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<p class=MsoNormal><span class=postbody><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>Hi: </span></font></span><br>
<br>
<span class=postbody>Actually I have a problem trying to send the calls from an
Asterisk to a Cisco AS5350. Sometimes the DTMF is not reconigzed. </span><br>
<br>
<span class=postbody>This is my cisco peer configuration: </span><br>
<br>
<span class=postbody>dial-peer voice 6 voip </span><br>
<span class=postbody>service debit </span><br>
<span class=postbody>session protocol sipv2 </span><br>
<span class=postbody>incoming called-number 571638.... </span><br>
<span class=postbody>dtmf-relay rtp-nte digit-drop </span><br>
<span class=postbody>codec g723r63 bytes 48 </span><br>
<span class=postbody>no vad </span><br>
<span class=postbody>! </span><br>
<br>
<span class=postbody>And this is my configuration in the sip.conf file: </span><br>
<br>
<span class=postbody>;GATEWAY IN BILLING </span><br>
<span class=postbody>[gw-saliente-5] </span><br>
<span class=postbody>type=friend </span><br>
<span class=postbody>host=xxx.xxx.xxx.xxx </span><br>
<span class=postbody>disallow=all </span><br>
<span class=postbody>allow=g729 </span><br>
<span class=postbody>allow=g723 </span><br>
<span class=postbody>allow=ulaw </span><br>
<span class=postbody>dtmfmode=rfc2833 </span><br>
<span class=postbody>qualify=yes </span><br>
<span class=postbody>canreinvite=yes </span><br>
<span class=postbody>insecure=very </span><br>
<span class=postbody>nat=no </span><br>
<span class=postbody>context=to-billing </span><br>
<span class=postbody>accountcode=BillingBoyra </span><br>
<br>
<span class=postbody>Best Regards, </span><br>
<br>
<st1:PersonName ProductID="Juan Jaramillo" w:st="on"><span class=postbody>Juan
Jaramillo</span></st1:PersonName><span class=postbody><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
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