<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt">Eric<br />
<br />
With 1.8.x I use. <span lang="EN">
<p>exten => Process,1,Set(SIP_CODEC=ulaw)<br />
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And the system kicks the call over to ulaw. Now this is just prior to the answer so I don't know if it meets your criteria. But it works great to enforce inline T.30 audio faxes. I also use the f/F option T.38 or T.30 on recevie fax. This option was added as part of a patch in 1.8 and is in the 1.10/2.0 branch. </p>
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<div id="divSignature">Thanks<br />
<br />
Bryant Zimmerman (ZK Tech Inc.)<br />
616-855-1030 Ext. 2003</div>
<br />
<br />
<br />
<div id="divSignature">Thanks<br />
<br />
Bryant Zimmerman (ZK Tech Inc.)<br />
616-855-1030 Ext. 2003</div>
<br />
<br />
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<b>From</b>: "Eric Wieling" <EWieling@nyigc.com><br />
<b>Sent</b>: Friday, July 22, 2011 11:06 AM<br />
<b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br />
<b>Subject</b>: Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X</span><br />
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<p class="MsoNormal"><span style="font-family: "calibri","sans-serif"; color: #1f497d; font-size: 11pt;">Asterisk supports reinvites (if reinvites are enabled in sip.conf), just not changing codecs in the middle of the call. If anyone has managed to get it to work, I’d love to hear about it. </span></p>
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<p class="MsoNormal"><b><span style="font-family: "tahoma","sans-serif"; font-size: 10pt;">From:</span></b><span style="font-family: "tahoma","sans-serif"; font-size: 10pt;"> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Matteo Campana<br />
<b>Sent:</b> Friday, July 22, 2011 11:00 AM<br />
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br />
<b>Subject:</b> Re: [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X</span></p>
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<p class="MsoNormal">On Fri, Jul 22, 2011 at 4:46 PM, Eric Wieling <<a href="mailto:EWieling@nyigc.com">EWieling@nyigc.com</a>> wrote:</p>
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<p class="MsoNormal" style="mso-margin-top-alt: auto; mso-margin-bottom-alt: auto;"><span style="color: #1f497d; font-size: 11pt;">Asterisk does not support changing codecs on the fly.</span></p>
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<p class="MsoNormal">And why asterisk sends 200 OK to the provider, if does not support its re-invite?</p>
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<p class="MsoNormal">M.</p>
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<p class="MsoNormal" style="mso-margin-top-alt: auto; mso-margin-bottom-alt: auto;"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Matteo Campana<br />
<b>Sent:</b> Friday, July 22, 2011 10:39 AM<br />
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br />
<b>Subject:</b> [asterisk-users] Question about codec re-negotiation in asterisk 1.4.X</span></p>
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<p class="MsoNormal" style="mso-margin-top-alt: auto; mso-margin-bottom-alt: auto;">Hi all,</p>
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<p class="MsoNormal" style="mso-margin-top-alt: auto; mso-margin-bottom-alt: auto;">I have a major issue with a codec renegotiation in an asterisk 1.4.33.1 setup, which leads me to ask a general question about asterisk 1.4.X codec negotiation: asterisk can support a re-negotiation of a codec "on the fly" through a re-Invite? If my SIP provider sends me a re-invite changing codec from g729 to g711, asterisk properly handle the matter?</p>
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<p class="MsoNormal" style="mso-margin-top-alt: auto; mso-margin-bottom-alt: auto;">I see in the trace that asterisk responds 200 OK to the provider, but <span style="text-decoration: underline;">does not send the re-invite to the UAC, and stops to send rtp to the UAC</span>.</p>
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