Re: [asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock
On Wed, Apr 29, 2015 at 8:42 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hello asterisk-users,
We've been having intermittent issues with chan_sip - it stops responding to
cli requests, trying to reload chan_sip from cli doesn't seem to have any
effect, initiated calls carry on for a short period, but no new SIP requests
are processed ('sip show channels' hangs forever, server stops responding to
SIP OPTIONS, or any other SIP messages). We have updated the build from
1.8.23.1 to the latest asterisk 1.8 (1.8.32.3), however the problem still
persists. We have gathered debugging information from 'core show locks' and
from gdb, attached to this message (with phone numbers and extension and
context names obscured). We are running realtime under CentOS 6.6, built
from source and packaged using rpmbuild, with the following menuselect
options (debugging version):
menuselect/menuselect --disable BUILD_NATIVE --enable DEBUG_THREADS --enable
DONT_OPTIMIZE --disable CORE-SOUNDS-EN-GSM --disable-category
MENUSELECT_EXTRA_SOUNDS --disable MOH-OPSOUND-WAV --enable-category
MENUSELECT_ADDONS --disable format_mp3 --disable cdr_tds --disable cel_tds
--disable cdr_pgsql --disable cel_pgsql --disable res_config_pgsql
menuselect.makeopts
under kernel 2.6.32-504.el6.x86_64, and linked against the following library
versions:
/usr/lib64/libssl.so.10:symbolic link to `libssl.so.1.0.1e'
/usr/lib64/libcrypto.so.10: symbolic link to `libcrypto.so.1.0.1e'
/lib64/libc.so.6: symbolic link to `libc-2.12.so'
/usr/lib64/libxml2.so.2:symbolic link to `libxml2.so.2.7.6'
/lib64/libz.so.1: symbolic link to `libz.so.1.2.3'
/lib64/libm.so.6: symbolic link to `libm-2.12.so'
/lib64/libdl.so.2: symbolic link to `libdl-2.12.so'
/lib64/libpthread.so.0: symbolic link to `libpthread-2.12.so'
/lib64/libtinfo.so.5: symbolic link to `libtinfo.so.5.7'
/lib64/libresolv.so.2: symbolic link to `libresolv-2.12.so'
/lib64/libgssapi_krb5.so.2: symbolic link to `libgssapi_krb5.so.2.2'
/lib64/libkrb5.so.3:symbolic link to `libkrb5.so.3.3'
/lib64/libcom_err.so.2: symbolic link to `libcom_err.so.2.1'
/lib64/libk5crypto.so.3:symbolic link to `libk5crypto.so.3.1'
/lib64/libkrb5support.so.0: symbolic link to `libkrb5support.so.0.1'
/lib64/libkeyutils.so.1:symbolic link to `libkeyutils.so.1.3'
We'd appreciate any possible assistance, as we're having problems working
out what exactly triggers the deadlock and we have not been able to find the
correct sequence of steps to reproduce the issue yet, other than waiting for
it to lock up at an arbitrary time with the debugging code in place. It does
seem to happen at least once a day, however.
What is the best way of getting the core show locks output for people to see
as it appears to be too big to mail?
Please go ahead and make an issue on the issue tracker. Make sure you
get both the output of 'core show locks', as well as a GDB backtrace.
Instructions for both can be found here:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
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[asterisk-users] DPMA - Asterisk Realtime
We love our Digium phones and DPMA - but we really need it to work on our Realtime Platform. Otherwise we lose all the cool features and they are just standard SIP phones. Anyone working on a solution for this? Or anyone from Digium see this on the roadmap? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock
Ishfaq Malik wrote:
Hello asterisk-users,
We've been having intermittent issues with chan_sip - it stops
responding to cli requests, trying to reload chan_sip from cli doesn't
seem to have any effect, initiated calls carry on for a short period,
but no new SIP requests are processed ('sip show channels' hangs
forever, server stops responding to SIP OPTIONS, or any other SIP
messages). We have updated the build from 1.8.23.1 to the latest
asterisk 1.8 (1.8.32.3), however the problem still persists. We have
gathered debugging information from 'core show locks' and from gdb,
attached to this message (with phone numbers and extension and context
names obscured). We are running realtime under CentOS 6.6, built from
source and packaged using rpmbuild, with the following menuselect
options (debugging version):
menuselect/menuselect --disable BUILD_NATIVE --enable DEBUG_THREADS
--enable DONT_OPTIMIZE --disable CORE-SOUNDS-EN-GSM --disable-category
MENUSELECT_EXTRA_SOUNDS --disable MOH-OPSOUND-WAV --enable-category
MENUSELECT_ADDONS --disable format_mp3 --disable cdr_tds --disable
cel_tds --disable cdr_pgsql --disable cel_pgsql --disable
res_config_pgsql menuselect.makeopts
After looking at the issue this appears to be a duplicate of an existing
one[1] and a known issue.
[1] https://issues.asterisk.org/jira/browse/ASTERISK-21228
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com www.asterisk.org
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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Re: [asterisk-users] DPMA - Asterisk Realtime
On Friday 01 May 2015, Robert Broyles wrote: We love our Digium phones and DPMA - but we really need it to work on our Realtime Platform. Otherwise we lose all the cool features and they are just standard SIP phones. Anyone working on a solution for this? Or anyone from Digium see this on the roadmap? The best solution would be a compatible, GPL-ed replacement for DPMA, and I'm really surprised nobody has attempted this yet. We surely cannot be the only company out there for whom absence of Source Code is a deal-breaker? -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP - sessions-timers support not working on
Thanks for replying Joshua. Cheers, Javier Riveros -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
Le 01/05/2015 00:05, Andrew Martin a écrit : - Original Message - From: Administrator TOOTAI ad...@tootai.net To: asterisk-users@lists.digium.com Sent: Thursday, April 30, 2015 4:43:33 PM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G or other Yealink model) on 192.168.32.0/24 which have an OpenVPN client configured on them that connects back to the LAN network through a pfSense gateway with OpenVPN configured on it. I faced problems with pfsense -no VPN involved- and finally installed siproxd on it. Also set the firewall mode to conservative. Daniel, Thanks for the information. Do you have an example or documentation on the siproxd configuration that you used? No, just follow the basis of the parameters given by the package. If I remember, SIP use the proxy siproxd and RTP is direct. Another solution I used on an not stable xDSL line, was to install asterisk on pfsense, this asterisk taking only care on the local traffic (call from local extension to local extension). The asterisk register with the main one as a trunk for incoming/outgoing calls. Worked too. -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
