<font color="#3333ff"><font face="comic sans ms,sans-serif">Hi</font></font><div><font color="#3333ff"><font face="comic sans ms,sans-serif"><br></font></font></div><div><font color="#3333ff"><font face="comic sans ms,sans-serif">Thats is also one of the reason<br>
</font></font><br><div class="gmail_quote">On Tue, Nov 15, 2011 at 20:27, <span dir="ltr"><<a href="mailto:isrlgb@gmail.com">isrlgb@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
The variable for outbound is (SIP_CODEC_OUTBOUND=g722)<br>
<br>
But I think asterisk will try to transcode then because the preferred codec on the phone is ulaw or so<br>
<br>
-----Original Message-----<br>
From: "Danny Nicholas" <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>><br>
Sender: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
Date: Tue, 15 Nov 2011 08:50:37<br>
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Subject: Re: [asterisk-users] Forcing a CODEC<br>
<div class="HOEnZb"><div class="h5"><br>
That's one of the uses of the SIP_CODEC dialplan variable. Just set it in<br>
the context or the sip.conf or users.conf. In your particular case, just<br>
set up a specific context for the IAX calls<br>
[iax-in]<br>
Exten => _X.,1,Set(SIP_CODEC=G722)<br>
Exten => _X.,n,answer()<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Jaap Winius<br>
Sent: Tuesday, November 15, 2011 8:47 AM<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: [asterisk-users] Forcing a CODEC<br>
<br>
Hi folks,<br>
<br>
How can I take advantage of a high-bandwidth CODEC, like G.722, for internal<br>
communications at my site, but use G.711 (alaw/ulaw) for all other outgoing<br>
calls? I need G.711 to support Inband DTMF signaling.<br>
<br>
As my site has multiple locations that are tied together with IAX trunks, I<br>
was hoping that it would be possible to specify alaw and ulaw as the first<br>
two CODEC choices for the SIP phones, as well as in their sip.conf<br>
configurations, but that I could use the IAX trunks (with bandwidth=high) to<br>
force the phones to use their third CODEC choice, g722, because that would<br>
be the only CODEC specified for the IAX trunks (following disallow=all).<br>
<br>
Unfortunately, that doesn't work. Although the Asterisk console reports that<br>
g722 is being used, when I listen to the connection it's obvious that a<br>
G.711 CODEC is being used. Curiously, the reverse does<br>
work: if g722 is specified as the first CODEC of choice for the phones, it<br>
is possible to use the IAX trunks to force them to use alaw/ulaw instead.<br>
<br>
Is a solution to this problem?<br>
<br>
I'm using Debian squeeze with Asterisk 1.6.2.9.<br>
<br>
Cheers,<br>
<br>
Jaap<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div></div><div><br></div><div>Amit Anand</div><div><br></div><div><font size="1"><span style="font-family:arial,sans-serif;border-collapse:collapse"><h1 style="font-family:arial,sans-serif;margin:12px 5px 5px 10px;padding:0px;color:rgb(0, 0, 0);background:inherit;border-right:inherit">
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