<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt"><div>Folks,<br><br>I have been using Asterisk for quite some time. Recently, I installed the latest version of Asterisk on a brand new box. We don't use any digium device anymore. Our inbound/outbound calls are through VoicePulse. <br><br>All the hard and soft phones are working as expected.<br><br>Later, I proceeded to configure MeetMe(). I am using DAHDI pseudo timer for it.<br><br>The problem I am running into is that after a connection is established to the conference room, I get disconnected in a minute or two. Here is the output information while running asterisk with a lots of -v option.<br><br> == Using SIP RTP CoS mark 5<br> -- Executing [600@FromCiscoPhone:1] MeetMe("SIP/101-b7b03298", "1234,s1") in new stack<br> == Parsing
'/etc/asterisk/meetme.conf': == Found<br> -- Created MeetMe conference 1023 for conference '1234'<br> -- <SIP/101-b7b03298> Playing 'conf-getpin.ulaw' (language 'en')<br> -- Hungup 'DAHDI/pseudo-1700836616'<br> == Spawn extension (FromCiscoPhone, 600, 1) exited non-zero on 'SIP/101-b7b03298'<br><br>More information about the problem and the steps I took can be found on the asterisk users forum at<span> <a target="_blank" href="http://forums.digium.com/viewtopic.php?f=1&t=71311">http://forums.digium.com/viewtopic.php?f=1&t=71311</a>.</span><br><br>I am thinking my last resort is to debug asterisk and see what is happening. I am a developer and am familiar with gdb. It also appears asterisk is compiled with -g3 option by default. I would appreciate any pointers on debugging asterisk, where to set the break points, etc.<br><br>Regards,<br>Peter<br><br></div>
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