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oliver,<br>
ugh, it is too obvious... why did it take me so long to figure it
out...<br>
<br>
both phones have to have to negotiate the same codec for audio... as
far as I know, * is supposed to do automatic translation and your
gateway should be doing translations only on the below codecs. I
haven't had that experience yet... <br>
<br>
one phone may be connected to your * box, but your other phone is
*not* connected to *. it is connected to a voip provider... since
they don't do any translation other than below. the * connection to
webcalldirect must have one of these codecs in the sip.conf for that
extension, the extension where webcalldirect is coming in, that is...<br>
<br>
phoneX -----> * ---------> webcalldirect -----> phoneY<br>
which one is phone1 and which is phone2?<br>
<br>
phoneX<---- * <---------- webcalldirect <-------phoneY<br>
-----------------| -------------------------| -----------------<br>
local LAN Internet local LAN<br>
some code no codec control no codec
control<br>
control little or no call quality control<br>
<br>
the phone connected to * will also select a code that matches up with
the caller (webcalldirect)... you have no advantage whether or not *
converts the audio to the phone connected to *. you won't get any
better reception from webcalldirect because you are not changing that
connection.<br>
<br>
also, I would change iLBC to ilbc, case may make a difference...
don't know for sure... perhaps someone else does...<br>
hope that is clearer...<br>
daveC<br>
<br>
<br>
<table class="sip" border="0" cellpadding="0" cellspacing="0"
width="100%">
<tbody>
<tr>
<td colspan="2"
style="border-bottom: 1px solid rgb(123, 123, 177); width: 100%;">
<p style="font-size: 12px;" id="inst"> <b>Codecs</b> </p>
</td>
</tr>
<tr>
<td
style="background: rgb(223, 238, 255) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; padding-left: 4px; height: 10px; vertical-align: top; padding-top: 3px; margin-top: 3px;"><img
src="cid:part1.06030801.02010906@iacnet.net" border="0"></td>
<td
style="background: rgb(223, 238, 255) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; padding-left: 0px;">G.711
(64 kbps) </td>
</tr>
<tr>
<td
style="padding-left: 4px; height: 10px; vertical-align: top; padding-top: 3px; margin-top: 3px;"><img
src="cid:part2.07070401.04000002@iacnet.net" border="0"></td>
<td style="padding-left: 0px;">G.726 (32 kbps)</td>
</tr>
<tr>
<td
style="background: rgb(223, 238, 255) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; padding-left: 4px; height: 10px; vertical-align: top; padding-top: 3px; margin-top: 3px;"><img
src="cid:part1.06030801.02010906@iacnet.net" border="0"></td>
<td
style="background: rgb(223, 238, 255) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; padding-left: 0px;">G.729
(8 kbps)</td>
</tr>
<tr>
<td
style="padding-left: 4px; height: 10px; vertical-align: top; padding-top: 3px; margin-top: 3px;"><img
src="cid:part2.07070401.04000002@iacnet.net" border="0"></td>
<td style="padding-left: 0px;">G.723 (5.3 & 6.3 kbps)</td>
</tr>
<tr>
<td
style="background: rgb(223, 238, 255) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; padding-left: 4px; height: 10px; vertical-align: top; padding-top: 3px; margin-top: 3px;"><img
src="cid:part1.06030801.02010906@iacnet.net" border="0"></td>
<td
style="background: rgb(223, 238, 255) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; padding-left: 0px;">GSMFR
(13.2 kbps) <b>Temporarily unavailable due to technical difficulties.</b></td>
</tr>
</tbody>
</table>
<br>
<br>
Oliver Brandt wrote:
<blockquote cite="mid20070428142435.GB7638@localhost.localdomain"
type="cite">
<pre wrap="">Hi Dave!
Thank you very much for replying!
</pre>
<blockquote type="cite">
<pre wrap="">what gateway provider are you referring to? doesn't your sip phone
</pre>
</blockquote>
<pre wrap=""><!---->
webcalldirect (it does not seam to support iLBC directly)
</pre>
<blockquote type="cite">
<pre wrap="">connect directly to * as your diagram indicated?
</pre>
</blockquote>
<pre wrap=""><!---->
Yes, my sipphone ist connected directly to * and also the gateway
provider is directly connected to *. My * is on a root server at hosting
provider (high bandwith internet connection to the gateway provider) but
my phone is connected through DSL with a very limited upstream. For this
reason I'd like asterisk to do the codec conversion from iLBC to ulaw.
I bett all I have to do is load the codec or/and the codec translator
for iLBC to ulaw. But when googleing I only find articles the describe,
that * is doing the codec translation automatically. I can't find any
information on how to load a codec or the translator manually. I'm
probably just using the wrong search string in google...
When * starts translators are beeing loaded, but as far as I can see non
for iLBC to ulaw.
I've put together another test setup with to sip phones to clarify the
problem:
[phone1]
disallow=all
allow=iLBC
[phone2]
disallow=all
allow=ulaw
When calling from one phone to the other I get the following message:
chan_sip.c:2841 sip_call: No audio format found to offer. Cancelling
call to phone2
Thank you very much again!
Oliver
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