What happens if in your sip.conf you set<br><br>disallow=all<br>allow=g723,gsm<br><br>And then allow both codec in the phone?<br><br><div><span class="gmail_quote">On 11/19/06, <b class="gmail_sendername">Ray Jackson</b> <
<a href="mailto:ray@jacksonz.net">ray@jacksonz.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;">All,<br><br>Our users have a softphone client that supports the G723 Codec which we
<br>want to use for bandwidth reasons, however we do not wish (or have the<br>funds) to license the codec on our Asterisk servers. We have G723<br>pass-through working between the clients just fine, however calls fail<br>
when terminating with Asterisk itself (i.e. Voicemail) or out to the<br>PSTN due to transcoding issues.<br><br>If it possible to build the config into our Asterisk servers so that<br>calls between the softphones defaults to G723 pass-through, whilst all
<br>other calls (PSTN, Voicemail etc.) default to GSM as their preferred<br>codec? Is there a way of getting Asterisk to be smart with Codec<br>negotitation and figure out which codec the other end of the call is<br>capable of before negotitating back to the Softphone with the selected
<br>codec? I assume you would have to do something in the dial plan? I saw<br>the SIP_CODEC variable, but couldn't make it work.<br><br>Any advice would be very welcome!<br><br>Cheers,<br>Ray<br><br>-----------------------------------------------------------------------------------------------
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