HI all,<br><br>I'm having the same problem here. Need to interconnect simple SIP phones (soft or hard) with H324M calls.<br>So I either need a SIP phone that speaks AMR, or asterisk needs to transcode.<br>Have tried to implement the amr codec as described by Paul (
<a href="http://lists.digium.com/pipermail/asterisk-dev/2007-March/026446.html">http://lists.digium.com/pipermail/asterisk-dev/2007-March/026446.html</a>)<br>All compiles well, but I don't get any audio throughput.<br>
<br>Has anybody had more luck with this?<br><br>For the record, I know there's a licensing issue with AMR.<br>But since VoiceAge is giving away free amr codec samples (<a href="http://www.voiceage.com/freecodecs.php">
http://www.voiceage.com/freecodecs.php</a>) for development purpose,<br>I guess we can try develop it into Asterisk...<br><br>Koen<br><br><div><span class="gmail_quote">On 7/18/07, <b class="gmail_sendername">Klaus Darilion
</b> <<a href="mailto:klaus.mailinglists@pernau.at">klaus.mailinglists@pernau.at</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;">
<br><br>Sergio Garcia wrote:<br>> As I said, the patch is not mine, I'll try to contact Paul to see if he can<br>> solve it.<br>><br><br><br>Hi Sergio!<br><br>Have you ever managed to gateway a call from H324M to a SIP client? If
<br>yes - did you used AMR at the SIP client or did Asterisk transcoding? If<br>asterisk did transcoding - which AMR codec do you used? If Asterisk did<br>not transcoding - which SIP client have you used?<br><br>thanks<br>
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