[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-12 Thread Launchpad Bug Tracker
This bug was fixed in the package asterisk - 1:13.1.0~dfsg-1.1ubuntu4.1

---
asterisk (1:13.1.0~dfsg-1.1ubuntu4.1) xenial; urgency=medium

  * debian/patches/fix_h264.patch: Backport fixes for H.264 support.
Thanks to Joshua Colp , Alexander Traud
, and Jörg Hänsel.  Closes LP: #1671767.

 -- Nishanth Aravamudan   Thu, 30 Mar
2017 16:43:30 -0700

** Changed in: asterisk (Ubuntu Xenial)
   Status: Fix Committed => Fix Released

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-12 Thread Robie Basak
Thank you for the detailed report on the testing you performed. This is
very much appreciated as it helps give us confidence about not
regression existing users by releasing the update.

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-05 Thread Jörg Hänsel
Hi Christian,
thank you for the info!

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-05 Thread ChristianEhrhardt
Jörg, you did perfectly right, thanks for your verification and also you
got the tag changed correctly.

It also has not caused any testing issues in [1].

According to the SRU process it will now sit in proposed for ~7 days - rarely 
less, sometimes more to give "the world" chance to realize if there is an issue.
You can see those in [2] and there is asterisk with currently 0 days and the 
bug in green/golden.

If nothing happens in approximately a week or so an SRU Team member will
accept it and then it will migrate from -proposed to -updates and be
available to everybody out there.

[1]: 
http://people.canonical.com/~ubuntu-archive/proposed-migration/xenial/update_excuses.html
[2]: http://people.canonical.com/~ubuntu-archive/pending-sru

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-04 Thread Jörg Hänsel
How can I change the tag from verification-needed to verification-done?

** Tags removed: verification-needed
** Tags added: verification-done

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-04 Thread Jörg Hänsel
Hello Łukasz,

I tested your asterisk 1:13.1.0~dfsg-1.1ubuntu4.1 package from the
proposed repository.

This packaged fixed the bug for me.

TEST CASE for 16.04 SRU:

1. use xenial
2. install asterisk
3. add SIP-clients to sip.conf and extensions.conf
4. dial h264 SIP client from asterisk console

Testing:

1. Check that system is up to date:
---
root@samson:~# apt-get update
Hit:1 http://de.archive.ubuntu.com/ubuntu xenial InRelease
Hit:2 http://de.archive.ubuntu.com/ubuntu xenial-updates InRelease
Hit:3 http://de.archive.ubuntu.com/ubuntu xenial-backports InRelease  
Get:4 http://security.ubuntu.com/ubuntu xenial-security InRelease [102 kB]
Fetched 102 kB in 0s (148 kB/s)
Reading package lists... Done
root@samson:~# apt-get upgrade
Reading package lists... Done
Building dependency tree 
Reading state information... Done
Calculating upgrade... Done
0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded.
root@samson:~#

2. Check that buggy package is installed:
-
root@samson:~# dpkg-query -l|grep asterisk
ii  asterisk1:13.1.0~dfsg-1.1ubuntu4
   amd64Open Source Private Branch Exchange (PBX)
ii  asterisk-config 1:13.1.0~dfsg-1.1ubuntu4
   all  Configuration files for Asterisk
ii  asterisk-core-sounds-en-gsm 1.4.22-1
   all  asterisk PBX sound files - en-us/gsm
ii  asterisk-modules1:13.1.0~dfsg-1.1ubuntu4
   amd64loadable modules for the Asterisk PBX
root@samson:~#

3. Test Case with buggy package:

root@samson:~# asterisk -rvvv
Asterisk 13.1.0~dfsg-1.1ubuntu4, Copyright (C) 1999 - 2014, Digium, Inc. and 
others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
Connected to Asterisk 13.1.0~dfsg-1.1ubuntu4 currently running on samson (pid = 
7163)
samson*CLI> console dial waldorf@Phones
-- Executing [waldorf@Phones:1] Dial("Console/default", "SIP/waldorf") in 
new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
-- Called SIP/waldorf
-- SIP/waldorf-0001 is ringing
samson*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
root@samson:~#

root@samson:~# ls -l /var/lib/asterisk/
insgesamt 32640
-rw-r--r-- 1 asterisk asterisk 3072 Apr  4 19:32 astdb.sqlite3
-rw--- 1 asterisk asterisk 98623488 Apr  4 19:36 core
drwxr-xr-x 2 asterisk asterisk 4096 Apr  5  2016 moh
drwxr-xr-x 3 asterisk asterisk 4096 Nov 16 10:15 sounds
-rw-r--r-- 1 asterisk asterisk 4096 Apr  4 19:48 sqlite.db
root@samson:~#

