Thanks Tony for you reply.<div><br></div><div>Do you have any idea why Asterisk require "t" in Dial command?</div><div><br></div><div>Cheers,</div><div><br></div><div>Moe<br><br><div class="gmail_quote">On Sun, May 18, 2008 at 1:14 AM, Tony Mountifield <<a href="mailto:tony@softins.clara.co.uk">tony@softins.clara.co.uk</a>> 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:28749f210805170447w7e2da378vb11d12bdf8dd4b81@mail.gmail.com">28749f210805170447w7e2da378vb11d12bdf8dd4b81@mail.gmail.com</a>>,<br>
<div class="Ih2E3d">Mohammad A. Navid <<a href="mailto:manavid@gmail.com">manavid@gmail.com</a>> wrote:<br>
><br>
> I'm implementing a simple calling card feature for testing purpose. I have a<br>
> DID number, when I called my DID number and enter the phone number to call,<br>
> Asterisk would dial the number for me but the sound was only one way.<br>
> After hours of struggling with the problem, I found out that I need to add<br>
> "t" to my dial options, this is the correct way of dialing out:<br>
><br>
> -> Dial(SIP/carrier/3105555555|20|t)<br>
><br>
> Now I need to know what was going on? Why with option "t" both parties can<br>
> hear each other, but without option "t" in dial cmd only one party could<br>
> hear?<br>
><br>
> Another interesting issue is, if I use Answer() command at the begining the<br>
> sound becomes one way even if I use "t" in options.<br>
<br>
</div>Try adding "reinvite=no" to the sip.conf or users.conf definition for your<br>
SIP service provider.<br>
<br>
Cheers<br>
Tony<br>
--<br>
Tony Mountifield<br>
Work: <a href="mailto:tony@softins.co.uk">tony@softins.co.uk</a> - <a href="http://www.softins.co.uk" target="_blank">http://www.softins.co.uk</a><br>
Play: <a href="mailto:tony@mountifield.org">tony@mountifield.org</a> - <a href="http://tony.mountifield.org" target="_blank">http://tony.mountifield.org</a><br>
<br>
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