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jonathan blais wrote:
<blockquote cite="mid9718b4f90601310740v1ed9002cn@mail.gmail.com"
type="cite">I'm using Linphone. I tested with Asterisk and Speex only,
I created a
channel with echo and it worked. It seems to have problem when using
app_conference.<br>
</blockquote>
<br>
If you just use app_echo, then asterisk won't be trying to decode your
frames; it will just be sending them back to you. Therefore, if your
client is using an incompatible packing of the frames, it won't matter,
as long as your client can also understand it.<br>
<br>
In order to determine if your client is using a compatible packing,
you'll need to have asterisk do something which requires asterisk to
decode the frames: Like bridge the call to a channel using another
codec, or record the call to a non-speex format (i.e. write to a PCM
WAV file), etc.<br>
<br>
-SteveK<br>
<br>
<blockquote cite="mid9718b4f90601310740v1ed9002cn@mail.gmail.com"
type="cite"><br>
Jonathan<br>
<br>
<div><span class="gmail_quote">2006/1/31, Steve Kann <<a
href="mailto:stevek@stevek.com">stevek@stevek.com</a>>:</span>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">jonathan
blais wrote:<br>
<br>
> Hi,<br>
><br>
> Does anyone ever used Speex with app_conference in Asterisk ? I'm<br>
> having a hard time to figure why I always get this error "warning:<br>
> Invalid mode encountered: corrupted stream?".
<br>
<br>
Yes, we do this all the time. Where are your speex frames coming from?<br>
There's some SIP phones out there which seem to use an incompatible<br>
speex encoding, I seem to recall.<br>
<br>
<br>
-SteveK<br>
<br>
<br>
<br>
</blockquote>
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<br>
</blockquote>
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