<div>From what I have seen if your sip provider does not take g722 then you will have problems with outgoing calls. When I tried this, the same way you did, I could make calles externally but had no audio each way reguardless of what I tried to pass to the sip provider. Best bet is to use what your sip provider can use or find another provider that that can do g722. That's what I did when I wanted to use g726.</div>
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<div>my2cents<br><br></div>
<div class="gmail_quote">On Tue, Jun 29, 2010 at 2:42 PM, Mindaugas Kezys <span dir="ltr"><<a href="mailto:mkezys@gmail.com">mkezys@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Try this: <a href="http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch" target="_blank">http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch</a><br>
<br>Regards,<br>Mindaugas Kezys<br><br>Kolmisoft UAB<br>VoIP Billing Solutions<br>e-mail: <a href="mailto:info@kolmisoft.com">info@kolmisoft.com</a><br>URL: <a href="http://www.kolmisoft.com/" target="_blank">http://www.kolmisoft.com</a><br>
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<div class="h5"><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 Steve Davies<br>
Sent: Tuesday, June 29, 2010 7:51 PM<br>To: Asterisk Users Mailing List - Non-Commercial Discussion<br>Subject: Re: [asterisk-users] Codec negotiation<br><br>On 26 June 2010 22:08, Ryan Wagoner <<a href="mailto:rswagoner@gmail.com">rswagoner@gmail.com</a>> wrote:<br>
> I have Polycom phones that support the g722 codec. Adding allow=g722<br>> to the [general] section of sip.conf works great and I can make calls<br>> between the phones using g722. However Asterisk is negotiating g722<br>
> for calls going out my voip provider and transcoding these to ulaw. In<br>> sip.conf for the provider I have deny=all and allow=ulaw. This can<br>> cause potential audio degrading and wastes cpu cycles. If Asterisk<br>
> knows the trunk only supports ulaw why would it offer g722 to the<br>> phone.<br>><br>> Ryan<br><br>Because the codec is already chosen before the call is made, and you<br>told it that g722 is permitted.<br>
<br>There are all sorts of discussions in play about codec negotiation,<br>but at this point in time, if you want different behaviour you'll need<br>to go and code it yourself, and cross-channeltype this is not going to<br>
be trivial :)<br><br>Cheers,<br>Steve<br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
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