[asterisk-bugs] [Asterisk 0010342]: On Mac OS X PowerPC, Asterisk 1.4.x stops bridging new calls shortly after start
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue Jul 31 09:37:44 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=10342
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Reported By: jcovert
Assigned To:
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Project: Asterisk
Issue ID: 10342
Category: Core/Channels
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.4.9
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 07-31-2007 09:10 CDT
Last Modified: 07-31-2007 09:37 CDT
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Summary: On Mac OS X PowerPC, Asterisk 1.4.x stops bridging
new calls shortly after start
Description:
I have unsuccessfully tried to get every 1.4.x version of Asterisk to work
on Mac OS X PowerPC, most recently Asterisk 1.4.9 on Mac OS X 10.4.10.
Asterisk starts up fine, and works for a short while. But after about the
third or fourth bridged call (local sip soft client, SJphone, or locally
connected SIP ATAs Cisco ATA 186), RTP read too short errors start
ocurring, or other dead call problems. It is still possible to connect to
IVRs within asterisk, and inbound IAX2 calls from a 1.2.22 version of
Asterisk to the 1.4.9 version still work, but originating SIP devices are
unable to reliably make outbound calls.
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jcovert - 07-31-07 09:37
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More information. If a open a second asterisk console at the same time,
i.e. another asterisk -vvvr, the following happens in the original console
(compare with the attached file with the full debug output):
-- Goto (cnetlookup,14559900,1)
-- Executing [14559900 at cnetlookup:1]
GotoIf("SIP/localsjphone-0184a600", "1?v1d4") in new stack
-- Goto (cnetlookup,14559900,6)
[Jul 31 10:39:45] WARNING[7939]: asterisk.c:923 netconsole: poll returned
< 0: Bad file descriptor
(the above error is repeated 82 times)
-- Remote UNIX connection disconnected
(at this point the second console is bounced off)
-- Executing [14559900 at cnetlookup:6] Set("SIP/localsjphone-0184a600",
"result=guest at speakup.octothorp.org/14559900") in new stack
and all is well and we actually get a good connection now.
But the very next call fails again. I note on that call that the RTP Read
too short message only appears 93 times and then stops, but there is still
no audio.
Issue History
Date Modified Username Field Change
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07-31-07 09:37 jcovert Note Added: 0068113
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