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[asterisk-users] Impossible to use any recent asterisk version with chan_sip


 
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ldardini at gmail.com
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PostPosted: Wed Jul 06, 2016 10:12 am    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

Hello,I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8  with chan_sip.


If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the asterisk log, but if I run "netstat -nap | grep 5060" I see the UDP buffer filled up.


If I step back to asterisk 13.2, then all is fine and asterisk is rock solid.


I know I should use PJSIP and chan_sip is no more supported, but at this point, if this is the working state of chan_sip, it should be completely removed.


Leandro
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jcolp at digium.com
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PostPosted: Wed Jul 06, 2016 10:14 am    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

Leandro Dardini wrote:
Quote:
Hello,
I'd like to know if anyone of you is finding my same problems using any
recent asterisk version, after 13.7 / 13.8 with chan_sip.

If I use any recent asterisk version, after just few seconds asterisk
completely locks up, stopping processing SIP/UDP packets. Nothing is
written in the asterisk log, but if I run "netstat -nap | grep 5060" I
see the UDP buffer filled up.

If I step back to asterisk 13.2, then all is fine and asterisk is rock
solid.

I know I should use PJSIP and chan_sip is no more supported, but at this
point, if this is the working state of chan_sip, it should be completely
removed.

It's not the normal working state of chan_sip, in fact it hasn't really
been touched in any way that would cause this to happen and I haven't
seen any bug reports along these lines. I also know that FreePBX is
running both chan_sip and chan_pjsip and haven't experienced anything so
it may be isolated to you. If you follow the instructions on the wiki[1]
it will provide a backtrace which will show where chan_sip is hanging.

[1]
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock

--
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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barryf-lists at flanag...
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PostPosted: Wed Jul 06, 2016 10:19 am    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

On 6 July 2016 at 16:14, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:


Quote:
Leandro Dardini wrote:
Quote:
Hello,
I'd like to know if anyone of you is finding my same problems using any
recent asterisk version, after 13.7 / 13.8  with chan_sip.

If I use any recent asterisk version, after just few seconds asterisk
completely locks up, stopping processing SIP/UDP packets. Nothing is
written in the asterisk log, but if I run "netstat -nap | grep 5060" I
see the UDP buffer filled up.

If I step back to asterisk 13.2, then all is fine and asterisk is rock
solid.

I know I should use PJSIP and chan_sip is no more supported, but at this
point, if this is the working state of chan_sip, it should be completely
removed.

It's not the normal working state of chan_sip, in fact it hasn't really been touched in any way that would cause this to happen and I haven't seen any bug reports along these lines. I also know that FreePBX is running both chan_sip and chan_pjsip and haven't experienced anything so it may be isolated to you. If you follow the instructions on the wiki[1] it will provide a backtrace which will show where chan_sip is hanging.



Joshua,


This was actually reported in https://issues.asterisk.org/jira/browse/ASTERISK-25468 with backtraces. It appeared to have started in 13.5, as I tested all from 13.2 to 13.6 at the time 


Hope this helps.


-Barry Flanagan




Quote:

--
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


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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              http://www.asterisk.org/hello

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jcolp at digium.com
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PostPosted: Wed Jul 06, 2016 10:22 am    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

Barry Flanagan wrote:

<snip>

Quote:

Joshua,

This was actually reported in
https://issues.asterisk.org/jira/browse/ASTERISK-25468 with backtraces.
It appeared to have started in 13.5, as I tested all from 13.2 to 13.6
at the time

Ah, that slipped my memory! Only the single person reporting it thus
far. If another backtrace can be added we can see if it's the same thing.

--
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


--
_____________________________________________________________________
-- 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
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jcolp at digium.com
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PostPosted: Wed Jul 06, 2016 2:47 pm    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

Leandro Dardini wrote:
Quote:
Hello,
I'd like to know if anyone of you is finding my same problems using any
recent asterisk version, after 13.7 / 13.8 with chan_sip.

If I use any recent asterisk version, after just few seconds asterisk
completely locks up, stopping processing SIP/UDP packets. Nothing is
written in the asterisk log, but if I run "netstat -nap | grep 5060" I
see the UDP buffer filled up.

If I step back to asterisk 13.2, then all is fine and asterisk is rock
solid.

I know I should use PJSIP and chan_sip is no more supported, but at this
point, if this is the working state of chan_sip, it should be completely
removed.

I've responded on the issue but the backtrace you've provided makes it
appear as though the issue is actually in ODBC, which since chan_sip is
using it in your deployment it causes it to lock up (why exactly is
unknown).

Since it's separate you should create a new issue. If you don't want to
I can do so tomorrow. The complete console output (with debug going to
console in logger.conf and core set debug 3) as well as the
configuration would also be useful.

Cheers,

--
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


--
_____________________________________________________________________
-- 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
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ldardini at gmail.com
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PostPosted: Wed Jul 06, 2016 3:49 pm    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info.

Leandro


2016-07-06 21:46 GMT+02:00 Joshua Colp <jcolp@digium.com (jcolp@digium.com)>:
Quote:
Leandro Dardini wrote:
Quote:
Hello,
I'd like to know if anyone of you is finding my same problems using any
recent asterisk version, after 13.7 / 13.8  with chan_sip.

If I use any recent asterisk version, after just few seconds asterisk
completely locks up, stopping processing SIP/UDP packets. Nothing is
written in the asterisk log, but if I run "netstat -nap | grep 5060" I
see the UDP buffer filled up.

If I step back to asterisk 13.2, then all is fine and asterisk is rock
solid.

I know I should use PJSIP and chan_sip is no more supported, but at this
point, if this is the working state of chan_sip, it should be completely
removed.

I've responded on the issue but the backtrace you've provided makes it appear as though the issue is actually in ODBC, which since chan_sip is using it in your deployment it causes it to lock up (why exactly is unknown).

Since it's separate you should create a new issue. If you don't want to I can do so tomorrow. The complete console output (with debug going to console in logger.conf and core set debug 3) as well as the configuration would also be useful.

Cheers,

--
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


--
_____________________________________________________________________
-- 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


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jcolp at digium.com
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PostPosted: Wed Jul 06, 2016 4:01 pm    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

Leandro Dardini wrote:
Quote:
This is a great news, thank you. I have open the issue,
https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the
relevant files, let me know if you need more info.

Nothing else is needed from you! All the information has shown it's
actually func_odbc. It's not using res_odbc like it should be, which
causes issues as a result of a fixup from some multi-connection
problems. We'll get it sorted.

--
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


--
_____________________________________________________________________
-- 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
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jcolp at digium.com
Guest





PostPosted: Wed Jul 06, 2016 4:10 pm    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

Leandro Dardini wrote:
Quote:
This is a great news, thank you. I have open the issue,
https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the
relevant files, let me know if you need more info.

And thank you for testing the release candidate so we can ensure the
issue is fixed before doing the actual release!

--
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


--
_____________________________________________________________________
-- 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
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ldardini at gmail.com
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PostPosted: Thu Jul 07, 2016 1:31 am    Post subject: [asterisk-users] Impossible to use any recent asterisk versi Reply with quote

No. I thank you for all the hard work done and dedication to the project.
Leandro Il 06/Lug/2016 11:10 PM, "Joshua Colp" <jcolp@digium.com (jcolp@digium.com)> ha scritto:
Quote:
Leandro Dardini wrote:
Quote:
This is a great news, thank you. I have open the issue,
https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the
relevant files, let me know if you need more info.

And thank you for testing the release candidate so we can ensure the issue is fixed before doing the actual release!

--
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


--
_____________________________________________________________________
-- 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
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