<div>hi Palma,</div>
<div>as the SJ initiate the call, it will allways go with GSM Codec as the codec should be identical used on both sides. as you do not have G729 on the SJ, it will never use G729.</div>
<div>furthermore, i think that if the GSM will not work, then the second option choosed would be PCMA</div>
<div> </div>
<div>i hope i gave you a way further.</div>
<div> </div>
<div>Mickey<br><br> </div>
<div><span class="gmail_quote">On 2/23/06, <b class="gmail_sendername">Álvaro Palma</b> <<a href="mailto:apalma@opschile.cl">apalma@opschile.cl</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000.<br>The codec order on each one is the next:
<br><br>SJPhone: GSM - iLBC - PCMA - PCMU<br>GXP2000: G729 - GSM - PCMA - PCMU<br><br>(I have a G729 license, so there's no problem with transcoding G729)<br><br>In my sip.conf, I've defined the following codec order:<br>
<br>disallow=all<br>allow=g729<br>allow=gsm<br>allow=g726<br>allow=alaw<br>allow=ulaw<br><br>And my peers shows this order correctly:<br><br>Codecs : 0x11e (gsm|ulaw|alaw|g726|g729)<br>Codec Order : (g729,gsm,g726,alaw,ulaw)
<br><br>Canreinvite is set to NO.<br><br>But, if I initiate a call from the softphone to GXP-2000, Asterisk<br>always to the GXP phone GSM as the first codec choice, instead of G729,<br>as I could check with ethereal running in the same server than Asterisk.
<br>The SIP INVITE from Asterisk to GXP looks like<br><br>*****************************************************************<br>Request-Line: INVITE <a href="mailto:sip:5805907@192.168.1.105">sip:5805907@192.168.1.105</a>;user=phone SIP/2.0
<br>Message Header<br>Message body<br> Session Description Protocol<br> Session Description Protocol Version (v): 0<br> Owner/Creator, Session Id (o): root 27682 27682 IN IP4 <a href="http://192.168.1.2">
192.168.1.2</a><br> Session Name (s): session<br> Connection Information (c): IN IP4 <a href="http://192.168.1.2">192.168.1.2</a><br> Time Description, active time (t): 0 0<br> Media Description, name and address (m): audio 14224 RTP/AVP 3
<br>18 111 8 0<br> Media Type: audio<br> Media Port: 14224<br> Media Proto: RTP/AVP<br> Media Format: GSM 06.10<br> Media Format: ITU-T G.729<br> Media Format: 111
<br> Media Format: ITU-T G.711 PCMA<br> Media Format: ITU-T G.711 PCMU<br> Media Attribute (a): rtpmap:3 GSM/8000<br> Media Attribute Fieldname: rtpmap<br> Media Attribute Value: 3 GSM/8000
<br> Media Attribute (a): rtpmap:18 G729/8000<br> Media Attribute Fieldname: rtpmap<br> Media Attribute Value: 18 G729/8000<br> Media Attribute (a): fmtp:18 annexb=no<br> Media Attribute Fieldname: fmtp
<br> Media Attribute Value: 18 annexb=no<br> Media Attribute (a): rtpmap:111 G726-32/8000<br> Media Attribute Fieldname: rtpmap<br> Media Attribute Value: 111 G726-32/8000<br> Media Attribute (a): rtpmap:8 PCMA/8000
<br> Media Attribute Fieldname: rtpmap<br> Media Attribute Value: 8 PCMA/8000<br> Media Attribute (a): rtpmap:0 PCMU/8000<br> Media Attribute Fieldname: rtpmap<br> Media Attribute Value: 0 PCMU/8000
<br> Media Attribute (a): silenceSupp:off - - - -<br> Media Attribute Fieldname: silenceSupp<br> Media Attribute Value: off - - - -<br>*****************************************************************
<br><br>So it can be clearly seen how GSM is before G729.<br><br>Anybody knows if this is an existing bug? Or am I doing something wrong?<br>Thanks a lot for your attention.<br><br>--<br>Atly.<br>Alvaro Palma<br>_______________________________________________
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