Hi Klaus,<br><br> I had done the patch from <a href="http://sip.fontventa.com">sip.fontventa.com</a>.<br><br> and also added the line <br>[amr]<br>octet-aligned=1<br><br>in my codecs.conf file.<br><br>but that did not help me.<br>
<br><br>regards,<br>aster<br><br><br><div><span class="gmail_quote">On 8/25/08, <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="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
Have you patched Asterisk with the AMR patch from <a href="http://sip.fontventa.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip.fontventa.com</a>?<br><br> klaus<br><br> aster vdo schrieb:<br>
<blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex"><span class="q"> Hi,<br>
<br> I am doing a video call from a 3G to SIP.<br><br> The video works fine, but there is no audio for both the parties.<br><br></span> I using Asterisk <a href="http://1.4.20.1" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">1.4.20.1</a> <<a href="http://1.4.20.1" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://1.4.20.1</a>><div>
<span class="e" id="q_11bf909e04bcb2c4_3"><br><br> and i get the following warning message on asterisk<br><br> WARNING[15840] chan_sip.c: Asked to transmit frame type 8192, while native formats is 0x4 (ulaw)(4) read/write = 0x0 (nothing)(0)/0x0 (nothing)(0)<br>
<br> and the show channel commands results in the following output.<br><br><br> *CLI> core show channel Zap/1-1<br> -- General --*CLI><br> Name: Zap/1-1<br> Type: Zap<br> UniqueID: 1219243635.23<br> Caller ID: XXXXXXXX<br>
Caller ID Name: (N/A)<br> DNID Digits: XXXXXXX<br> State: Up (6)<br> Rings: 1<br> NativeFormats: 0x44 (ulaw|slin)<br> WriteFormat: 0x4 (ulaw)<br> ReadFormat: 0x4 (ulaw)<br> WriteTranscode: No<br> ReadTranscode: No<br> 1st File Descriptor: 19<br>
Frames in: 3275<br> Frames out: 2532<br> Time to Hangup: 0<br> Elapsed Time: 0h1m5s<br> Direct Bridge: <none><br> Indirect Bridge: <none><br> -- PBX --<br> Context: default<br> Extension: s<br> Priority: 2<br>
Call Group: 0<br> Pickup Group: 0<br> Application: h324m_gw<br> Data: dial@cell_to_sip<br> Blocking in: ast_waitfor_nandfds<br> Variables:<br> ul1=65535<br> CALLEDTON=33<br> ANI2=0<br> TRANSFERCAPABILITY=DIGITAL<br><br> CDR Variables:LI><br>
level 1: clid=XXXXXXXX<br> level 1: src=XXXXXXXX<br> level 1: dst=s<br> level 1: dcontext=default<br> level 1: channel=Zap/1-1<br> level 1: lastapp=h324m_gw<br> level 1: lastdata=dial@cell_to_sip<br> level 1: start=2008-08-20 15:47:15<br>
level 1: answer=2008-08-20 15:47:30<br> level 1: end=2008-08-20 15:47:30<br> level 1: duration=0<br> level 1: billsec=0<br> level 1: disposition=ANSWERED<br> level 1: amaflags=DOCUMENTATION<br> level 1: uniqueid=1219243635.23<br>
<br><br><br> Is there any thing i am doing wrong..<br><br><br> regards<br><br> aster<br></span></div></blockquote></blockquote></div><br>