Matt,<br><br>Asterisk version == 1.4.22<br>dtmfmode == info<br>calls are bridged through Asterisk (canreinvite=no)<br><br>Jonathan<br><br><div class="gmail_quote">On Sun, Dec 28, 2008 at 3:23 PM, Matt Florell <span dir="ltr"><<a href="mailto:astmattf@gmail.com">astmattf@gmail.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><div></div><div class="Wj3C7c">On 12/28/08, jonathan augenstine <<a href="mailto:jaugenstine@gmail.com">jaugenstine@gmail.com</a>> wrote:<br>
> I am trying to resolve an issue and I believe it is my configuration. The<br>
> scenario is that I have a SIP detected on the server. The dial plan then<br>
> makes a local connection to another part of the dial plan. The new dial<br>
> plan extension then places another SIP call out to a SIP phone. When the<br>
> call is accepted there is streamed from the calling SIP phone. When the<br>
> audio is complete a DTMF is transmitted to Asterisk. The DTMF is detected<br>
> by Asterisk but it does not get passed through to the other SIP phone. I<br>
> would like the DTMF to pass-through to the other SIP phone. Is this a<br>
> configuration issue? Or do I need to handle this on the dial plan level?<br>
><br>
> Jonathan<br>
<br>
</div></div>Asterisk version?<br>
<br>
What are both dtmfmodes set to for each SIP endpoint?<br>
<br>
Are the calls natively bridged or bridged through Asterisk?<br>
<br>
MATT---<br>
<br>
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