br>
</span>Yes, I tried Progress already, and it didn't help. But thanks for<br>
the suggestion!<br>
<span><font color="#888888"><br>
Tony<br>
</font></span><div><div><br>
> I have a puzzling situation, and would be grateful for any insight.<br>
><br>
> I have a dialplan that forwards an incoming call out to another<br>
> number via the same SIP trunk as it came in on. e.g.<br>
><br>
> [from-siptrunk]<br>
> exten => 0123456789,1,NoOp<br>
> exten => 0123456789,n,Dial(SIP/siptrunk/0987654321)<br>
><br>
> Now, if I use a different SIP trunk for the outbound call, than the<br>
> inbound call came on, the call is set up fine - the Answer signal from the<br>
> called party gets propagated back to the caller, and they can hear each<br>
> other.<br>
><br>
> But if the outbound SIP trunk is the same as the one the call came in on,<br>
> the caller doesn't hear any progress, and has no notification of when the<br>
> call was answered. Neither can the parties hear each other.<br>
><br>
> I have tried this on two different machines using two different SIP<br>
> providers.<br>
><br>
> However, if I change the above NoOp to be Answer(100), i.e. answer the<br>
> inbound call before placing the outbound Dial, the caller hears progress<br>
> and when the called party answers, they hear each other fine.<br>
><br>
> Of course, if the called party is busy, the caller just hears in-band<br>
> busy tone, as the caller's inbound call was already answered.<br>
><br>
> Can anyone explain why I need the Answer? It feels wrong that I should.<br>
><br>
> The siptrunk entry contains canreinvite=no and directmedia=no.<br>
><br>
> The version of Asterisk on these boxes is 10.5.1, if that's relevant.<br>
><br>
> Thanks for any insight!<br>
><br>
> Cheers<br>
> Tony<br>
><br>
> --<br>
> Tony Mountifield<br>
> Work: <a href="mailto:tony@softins.co.uk" target="_blank">tony@softins.co.uk</a> - <a href="http://www.softins.co.uk" rel="noreferrer" target="_blank">http://www.softins.co.uk</a><br>
> Play: <a href="mailto:tony@mountifield.org" target="_blank">tony@mountifield.org</a> - <a href="http://tony.mountifield.org" rel="noreferrer" target="_blank">http://tony.mountifield.org</a><br>
><br>
> --<br>
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<br><span class="HOEnZb"><font color="#888888">
<br>
--<br>
Tony Mountifield<br>
Work: <a href="mailto:tony@softins.co.uk" target="_blank">tony@softins.co.uk</a> - <a href="http://www.softins.co.uk" rel="noreferrer" target="_blank">http://www.softins.co.uk</a><br>
Play: <a href="mailto:tony@mountifield.org" target="_blank">tony@mountifield.org</a> - <a href="http://tony.mountifield.org" rel="noreferrer" target="_blank">http://tony.mountifield.org</a><br>
<br>
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</font></span></div></div></blockquote></div><br></div>
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