Comment 6 for bug 1671767

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 asterisk 1:13.1.0~dfsg-1.1ubuntu4 amd64 Open 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-modules 1:13.1.0~dfsg-1.1ubuntu4 amd64 loadable 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 <email address hidden>
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-00000001 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 asterisk 1:13.1.0~dfsg-1.1ubuntu4.1 amd64 Open 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-modules 1:13.1.0~dfsg-1.1ubuntu4.1 amd64 loadable 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 <email address hidden>
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-00000004 is ringing
    -- SIP/waldorf-00000004 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-00000003]: chan_console.c:649 console_indicate: Don't know how to display condition 26 on Console/default
    -- Channel SIP/waldorf-00000004 joined 'simple_bridge' basic-bridge <72502808-e34c-417d-abc0-59182eba454a>
[Mar 22 20:33:13] WARNING[14446][C-00000003]: channel.c:5070 ast_write: Codec mismatch on channel SIP/waldorf-00000004 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-417d-abc0-59182eba454a>
  == Spawn extension (Phones, waldorf, 1) exited non-zero on 'Console/default'
  --- <("<) --- Hangup on Console --- (>")> ---
    -- Channel SIP/waldorf-00000004 left 'simple_bridge' basic-bridge <72502808-e34c-417d-abc0-59182eba454a>
samson*CLI>

no crash anymore!

[Regression Potential]
======================
Since the patch is already included in more recent versions of asterisk there is no regression.

[Other Info]
============
none