[asterisk-dev] How to debug DAHDI pseudo timer problem..

[Peter Tap]

Folks,

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. 

All the hard and soft phones are working as expected.

Later, I proceeded to configure MeetMe(). I am using DAHDI pseudo timer for it.

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.

  == Using SIP RTP CoS mark 5
    -- Executing [600 at FromCiscoPhone:1] MeetMe("SIP/101-b7b03298", "1234,s1") in new stack
  == Parsing '/etc/asterisk/meetme.conf':   == Found
    -- Created MeetMe conference 1023 for conference '1234'
    -- <SIP/101-b7b03298> Playing 'conf-getpin.ulaw' (language 'en')
    -- Hungup 'DAHDI/pseudo-1700836616'
  == Spawn extension (FromCiscoPhone, 600, 1) exited non-zero on 'SIP/101-b7b03298'

More information about the problem and the steps I took can be found on the asterisk users forum athttp://forums.digium.com/viewtopic.php?f=1&t=71311.

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.

Regards,
Peter


      
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