[asterisk-dev] 1.4 open issues (Major and Crash)

[Dan Austin]

Sorted by Severity as listed in Mantis (a couple maye have
changed as I transcribed this list) 32 total issues listed
as open with a severity of major or above.

Reported against 1.4.0b3
Crash (3):
8232	SIP Segfault with high call setup volume 
		*  A good bit of discussion and feedback, but
			the root cause remains elusive
	Last Activity: 11/12

8228	1.4 crashes with Segmentation Fault when a call is transfered
	from a queue 
		*  bt and logs attached
	Last Activity: 10/27

8138	Asterisk crash when receving or sending out call for Google 
	Talk 
		*  Refers to a patch in comment 53017.  No further
			feedback.
	Last Activity: 10/19

Major(5):
8340	Transfers are not bridging properly 
		*  Debug logs attached, no patch
	Last Activity: 11/13

8298	Recording synchronization fails due to bad number of 
	samples correlation in ast_read / ast_write 
		* Trivial code posted inline, original poster
			and one tester indicate it is a valid fix.
			Awaiting developer review and commit
	Last Activity:  11/14

8287	SLIN codec noise 
		*  Waiting on proper debug logs
	Last Activity: 11/09

8193	Asterisk to Gtalk audio shuts off after 30 seconds into call 
		* Refers to a possible fix already in 1.4, not further
			feedback.
	Last Activity: 11/01

8189	Jitterbuffer PLC fix for IAX2 channel and other issues with
	 jitterbuffer 
		*  Reporter has posted a patch and test results.
			No feedback or other developer comments.
	Last Activity: 10/23

Reported against SVN (1.4):
Crash(4):
8183	Module unload causes segfault 
		*  Waiting on more detailed debugging logs
	Last Activity: 11/13

8146	Asterisk crashes with jitterbuffer + mixmonitor 
		*  Possible fix attached, waiting on testing feedback
			and developer review
	Last Activity: 11/09

8068	Asterik 1.2 - > asterisk 1.4 (trunk) ooh323 crash 
		*  Logs an bt attached.  Waiting on developer review
	Last Activity: 11/02

8229	Crash in chan_h323 when dialing invalid non existing extension
		*  No comments or feedback, inline bt
	Last Activity: 10/25

Major(4):
8365	Reinvite is using local IP of NATed device 
		*  No comments or feedback
	Last Activity: 11/14

8325	IAX - one way audio, when network jitter occur 
		*  Possible relationship to 8273.  Log attached
	Last Activity: 11/10

8214	1.4 trunk with-odbc fails on RHEL4 		
		*  Build tools related.  A reporter suggests installing
			upgrading libtool-ltdl-devel resolves the issue
	Last Activity: 11/09

8273	After a while of operation, IAX becomes behaving incorrectly,
	no audio or 1-way, and no-answer 
		*  Logs attached, waiting on developer review.  Might 
			identify which commit introduced the issue
	Last Activity: 11/02

Reported against Trunk (but appears to be meant for 1.4)
Crash(5):
7774	Asterisk crashes with chan_skinny 
		*  Patch and positive feedback.  Was waiting on
			additional feedback.
	Last Activity: 11/09

8238	Asterisk 1.4 99% cpu usage and crashing 
		*  May have been resolved, bug appears to be going
			off-topic.
	Last Activity: 11/08

8305	app_mixmonitor crashes asterisk 
		*  Debug logs attached 
	Last Activity: 11/14

7607	coredump on blind transfer unless compiled with 
	DEBUG_CHANNEL_LOCKS 
		*  Patch attached.  feedback indicates partial
			success, no longer crashing on the unlock
			but transfers do not succeed.
	Last Activity: 11/13

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: 11/07

Major(11):
8338	T.38 Fallback fails 
		*  Active feedback and testing logs attached.
	Last Activity: 11/14

8152	Transcoding not working for SIP calls with reinvite=yes 
		*  Original reporter can not perform any testing
			for some time, another reporter has volunteered.
	Last Activity: 11/14

7351	SIP CANCEL fails due to wrong Contact: URI 
		*  Needs testing against recent commit
	Last Activity: 11/14

7844	t.38 passthrough not working when endpoints are behind a NAT 
		*  Appears to be resolved, awaiting feedback from
			confirmed working T38 endpoints.
	Last Activity: 11/12

7679	T.38 passthrough is not working between two Sipuras 2100 
		*  Awaiting feedback (closely related to 7844)
	Last Activity: 11/12

8078	T38 relay doesn't work between Audiocodes Tulip AC494 ATAs 
		*  First issue (case sensitivity in SDP resolved)
			suffers from the same failure modes as
			7844 and 7679, awating feedback.
	Last Activity: 11/12

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: 11/09

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

8192	SDP information incorrect if canreinvite is globally undefined 
		*  Waiting on reporter to provide debug details
	Last Activity: 11/06

7988	cancellation does not stop ooh323 dialing an empty/no route
device 
		*  Logs attached, waiting on developer.
	Last Activity: 11/06

8066	hanguponpolarityswitch hangs up on incoming call during ring
phase 
		*  Two reporters took this offline to tag team a fix
	Last Activity: 10/23