4.+5. Adding proposed repository:
-
root@samson:~# tail -1 /etc/apt/sources.list
deb http://archive.ubuntu.com/ubuntu/ xenial-proposed restricted main 
multiverse universe
root@samson:~# apt-get update
Get:1 http://security.ubuntu.com/ubuntu xenial-security InRelease [102 kB]
Get:2 http://archive.ubuntu.com/ubuntu xenial-proposed InRelease [253 kB]  
Hit:3 http://de.archive.ubuntu.com/ubuntu xenial InRelease  
  
Hit:4 http://de.archive.ubuntu.com/ubuntu xenial-updates InRelease  
  
Hit:5 http://de.archive.ubuntu.com/ubuntu xenial-backports InRelease
  
Get:6 http://archive.ubuntu.com/ubuntu xenial-proposed/restricted 
Translation-de [2752 B]
Get:7 http://archive.ubuntu.com/ubuntu xenial-proposed/main amd64 Packages [106 
kB]
Get:8 http://archive.ubuntu.com/ubuntu xenial-proposed/main i386 Packages [103 
kB]
Get:9 http://archive.ubuntu.com/ubuntu xenial-proposed/main Translation-en 
[39.3 kB]
Get:10 http://archive.ubuntu.com/ubuntu xenial-proposed/main Translation-de 
[501 kB]
Get:11 http://archive.ubuntu.com/ubuntu xenial-proposed/multiverse amd64 
Packages [2728 B]
Get:12 http://archive.ubuntu.com/ubuntu xenial-proposed/multiverse i386 
Packages [3056 B]
Get:13 http://archive.ubuntu.com/ubuntu xenial-proposed/multiverse 
Translation-en [1760 B]
Get:14 http://archive.ubuntu.com/ubuntu xenial-proposed/multiverse 
Translation-de [94.1 kB]
Get:15 http://archive.ubuntu.com/ubuntu xenial-proposed/universe amd64 Packages 
[47.5 kB]
Get:16 http://archive.ubuntu.com/ubuntu xenial-proposed/universe i386 Packages 
[45.5 kB]
Get:17 http://archive.ubuntu.com/ubuntu xenial-proposed/universe Translation-en 
[20.8 kB]
Get:18 http://archive.ubuntu.com/ubuntu xenial-prop

[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-04 Thread Łukasz Zemczak
Hello Jörg, or anyone else affected,

Accepted asterisk into xenial-proposed. The package will build now and
be available at
https://launchpad.net/ubuntu/+source/asterisk/1:13.1.0~dfsg-1.1ubuntu4.1
in a few hours, and then in the -proposed repository.

Please help us by testing this new package.  See
https://wiki.ubuntu.com/Testing/EnableProposed for documentation on how
to enable and use -proposed.Your feedback will aid us getting this
update out to other Ubuntu users.

If this package fixes the bug for you, please add a comment to this bug,
mentioning the version of the package you tested, and change the tag
from verification-needed to verification-done. If it does not fix the
bug for you, please add a comment stating that, and change the tag to
verification-failed.  In either case, details of your testing will help
us make a better decision.

Further information regarding the verification process can be found at
https://wiki.ubuntu.com/QATeam/PerformingSRUVerification .  Thank you in
advance!

** Changed in: asterisk (Ubuntu Xenial)
   Status: In Progress => Fix Committed

** Tags added: verification-needed

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Re: [Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-03 Thread Nish Aravamudan
On Mon, Apr 3, 2017 at 5:03 AM, Jörg Hänsel <1671...@bugs.launchpad.net> wrote:
> Can I do anything else to help to get the fixed package released?

As of right now, no. It's sitting in the unapproved queue waiting for
an SRU team member to process it:
https://launchpad.net/ubuntu/xenial/+queue?queue_state=1&queue_text=

It will get processed this week, hopefully.

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-04-03 Thread Jörg Hänsel
Can I do anything else to help to get the fixed package released?

