asterisk users - Jun 2010 - 50 mantis issues marked 'Ready for Testing'

List,

Over the last few months we have managed to bring the total number of
issue on the tracker from 610+ to 537 (as of writing). While this is
good news, we still have a number of open issues that require testers
to help move them along.  Below, I have posted the oldest 50 issues
that are in the 'Ready for Testing' state.

Basically, we are looking for more people to step-up and test theses
patches.  Each issue below _should_ have an existing patch attached to
it. Simply download, patch asterisk, compile, installed and verify
asterisk works as expected; post your results to the existing mantis
issue.  Once each issue has been properly tested, we can continue
triaging and step closer to merging the code.

Let me know if you have questions / comments.

---
[patch] Calls are not matched to correct peer when using
callbackextension parameter
https://issues.asterisk.org/view.php?id=14340

[patch] Strange nasty sound (Because Asterisk tryes to handle new
voicemail, but there is no voicemails, voicemail isn't used)
https://issues.asterisk.org/view.php?id=15999

[patch] default say.conf for new number method doesnt handle all numbers
https://issues.asterisk.org/view.php?id=16102

[patch] Incorrectly configure (autoconf) when using the
--with-something=directory construct with non standard directories
https://issues.asterisk.org/view.php?id=14906

[patch] "make config" creates really wrong runlevels in Debian
(includes patch)
https://issues.asterisk.org/view.php?id=16172

[patch] Automatic gain normalization in meetme
https://issues.asterisk.org/view.php?id=14433

[patch] SMS FIX for motorola phones
https://issues.asterisk.org/view.php?id=15276

[patch] Hints do not have the correct state on initialization
https://issues.asterisk.org/view.php?id=16355

[patch] mpg123 <defunct>
https://issues.asterisk.org/view.php?id=16378

[patch] Asterisk will never retry after the first register to H.323 gk fails.
https://issues.asterisk.org/view.php?id=16076

[patch] [regression] The status of External SIP peer used as Queue
member is not updating correctly
https://issues.asterisk.org/view.php?id=16245

[patch] configure fails to detect spandsp/expose.h when not in system
include path
https://issues.asterisk.org/view.php?id=16342

[patch] Announce to user when they have been muted/unmuted from the AMI
https://issues.asterisk.org/view.php?id=16617

RFC2833 DTMF is not passed correctly when going SIP->IAX2->SIP
https://issues.asterisk.org/view.php?id=16625

[patch] Asterisk does not fully support SIP connections to Internet
Telephony Service Providers
https://issues.asterisk.org/view.php?id=16585

[patch] There is an Active call, even though device is Unregistered
from asterisk!
https://issues.asterisk.org/view.php?id=16693

[patch] Add AMI support for device states
https://issues.asterisk.org/view.php?id=16732

[patch] After AMI Bridge action the callerid's on the phones are not
updated.
https://issues.asterisk.org/view.php?id=16772

[patch] chan_sip will not retransmit an ACK
https://issues.asterisk.org/view.php?id=15802

[patch] Asterisk man page outdated
https://issues.asterisk.org/view.php?id=16505

[patch] Automatic add UniqueID to user event
https://issues.asterisk.org/view.php?id=16962

[patch] Perl script to import CDR text file to ODBC database table
https://issues.asterisk.org/view.php?id=17036

[patch] Problems with MeetMe and RT schedule dates
https://issues.asterisk.org/view.php?id=17034

[patch] Fix query with double backslash in string literals and stop log warnings
https://issues.asterisk.org/view.php?id=17077

[patch] Ability to use DUNDi channel variables when using dynamic weights
https://issues.asterisk.org/view.php?id=14560

[patch] app_festival hangs on reading from spawned subprocess
https://issues.asterisk.org/view.php?id=15706

[patch] Segmentation fault when using two codec modules that register
the same src and dst format
https://issues.asterisk.org/view.php?id=17092

[patch] Add busy detection
https://issues.asterisk.org/view.php?id=15581

[patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP
channel leak
https://issues.asterisk.org/view.php?id=17099

[patch] Updates to Application Documentation
https://issues.asterisk.org/view.php?id=17184

[patch] [regression] Overlap dialing to PSTN failing after 0016789
https://issues.asterisk.org/view.php?id=17085

[patch] Parking a call, then retrieving it with ParkedCall() kills the
ability to transfer the retrieved call.
https://issues.asterisk.org/view.php?id=16757

[patch] Proposed patch to associate ActionId with UniqueId earlier
when originating a call
https://issues.asterisk.org/view.php?id=17251

[patch] Autocreated peers not deleted when unregister explicitly, become zombies
https://issues.asterisk.org/view.php?id=16033

[patch] [OpenSolaris] wav format produces garbage files
https://issues.asterisk.org/view.php?id=16610

[patch] chan_iax2 ignores the port in an SRV record
https://issues.asterisk.org/view.php?id=17291

[patch] [branch] Implement standard XMPP Jingle in Asterisk
https://issues.asterisk.org/view.php?id=15634

[branch] gtalk web no incoming or outgoing calls
https://issues.asterisk.org/view.php?id=13971

[patch] JSON Manager Event Interface
https://issues.asterisk.org/view.php?id=14281

[patch] Originate Action output is inconsistent with other manager actions
https://issues.asterisk.org/view.php?id=17221

[branch] RTMP support in Asterisk
https://issues.asterisk.org/view.php?id=15484

[patch] Add ability to log CONGESTION calls to CDR
https://issues.asterisk.org/view.php?id=15907

[patch] Passing the mute flag to MeetMe() makes the new user have
"muted" himself, not an admin mute
https://issues.asterisk.org/view.php?id=15707

[patch] Patch that makes chan_sip check if the forward domain is
itself on a 302 response
https://issues.asterisk.org/view.php?id=15016

[patch] ENUMQUERY does not differentiate non-existant domain vs. no DNS records
https://issues.asterisk.org/view.php?id=14691

[patch] add support for circular searching for free devices in a group of phones
https://issues.asterisk.org/view.php?id=15582

[patch] Opening voice channel on FastStartAcknowledged before Answer.
Remove H245inSetupOptions for better capability.
https://issues.asterisk.org/view.php?id=15004

[patch] Add voicefile and dtmf options to res/res_agi.c
https://issues.asterisk.org/view.php?id=15531

[patch] MGCP Business Phone Packages patch
https://issues.asterisk.org/view.php?id=15159

[patch] chan_mgcp new feature: digitmaps definitions
https://issues.asterisk.org/view.php?id=16173

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com