[asterisk-dev] 1.4 open issues (Major and Crash) (second pass)
Dan Austin
Dan_Austin at Phoenix.com
Fri Dec 22 14:18:44 MST 2006
Sorted by Severity as listed in Mantis (a couple maye have
changed as I transcribed this list) 34 total issues listed
as open with a severity of major or above. Issues reported
in 1.2 that may effect 1.4 are not covered here, only new
issues/regressions/etc
Reported against 1.4.0b4
Crash (1):
8639 iax2 crash on transfer
* No comments or feedback
Last Acitivy: 12/20
Major (2):
8644 IAX2 outgoing calls not working
* Reporter's tests show it might be related
to managed DNS lookups
Last Activity: 12/20
8662 Parked Calls drop immediately
* Debug log attached. No comments
Last Activity: 12/22
Reported against 1.4.0b3
Crash (2) (was 3):
8228 1.4 crashes with Segmentation Fault when a call is transfered
from a queue
* Still waiting on feedback and
Last Activity: 12/18
8434 1.4.0b3 crashed during call transfer
* Crash log posted. Dewveloper is looking into
whisper paging as a possible cause
Last Activity: 12/22
8573 Asterisk core when busy in a zap call
* Testing requested against latest beta/branch
Last Activity: 12/14
Major(4)(was 5):
8298 Recording synchronization fails due to bad number of
samples correlation in ast_read / ast_write
* Moved to 1.2 chain. Waiting on confirmation of
the fix.
Last Activity: 12/14
8193 Asterisk to Gtalk audio shuts off after 30 seconds into call
* Patches tested, issue remains
Last Activity: 12/20
8189 Jitterbuffer PLC fix for IAX2 channel and other issues with
jitterbuffer
* Fix implimented and commited. Waiting on feedback?
Last Activity: 11/24
8521 Reading DTMF fails on SIP and IAX2
* Looks to be related to IAX softphones (opal library)
Last Activity: 12/20
Reported against SVN (1.4):
Crash(3) (was 4):
8183 Module unload causes segfault
* More details provided, no resolution listed
Last Activity: 12/05
8146 Asterisk crashes with jitterbuffer + mixmonitor
* Possible fix attached, waiting on testing feedback
and developer review.
Last Activity: 12/15
8068 Asterik 1.2 - > asterisk 1.4 (trunk) ooh323 crash
* Logs an bt attached. Waiting on developer review
Last Activity: 11/02 (no change)
Major(3)(was 4):
8325 IAX - one way audio, when network jitter occur
* Might be related to use of chan_skinny.
Waiting on test results without skinny endpoint
Last Activity: 12/22
8214 1.4 trunk with-odbc fails on RHEL4
* Developer is working on improving the build tool
output to identify the need to upgrade a
dependancy
Last Activity: 12/22
8273 After a while of operation, IAX becomes behaving incorrectly,
no audio or 1-way, and no-answer
* Additional debugging output requested.
Last Activity: 12/22
Reported against Trunk (but appears to be meant for 1.4)
Crash(3) (was 5):
8305 app_mixmonitor crashes asterisk
* Suggested relationship to 8146
Last Activity: 11/16
7607 coredump on blind transfer unless compiled with
DEBUG_CHANNEL_LOCKS
* Patch attached. Waiting on test results agaist
current branch/trunk
Last Activity: 11/17
7885 segfault when zap channels are full (calls are
Originate'd via AMI and exacerbated by app_amd)
* Seems to have made progress on the segfault
issue, but may have a related memory leak.
Last Activity: 11/22
Major(16) (was 11):
8338 T.38 Fallback fails
* Active feedback and testing logs attached.
Last Activity: 11/16
8152 Transcoding not working for SIP calls with reinvite=yes
* Still present as of 47654
Last Activity: 11/20
7351 SIP CANCEL fails due to wrong Contact: URI
* Needs testing against recent commit
Last Activity: 11/16
7844 t.38 passthrough not working when endpoints are behind a NAT
* Appears to be resolved, awaiting feedback from
confirmed working T38 endpoints.
(some fixes committed)
Last Activity: 12/04
7679 T.38 passthrough is not working between two Sipuras 2100
* Awaiting feedback (closely related to 7844)
Last Activity: 12/06
7706 Redirecting Local channels to meetme causes deadlock upon hangup
* Reporter modified the dialplan to elimiate Local
channel usage, but the issue persists. Now
suspected to reside in AMI.
Last Activity: 11/16
7987 ooh323 does not work in failover test case if the first
destination is an empty/no route device
* Logs attached, waiting on developer.
Last Activity: 11/07 (no change)
7988 cancellation does not stop ooh323 dialing an empty/no route
device
* Logs attached, waiting on developer.
Last Activity: 11/06 (no change)
8066 hanguponpolarityswitch hangs up on incoming call during ring
phase
* Reporter is waiting on the carrier to enable
a feature to test this.
Last Activity: 11/14
8416 Asterisk process at 100% CPU
* A lot of testing has not shown the cause.
Developer suggests increasing the debug log
level
Last Activity: 12/22
8597 SIP, dtmf-relay, feature key presses being ignored
* This does not appear to be a configuration issue,
at least not directly with Asterisk. Feedback
needed to see if an endpoint is using VAD or
silence suppression.
Last Activity: 12/22
8555 even though configure is ran with --prefix, make install
tries to mkdir /var/lib/asterisk
* Build tool related. Being discussed
Last Activity: 12/21
8562 Asterisk stays in the audio path if "t" option in Dial is used
* Looks like a feature enhancement for additional
SIP-INFO DTMF support vs. a bug.
Last Activity: 12/19
8593 dstchannel in cdr is empty when transfer call
* Related to 8221
Last Activity: 12/19
8571 REFER not working with Cisco hardware when doing local attended
call transfer
* Debug logs attached.
Last Activity: 12/18
8524 Via: header may contain multiple values
* Patch provided, waiting on developer feedback
Last Activity: 12/18
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