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-31 Thread Jörg Hänsel
Thank you Nish!

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-30 Thread Nish Aravamudan
I have sponsored the change, although it needed some retooling to be
DEP3 compliant. I am attaching the debdiff I generated.

** Patch added: "asterisk.debdiff"
   
https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/1671767/+attachment/4850924/+files/asterisk.debdiff

** Changed in: asterisk (Ubuntu Xenial)
   Status: New => In Progress

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-30 Thread Nish Aravamudan
** Changed in: asterisk (Ubuntu)
   Status: Triaged => Fix Released

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-30 Thread Nish Aravamudan
** Also affects: asterisk (Ubuntu Xenial)
   Importance: Undecided
   Status: New

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-30 Thread Jörg Hänsel
Hi Christian,
can you tell me when or if the fixed version of asterisk will be released?

Thank you
Jörg

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Re: [Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-23 Thread ChristianEhrhardt
On Thu, Mar 23, 2017 at 1:51 PM, JörgHänsel <1671...@bugs.launchpad.net>
wrote:

> sorry, I am still not familar with the full the process. I used the
> "Subscribe someone else" link to subscribe "ubuntu-sponsors". Hope that was
> correct.


It was, now you are a bit more familiar than before.
thank you!

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-23 Thread JörgHänsel
Hi Christian,
sorry, I am still not familar with the full the process. I used the "Subscribe 
someone else" link to subscribe "ubuntu-sponsors". Hope that was correct.

Best regards,
Jörg

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-23 Thread ChristianEhrhardt
Hi Jörg,
I moved it up to the description and extended the regression potential.
It is always important to document that you (or I in that case) really thought 
about potential regressions and if/how they would manifest.

On point 3 - a TL;DR:
- What that does is get attention of people that are allowed to "ack" on this 
making its way into the release.
- First read link [2] in my post
- You'll realize that for this case this essentially boils down to just 
subscribing "ubuntu-sponsors" to this bug

Tomorrow is a special day [1] to sponsor a bunch of lost things, so
adding that today increases the chance to get things rather soon.

I can do that for you if that was yet unclear, but since this is also
about learning I'd encourage you to try.

[1]: https://lists.ubuntu.com/archives/ubuntu-devel/2017-March/039714.html
[2]: https://wiki.ubuntu.com/SponsorshipProcess

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-23 Thread ChristianEhrhardt
** Description changed:

+ [Impact]
+ 
+ when dialing a h264 video sip device (Grandstream GXV3674_FHD_VF 1.0.3.17) 
asterisk crashes with a core dump
+ 
+ [Test Case]
+ ===
+ 1. See comment #6
+ 
+ [Regression Potential]
+ ==
+ Since the patch is already included in more recent versions of asterisk there 
is no regression.
+ 
+ Due to the location of code changes that are applied to - If an
+ unexpected error manifests, it should be local to the h264 encoding
+ which is broken today.
+ 
+ [Other Info]
+ 
+ none
+ 
+ ---
+ 
  asterisk 1:13.1.0~dfsg-1.1ubuntu4
  
  lsb_release -rd:
  Description:  Ubuntu 16.04.2 LTS
  Release:  16.04
- 
  
  Bug details:
  
  when dialing a h264 video sip device (Grandstream GXV3674_FHD_VF 1.0.3.17) 
asterisk crashes with a core dump:
  
  Connected to Asterisk 13.1.0~dfsg-1.1ubuntu4 currently running on samson (pid 
= 29051)
  samson*CLI> console dial waldorf@Phones
- -- Executing [waldorf@Phones:1] Dial("Console/default", "SIP/waldorf,60") 
in new stack
-   == Using SIP VIDEO CoS mark 6
-   == Using SIP RTP CoS mark 5
- -- Called SIP/waldorf
- -- SIP/waldorf- is ringing
+ -- Executing [waldorf@Phones:1] Dial("Console/default", "SIP/waldorf,60") 
in new stack
+   == Using SIP VIDEO CoS mark 6
+   == Using SIP RTP CoS mark 5
+ -- Called SIP/waldorf
+ -- SIP/waldorf- is ringing
  samson*CLI>
  Disconnected from Asterisk server
  Asterisk cleanly ending (0).
  Executing last minute cleanups
  
