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Hi Raul<br>
<br>
Raúl Gómez C. said the following on 05-Mar-08 07:40 PM:
<blockquote
cite="mid:684b0a740803050940h5cda9c67k1cecffab96528fe3@mail.gmail.com"
type="cite">Ian,<br>
<br>
I'm unable to transfer calls using *2, I'm not sure why. Here's my
configs:<br>
</blockquote>
<snip><br>
<blockquote
cite="mid:684b0a740803050940h5cda9c67k1cecffab96528fe3@mail.gmail.com"
type="cite"><span style="font-family: courier new,monospace;"><br>
<span style="font-family: arial,sans-serif;">In the phones the "</span></span><i
style="font-family: arial,sans-serif;">Send DTMF:"</i><span
style="font-family: arial,sans-serif;"> is set to "in-audio" and "via
SIP INFO"</span><br>
</blockquote>
It should only be set to SIP INFO, or else the audio comes out too
choppy (in my case though) for asterisk to recognize it.<br>
<br>
It took me about half a days experimenting with sip debug enabled on
the channel to find that out, btw if you take longer than the 500ms
feature timeout, asterisk still forwards the DTMF to the other channel.<br>
<br>
Regards<br>
Ian<br>
<blockquote
cite="mid:684b0a740803050940h5cda9c67k1cecffab96528fe3@mail.gmail.com"
type="cite"><span style="font-family: courier new,monospace;"></span><br>
What I'm missing here?<br>
-- <br>
Raul<br>
Linux Counter #156439<br>
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