[asterisk-bugs] [Asterisk 0017232]: Extensions with playback stalls in different points in bridged calls
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Apr 23 13:04:29 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17232
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Reported By: EduFrazao
Assigned To:
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Project: Asterisk
Issue ID: 17232
Category: Applications/app_playback
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: 1.6.2.6
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2010-04-22 16:15 CDT
Last Modified: 2010-04-23 13:04 CDT
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Summary: Extensions with playback stalls in different points
in bridged calls
Description:
Hi all.
Im tryng to connect Two Asterisk Boxes, via IAX, both with version
1.6.2.6, or any 1.6.2.x ( same result ).
This is the scenario: SIP Phone -> Local Asterisk (IAX) Remote Asterisk ->
Sip Phone.
All works fine with the calls, when it terminates on a remote SIP Phone.
But, When I try to use any application on the remote server Like a
Voicemail, or Meetme, o even a simple extension that uses some Playback
statements, the dialplan stalls before few playbacks.
Absolutelly nothing is reported on the CLI, even with verbose and debug in
999.
Anyway, ive made this tests:
1) Downgrade Asterisk to 1.6.1.18, and run it, with exactly same
configuration files. For my surprise, all works, perfecly!
2) Build a VPN, and connects a SIP Phone, directly to the remote server,
and dial the test extension, made by some Playback statements. It works
nice!
To make more tests, I upgraded the local server to 1.6.2.6 again, and call
the playback tests again, on the remote server, that is using 1.6.1.18.
Works too. If I start a call from 1.6.1.18 to 1.6.2.6, the problem occours.
It means, that the problem is with the playback on the bridged call, from
1.6.2.6 box.
Ive tried to connect the servers with SIP, and test if this problem is
only with the IAX trunk, but the problem persists even with SIP, so, ive
back to my IAX Configuration.
About my System:
Two boxes, on a Dual XEON E5410, with kernel 2.6.31, 2GB RAM, and GCC
4.3.4.
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(0120840) pabelanger (manager) - 2010-04-23 13:04
https://issues.asterisk.org/view.php?id=17232#c120840
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More information about the issue. The dialplan seems to stop at random
times. (After digit 6 in the attached debug logs).
---
<Peste_Bubonica> pabelanger, the sound is being with poor qualit, so, all
stalls
<pabelanger> Peste_Bubonica: timestamp? or which digits are affected?
<Peste_Bubonica> pabelanger, normally the digits are affected...
<Peste_Bubonica> parts of the playbacks are lost, and so, stall
<Peste_Bubonica> pabelanger, with the 1.6.1.18 version, the sound quality
is impressive better, and this problem does not occour
<Peste_Bubonica> sorry by bad english
<pabelanger> Peste_Bubonica: Yes, I understand that. I'm looking at the
debug log now, but need you either the time of the audio quality or between
which digits (1234567890) is the problem.
<pabelanger> IE: [Apr 23 12:37:03]
<Peste_Bubonica> is varying...
<Peste_Bubonica> sometimes, all digits pass, the stalls occour on another
playback, like exten => 302,n,Playback(pls-try-call-later)
<Peste_Bubonica> some times even complete the extension ( very rare )
<pabelanger> And when you stay 'stalls' you hear
1,2,3...<silence>...4,5,6,7,8,9 or 1,2,3...<silence>...8,9
<pabelanger> s/stay/say/g
<infobot> pabelanger meant: And when you say 'stalls' you hear
1,2,3...<silence>...4,5,6,7,8,9 or 1,2,3...<silence>...8,9
<Peste_Bubonica> stalls = stop...
<Peste_Bubonica> the dial stops completly..
<Peste_Bubonica> the dialplan...
<Peste_Bubonica> breaks in the middle of the statements of the
extension... I need to hangup the call
<Peste_Bubonica> like
<Peste_Bubonica> 1,2,3,(some distorcion),4, all stops..
<Peste_Bubonica> with no error on the cli or logs
<pabelanger> and if it stops, how do you restart it? Will it restart if
you don't hangup?
<Peste_Bubonica> pabelanger, http://pastebin.com/izjytsnV
<Peste_Bubonica> pabelanger, I hagup manually, and call again
<Peste_Bubonica> it does not restart automatically
<Peste_Bubonica> stays stoped for many time as needed until hangup
<Peste_Bubonica> i never seen nothing like that
<pabelanger> any output from 'core show channels'?
<pabelanger> after you hangup
<Peste_Bubonica> let me see
<Peste_Bubonica> pabelanger, http://pastebin.com/nhPQ49LS
<Peste_Bubonica> all zero
<pabelanger> how hard to reproduce this? Make another call and this time
when the playback stops, from the CLI run 'core show locks'.
<Peste_Bubonica> is extremelly easy to reproduce... amolst all the
time...
<Peste_Bubonica> pabelanger, core show locks, whit the extension stalled:
http://pastebin.com/CZXzxckL
<pabelanger> don't hangup
<pabelanger> if you dial the context from a local extension on this
asterisk box, does the stall happen?
<pabelanger> Trying to see if it is related to IAX
<Peste_Bubonica> pabelanger, if I call from a phone directly on the box,
works nice
<Peste_Bubonica> pabelanger, anyway, I make the test with SIP two... Ive
conected the two boxes togheter via SIP
<Peste_Bubonica> the problem persist
<Peste_Bubonica> only when the call is bridged
Issue History
Date Modified Username Field Change
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2010-04-23 13:04 pabelanger Note Added: 0120840
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