- 
  Analysis:
  =
  gdb reveals that the module "res_format_attr_h264.so" is resposible due to a 
memory allocation failure while examining tokens of the "sprop-parameter-sets" 
string in the SIP header.
- 
  
  Proposed Solution:
  ==
  This bug is already fixed by 2 small patches included in a more recenent 
versions of "res/res_format_attr_h264.c"
  
  https://issues.asterisk.org/jira/browse/ASTERISK-24616
  Crash in res_format_attr_h264 due to invalid string copy
  
  https://issues.asterisk.org/jira/browse/ASTERISK-25573
  [patch] H.264 format attribute module: resets whole SDP
  
  This fixed version of "res/res_format_attr_h264.c" is included e.g. in
  asterisk (1:13.13.1~dfsg-4ubuntu1) zesty.
  
- 
  Fixed+Tested:
  =
  I tested by rebuiling asterisk 1:13.1.0~dfsg-1.1ubuntu4 packages with the 
"res/res_format_attr_h264.c" taken from asterisk 1:13.13.1~dfsg-4ubuntu1 and 
could connect to the sip device without any problems.
  
  I would be great, if you could add this patch into asterisk
  1:13.1.0~dfsg-1.1ubuntu4, since I don't want to use self built debs on a
  16.4 LTS production system.
  
- 
  Thanks a lot
  Jörg

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-23 Thread JörgHänsel
Hi Christian,

thank you for all the coordiniation and for preparing the test package!
I tested the package(s) yesterday evening on my test server successfully and 
filled out the SRU template with my test results (s.b.).

Could you please tell me, what exactly I have to a point 3 in your list?


Best regards
Jörg


[Impact]

when dialing a h264 video sip device (Grandstream GXV3674_FHD_VF 1.0.3.17) 
asterisk crashes with a core dump

[Test Case]
===
1. Asterisk configuration:

1.1. sip.conf:
[...]
videosupport=yes

[waldorf]
allow=h264
context=Phones
host=dynamic
secret=12345
type=friend
directmedia=no

1.2. extensions.conf:
[...]

[Phones]
exten => waldorf,1,Dial(SIP/${EXTEN},10)
 same => n,Hangup()


2.1. Reproducible crash with current version of asterisk (asterisk 
1:13.1.0~dfsg-1.1ubuntu4) in 16.4 LTS:

root@samson:~# dpkg-query -l|grep asterisk
ii  asterisk1:13.1.0~dfsg-1.1ubuntu4
   amd64Open Source Private Branch Exchange (PBX)
ii  asterisk-config 1:13.1.0~dfsg-1.1ubuntu4
   all  Configuration files for Asterisk
ii  asterisk-core-sounds-en-gsm 1.4.22-1
   all  asterisk PBX sound files - en-us/gsm
ii  asterisk-modules1:13.1.0~dfsg-1.1ubuntu4
   amd64loadable modules for the Asterisk PBX
root@samson:~# asterisk -rvvv
Asterisk 13.1.0~dfsg-1.1ubuntu4, Copyright (C) 1999 - 2014, Digium, Inc. and 
others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
Connected to Asterisk 13.1.0~dfsg-1.1ubuntu4 currently running on samson (pid = 
5866)
samson*CLI> console dial waldorf@Phones
-- Executing [waldorf@Phones:1] Dial("Console/default", "SIP/waldorf") in 
new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
-- Called SIP/waldorf
-- SIP/waldorf-0001 is ringing
samson*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups


2.2. No Crash after installing patched version (1:13.1.0~dfsg-1.1ubuntu4.1 ) 
(https://launchpad.net/~ci-train-ppa-service/+archive/ubuntu/2622):

