I should have posted the logs when the call is accepted....here it is:<br><br><font size="1">-- SIP/5001-081ef020 answered Gtalk/guan.alex-e086<br>[Sep 19 12:13:47] DEBUG[31226]: rtp.c:1402 ast_rtp_early_bridge: Channel 'Gtalk/guan.alex-e086'
<span style="font-weight: bold;">has no RTP</span>, not doing anything<br>[Sep 19 12:13:47] NOTICE[31226]: chan_gtalk.c:1331 gtalk_indicate: Don't know how to indicate condition '-1'<br>[Sep 19 12:13:47] DEBUG[31226]: channel.c
:2652 set_format: Set channel Gtalk/guan.alex-e086 to write format ulaw<br>[Sep 19 12:13:47] DEBUG[31226]: chan_gtalk.c:513 gtalk_answer: Answer!<br><br>JABBER: asterisk OUTGOING: <iq type='set' to='<a href="http://guan.alex@gmail.com/Talk.v96A3F055BC">
guan.alex@gmail.com/Talk.v96A3F055BC</a>' from='<a href="http://leozhang99@gmail.com/asterisk9642378C">leozhang99@gmail.com/asterisk9642378C</a>' id='aaaai'><session xmlns='<a href="http://www.google.com/session">http://www.google.com/session
</a>' type='accept' initiator='<a href="http://guan.alex@gmail.com/Talk.v96A3F055BC">guan.alex@gmail.com/Talk.v96A3F055BC</a>' id='1314397402'><description xmlns='<a href="http://www.google.com/session/phone">http://www.google.com/session/phone
</a>' xml:lang='en'><payload-type id='0' name='PCMU' clockrate='8000' bitrate='64000'/><payload-type id='100' name='EG711U' clockrate='8000' bitrate='64000'/><payload-type id='102' name='iLBC' clockrate='8000' bitrate='13300'/><payload-type id='106' name='telephone-event' clockrate='8000'/></description><transport xmlns='
<a href="http://www.google.com/transport/p2p'/">http://www.google.com/transport/p2p'/</a>></session></iq><br>[Sep 19 12:13:47] WARNING[31226]: rtp.c:3019 ast_rtp_bridge: <span style="font-weight: bold;">Can't find native functions for channel 'Gtalk/guan.alex-e086'
</span><br> -- Native bridging Gtalk/guan.alex-e086 and SIP/5001-081ef020 ended<br></font><br>Thanks again,<br>Alex<br><br><br><div><span class="gmail_quote">On 9/19/06, <b class="gmail_sendername">Alex Guan</b> <<a href="mailto:guan.alex@gmail.com">
guan.alex@gmail.com</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;"><div>Gang,<br><br>With the latest code I am having audio issue with the chan_gtalk and wondering if anyone else had the same problem
<br><br>- I was able to set up the call between Asterisk and my gTalk account, but there was no audio
<br>- Looking closer, I am seeing these messages for an incoming call: <br><br><div style="margin-left: 40px;"><font size="1"> -- SIP/5001-081ef020 is ringing<br>[Sep 19 12:13:44] DEBUG[31226]: rtp.c:1402 ast_rtp_early_bridge: Channel 'Gtalk/guan.alex-e086'
<span style="font-weight: bold;">has no RTP</span>, not doing anything<br>[Sep 19 12:13:44] NOTICE[31226]: chan_gtalk.c:1331 gtalk_indicate: Don't know how to indicate condition '3'<br>[Sep 19 12:13:44] DEBUG[31226]: channel.c
:2278 ast_indicate_data: Driver for channel 'Gtalk/guan.alex-e086' does not support indication 3, emulating it<br>[Sep 19 12:13:44] DEBUG[31226]: channel.c:2427 ast_prod: Prodding channel 'Gtalk/guan.alex-e086'<br>[Sep 19 12:13:44] DEBUG[31226]:
channel.c:2652 set_format: Set channel Gtalk/guan.alex-e086 to write format slin</font><br></div><br><div style="margin-left: 40px;"><font size="1">JABBER: asterisk INCOMING: <iq to="<a href="http://leozhang99@gmail.com/asterisk9642378C" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
leozhang99@gmail.com/asterisk9642378C</a>" type="set" id="98" from="<a href="http://guan.alex@gmail.com/Talk.v96A3F055BC" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
guan.alex@gmail.com/Talk.v96A3F055BC</a>"><session type="transport-info" id="1314397402" initiator="
<a href="http://guan.alex@gmail.com/Talk.v96A3F055BC" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">guan.alex@gmail.com/Talk.v96A3F055BC</a>" xmlns="<a href="http://www.google.com/session" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
http://www.google.com/session</a>"><transport xmlns="
<a href="http://www.google.com/transport/p2p" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.google.com/transport/p2p</a>"><candidate name="rtp" address="<a href="http://10.10.150.96" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
10.10.150.96</a>" port="4923" preference="1" username="NXkxfCIYx2p8tMNc" protocol="udp" generation="0" password="mVSwEuvfiU9y062J" type="local" network="0"/></transport></session></iq>
<br> -- JABBER: I Dont have an IQ!!!</font><br></div><br style="background-color: rgb(51, 51, 51); color: rgb(255, 0, 0);">Does anybody know why there is no RTP? What am I missing here?<br><br>And here is my gtalk.conf
:<br><br><div style="margin-left: 40px;"><font size="1">[general]<br>context=gtalk<br>allowguest=yes <br><br>[guest] <br>disallow=all<br>allow=ulaw<br>context=guest<br><br>[guan.alex]<br>username=
<a href="mailto:guan.alex@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">guan.alex@gmail.com</a> <br>disallow=all<br>allow=ulaw<br>allow=ilbc<br>allow=isac<br>context=gtalk<br>connection=asterisk
<br><br></font></div>My jabber.conf:<br>
<br><br><div style="margin-left: 40px;"><font size="1">[general]<br>debug=yes <br>autoprune=yes <br>autoregister=yes <br><br>[asterisk]
<br>type=client <br>serverhost=<a href="http://talk.google.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">talk.google.com</a> <br>username=<a href="mailto:xxxxxxx@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
xxxxxxx@gmail.com</a> ;;<br>secret=xxxxxx
<br>port=5222 <br>usetls=yes ;<br>usesasl=yes <br>buddy=<a href="mailto:guan.alex@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
guan.alex@gmail.com</a> <br>statusmessage="online"
<br>timeout=100 </font><br></div><br>Your help is greatly appreciated!<br><br>Thanks,<br></div><div><span class="sg">Alex<br>
</span></div></blockquote></div><br>