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 <marks...@digium.com>
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 <marks...@digium.com>
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
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https://bugs.launchpad.net/bugs/1671767
Title:
asterisk crashes dialing h264 video sip device
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