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<DIV><FONT face=Arial size=2>All,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I've got a strange issue here. When I make a SIP
call to say my voicemail app, I hear audio just fine. However when I dial from
PSTN into my Asterisk box, I see that its playing the voice files, but I hear
nothing, then the call drops. I'm running Fedora Core 6, and Asterisk 1.2.24.
CLI output below. T-1 is PRI, showing normal, dchannel is up as well. Any help
is greatly appreciated.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2>Jon</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> -- Accepting call from '2125551212' to
'6465551212' on channel 0/23, span 4<BR> -- Executing
VoiceMail("Zap/95-1", "u100") in new stack<BR> -- Playing
'vm-theperson' (language 'en')<BR> -- Playing 'digits/1'
(language 'en')<BR> -- Playing 'digits/0' (language
'en')<BR> -- Playing 'digits/0' (language
'en')<BR> -- Playing 'vm-isunavail' (language
'en')<BR> -- Playing 'vm-intro' (language
'en')<BR> -- Channel 0/23, span 4 got hangup request, cause
34<BR> == Spawn extension (default, 6465551212, 1) exited non-zero on
'Zap/95-1'<BR> -- Hungup
'Zap/95-1'<BR></FONT></DIV></BODY></HTML>