[Asterisk-Users] Problems with meetme dropping audio during call

[James Moore]

I'm using meetme with three SIP calls, updated to the latest Asterisk CVS
(as of around 10am Nov 3).  After a minute or two we start getting
substantial cutouts of the audio during the call.  I'm using the ztdummy
timer, also the latest CVS release.

Suggestions for things to look at?

There are some warnings on the console - I've included that output below.

Fedora Core 4, 2.6.13-1.1532_FC4 kernel.

- James

---------------------------

    -- Created MeetMe conference 1023 for conference '1234'
    -- Playing 'conf-onlyperson' (language 'en')
    -- Executing MeetMe("SIP/xx.xx.xx.xx-097cc028", "1234") in new stack
    -- Executing MeetMe("SIP/xx.xx.xx.xx-097e6b30", "1234") in new stack
    -- Executing MeetMe("SIP/xx.xx.xx.xx-097e0ce0", "1234") in new stack
set verbose 9
Verbosity is at least 9
*CLI> Nov  3 10:31:58 WARNING[17653]: app_meetme.c:1467 conf_run: Unable to
writ
e frame to channel: Success
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097cc028'
Nov  3 10:31:58 WARNING[17673]: app_meetme.c:1467 conf_run: Unable to write
fram
e to channel: Success
Nov  3 10:31:58 WARNING[17647]: app_meetme.c:1467 conf_run: Unable to write
fram
e to channel: No such file or directory
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097e0ce0'
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097b7a00'
Nov  3 10:31:59 WARNING[17658]: app_meetme.c:1467 conf_run: Unable to write
fram
e to channel: Success
    -- Hungup 'Zap/pseudo-477583753'
  == Spawn extension (default, 12069737581, 1) exited non-zero on
'SIP/xx.xx.xxx
.141-097e6b30'

*CLI>