Try ilbc if the phone supports (free) or g729 ( better but your asterisk will need license if you want to transcode calls from g729 to other codecs or want to record calls ) . Also check your phones config if its support multiple codecs . .
<br><br><div><span class="gmail_quote">On 02/05/07, <b class="gmail_sendername">Rob Schall</b> <<a href="mailto:rschall@callone.net">rschall@callone.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000">
So I reloaded things and had just gsm set for 2 of my polycom 501
phones. However, the logs say "No codec found", which leads me to
believe that polycom 501 phones can't use gsm. Does anyone have
something like this working? If not gsm, is there a better option with
these phones over a low bandwidth situation?<br>
<br>
Rob<br>
<br>
Ed Nuņez wrote:
<blockquote cite="http://mid00d801c78c21$be6b7030$3b425090$@com" type="cite"><div><span class="e" id="q_1124d561873da15c_1">
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Reload
will reload your sip.conf file! As well as iax.conf,
extensions.conf, queues.conf, voicemail.conf, users.conf</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
</div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<div>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt; color: windowtext;">From:</span></b><span style="font-size: 10pt; color: windowtext;">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>
[<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">mailto:asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Rob
Schall<br>
<b>Sent:</b> Tuesday, May 01, 2007 2:06 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Calls in ulaw, not gsm as desired</span></p>
</div>
</div>
<p> </p>
<p>I was in the asterisk console and I typed
"reload". Is this not enough to reload the sip.conf file?<br>
<br>
Rob<br>
<br>
Andreas Sikkema wrote: </p>
<blockquote style="margin-top: 5pt; margin-bottom: 5pt;">
<pre>However, even once I reloaded the extensions, its still only </pre>
<pre>using ulaw.</pre>
<pre> </pre>
</blockquote>
<pre> </pre>
<pre>You didn't reload the sip config? Maybe that's your problem?</pre>
<pre> </pre>
<pre> </pre>
<p> </p>
</div>
</span></div><pre><hr size="4" width="90%"><span class="q">
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</blockquote>
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