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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to