root@samson:~# add-apt-repository ppa:ci-train-ppa-service/2622
[...]
root@samson:~# apt-get update
[...]
root@samson:~# apt-get upgrade
[...]
root@samson:~# dpkg-query -l|grep asterisk
ii  asterisk1:13.1.0~dfsg-1.1ubuntu4.1  
   amd64Open Source Private Branch Exchange (PBX)
ii  asterisk-config 1:13.1.0~dfsg-1.1ubuntu4.1  
   all  Configuration files for Asterisk
ii  asterisk-core-sounds-en-gsm 1.4.22-1
   all  asterisk PBX sound files - en-us/gsm
ii  asterisk-modules1:13.1.0~dfsg-1.1ubuntu4.1  
   amd64loadable modules for the Asterisk PBX
root@samson:~# systemctl restart asterisk
root@samson:~# asterisk -rvvv
Asterisk 13.1.0~dfsg-1.1ubuntu4.1, Copyright (C) 1999 - 2014, Digium, Inc. and 
others.
Created by Mark Spencer 
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=
Connected to Asterisk 13.1.0~dfsg-1.1ubuntu4.1 currently running on samson (pid 
= 13596)
samson*CLI> console dial waldorf@Phones
-- Executing [waldorf@Phones:1] Dial("Console/default", "SIP/waldorf") in 
new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
-- Called SIP/waldorf
-- SIP/waldorf-0004 is ringing
-- SIP/waldorf-0004 answered Console/default
  --- <("<) --- Call from Console has been Answered --- (>")> ---
-- Channel Console/default joined 'simple_bridge' basic-bridge 
<72502808-e34c-417d-abc0-59182eba454a>
[Mar 22 20:33:13] WARNING[14441][C-0003]: chan_console.c:649 
console_indicate: Don't know how to display condition 26 on Console/default
-- Channel SIP/waldorf-0004 joined 'simple_bridge' basic-bridge 
<72502808-e34c-417d-abc0-59182eba454a>
[Mar 22 20:33:13] WARNING[14446][C-0003]: channel.c:5070 ast_write: Codec 
mismatch on channel SIP/waldorf-0004 setting write format to slin from 
slin16 native formats (h264|alaw)
samson*CLI> console hangup
-- Channel Console/default left 'simple_bridge' basic-bridge 
<72502808-e34c-41

[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-22 Thread ChristianEhrhardt
Hi Jörg,
I picked this issue as the first starting on our Day.
I'm at a point where I think I could pass over to you.

I have done the prep of packaging the suggested fix.
If you are interested, Doc will be in our Ubuntu Server Bug Squashing Day 
Minutes I guess.
An unpolished version is at http://paste.ubuntu.com/24227424/ for you.

Eventually that means I prepared a fix in the following Bileto ppa:
https://launchpad.net/~ci-train-ppa-service/+archive/ubuntu/2622

The next steps I'd count on you if that is ok, you'd need to:
1. check the fix in the ppa if it works
2. add an SRU Template here (copy and modify [1]
3. following [2] then please subscribe ubuntu-sponsors as I can't upload myself 
- they can just publish from the bileto ticket [3] if they want

Update this bug for each step that you have taken, no matter if successful or 
failing so that we can help.
I'm available as cpaelzer on #ubuntu-server if you have questions, so are 
rbasak and nacc

[1]: https://wiki.ubuntu.com/StableReleaseUpdates#SRU_Bug_Template
[2]: https://wiki.ubuntu.com/SponsorshipProcess
[3]: https://bileto.ubuntu.com/#/ticket/2622

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  asterisk crashes dialing h264 video sip device

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-21 Thread Jörg
Hi Christian,
this sounds great!
It's a pleasure for me to support the community.

Thank you again,
Jörg

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  asterisk crashes dialing h264 video sip device

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-20 Thread ChristianEhrhardt
Hi Jörg,
thank you so much for your report and all the work already identifying the fix 
and all that!
That is a great help to make Ubuntu better.

Due to your good preparation this is great to be picked up soon IMO as
one finds a few minutes to prep all that is needed. We have an activity
on Wednesday trying to more actively include our great community in
issues just as such [1].

I consider this a great bug to be handled there and will mark it
accordingly to have a good chance to be picked up there.

[1]: https://lists.ubuntu.com/archives/ubuntu-
server/2017-March/007502.html

** Tags added: patch server-next

** Changed in: asterisk (Ubuntu)
   Status: New => Triaged

** Changed in: asterisk (Ubuntu)
   Importance: Undecided => High

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-17 Thread Jörg
still status "New" after one week?

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[Bug 1671767] Re: asterisk crashes dialing h264 video sip device

2017-03-10 Thread Jörg Hänsel
attached the patch generated with quilt in debian/patches in my testing
package

** Attachment added: "sprop-parameter-sets"
   
https://bugs.launchpad.net/ubuntu/+source/asterisk/+bug/1671767/+attachment/4835157/+files/sprop-parameter-sets

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  asterisk crashes dialing h264 video sip device

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