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Summary:ASTERISK-11689: problem with autoservice
Reporter:Dmitry Andrianov (dimas)Labels:
Date Opened:2008-03-20 11:40:24Date Closed:2008-04-09 15:18:58
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:The ast_autoservice_stop methods waits while channel list version (as_chan_list_state) changes before exiting. This is to make sure channel we are removing is not in the list which autoservice_run processes anymore.

he problem is that this wayt can last as long as 500 ms during which channel is not serviced at all. So on Zap channels voice frames are lost and DTMF detector misses digits.


****** ADDITIONAL INFORMATION ******

While ast_autoservice_stop methods even kills the autoservice_run to break it out of the poll call. The problem that if autoservice_run could not be in the poll yet.

Either the thread needs to be killed repeatedly in the wait loop or autoservice_run timeout needs to be changed from 500 ms to 50 ms. I did the last as the workaround.
Comments:By: Russell Bryant (russell) 2008-03-20 18:04:24

Hm, I guess changing the timeout is harmless.  However, this really should not ever happen unless there are channels in autoservice that are not generating any frames, which would be quite odd ...
By: Digium Subversion (svnbot) 2008-03-20 18:09:46

Repository: asterisk
Revision: 110395

U   branches/1.4/main/autoservice.c

------------------------------------------------------------------------
r110395 | russell | 2008-03-20 18:09:45 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue ASTERISK-11689, reported by dimas)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=110395
By: Digium Subversion (svnbot) 2008-03-20 18:10:07

Repository: asterisk
Revision: 110396

_U  trunk/
U   trunk/main/autoservice.c

------------------------------------------------------------------------
r110396 | russell | 2008-03-20 18:10:06 -0500 (Thu, 20 Mar 2008) | 17 lines

Merged revisions 110395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue ASTERISK-11689, reported by dimas)

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=110396
By: Digium Subversion (svnbot) 2008-03-20 18:10:10

Repository: asterisk
Revision: 110397

_U  branches/1.6.0/
U   branches/1.6.0/main/autoservice.c

------------------------------------------------------------------------
r110397 | russell | 2008-03-20 18:10:09 -0500 (Thu, 20 Mar 2008) | 25 lines

Merged revisions 110396 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008) | 17 lines

Merged revisions 110395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue ASTERISK-11689, reported by dimas)

........

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=110397
By: Digium Subversion (svnbot) 2008-04-09 15:15:44

Repository: asterisk
Revision: 113924

U   team/group/NoLossCDR-Redux2/build_tools/cflags.xml
U   team/group/NoLossCDR-Redux2/pbx/pbx_ael.c
U   team/group/NoLossCDR-Redux2/phoneprov/000000000000-directory.xml
U   team/group/NoLossCDR-Redux2/phoneprov/polycom.xml
A   team/group/NoLossCDR-Redux2/phoneprov/polycom_line.xml

------------------------------------------------------------------------
r113924 | juggie | 2008-04-09 15:15:41 -0500 (Wed, 09 Apr 2008) | 211 lines

Merged revisions 110020,110023,110036,110084,110087,110132,110161,110164,110211,110237,110268,110270,110272,110303,110337,110339,110396,110444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r110020 | file | 2008-03-19 14:25:33 -0400 (Wed, 19 Mar 2008) | 14 lines

Merged revisions 110019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines

Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue ASTERISK-10940)
Reported by: sperreault
Patches:
     11429-frametype.diff uploaded by qwell (license 4)

........

................
r110023 | russell | 2008-03-19 14:57:16 -0400 (Wed, 19 Mar 2008) | 2 lines

remove svnmerge-blocked property that is not supposed to be here

................
r110036 | file | 2008-03-19 15:13:39 -0400 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines

Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue ASTERISK-11136)
Reported by: junky

........

................
r110084 | mmichelson | 2008-03-19 16:34:13 -0400 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines

Add a missing unlock in the case that memory allocation fails in app_chanspy.
Thanks to Russell for confirming that this was an issue.


........

................
r110087 | jpeeler | 2008-03-19 17:05:24 -0400 (Wed, 19 Mar 2008) | 2 lines

This change adds DNS manager support for registrations not referencing a peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries.

................
r110132 | qwell | 2008-03-19 17:56:15 -0400 (Wed, 19 Mar 2008) | 1 line

Rename very poorly named function to reflect what it actually does.  This was causing quite a bit of confusion for me...
................
r110161 | qwell | 2008-03-19 18:25:34 -0400 (Wed, 19 Mar 2008) | 5 lines

Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading.

Early cleanup for issue ASTERISK-11413

................
r110164 | russell | 2008-03-19 18:58:33 -0400 (Wed, 19 Mar 2008) | 13 lines

Merged revisions 110163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines

Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected.

(closes issue ASTERISK-11434, reported by anakaoka, patched by me)

........

................
r110211 | tilghman | 2008-03-19 23:14:59 -0400 (Wed, 19 Mar 2008) | 2 lines

Fix recent trunk breakage

................
r110237 | tilghman | 2008-03-20 01:06:12 -0400 (Thu, 20 Mar 2008) | 5 lines

Upgrade the sounds version; add several directory enhancements:
1) Number of digits to enter can now be configured
2) The digits can now match on both first AND last name, instead of only one or the other
(Closes issue ASTERISK-6965)

................
r110268 | russell | 2008-03-20 13:41:22 -0400 (Thu, 20 Mar 2008) | 27 lines

Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.

(issue ASTERISK-11594, reported by milazzo and jsmith, patches by me)

res/res_musiconhold.c:
- I moved a single line so that the sample queue update happened before
  ast_write().  The reason that this was a bug is that the G.722 frame
  originally says it has 320 samples in it (which is correct).  However,
  when the frame is written to a channel that uses RTP, main/rtp.c modifies
  the frame to cut the number of samples in half before it sends it on
  the wire.  This is to account for the stupid incorrect G.722 spec that
  makes it so we have to lie about the number of samples with RTP.  I should
  probably go and re-work the RTP code so it doesn't modify the frame so
  that a bug like this won't happen in the future.  However, this change to
  MOH is harmless.

main/channel.c:
- I made two fixes in regards to generator timing.  Generators use samples
  for timing.  However, this code assumed 8 kHz samples.  In one case, it was
  a hard coded 160 samples, that is now written as the sample rate / 50.  The
  other place was dealing with timing a generator based on frames coming from
  the other direction.  However, that would have only worked if the sample
  rates for the formats in both directions were the same.  The code now takes
  into account that the sample rates may differ, and scales the generator
  samples accordingly.

................
r110270 | russell | 2008-03-20 13:45:29 -0400 (Thu, 20 Mar 2008) | 2 lines

Remove astobj.h from some places where it wasn't needed

................
r110272 | mmichelson | 2008-03-20 14:01:36 -0400 (Thu, 20 Mar 2008) | 3 lines

Add missing unlock


................
r110303 | russell | 2008-03-20 16:08:26 -0400 (Thu, 20 Mar 2008) | 8 lines

Fix a bug when using zaptel timing for playing back files that have a sample rate
other than 8 kHz.  The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame.  However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.

(another part of issue ASTERISK-11594, reported by milazzo and jsmith, patch by me)

................
r110337 | russell | 2008-03-20 17:55:50 -0400 (Thu, 20 Mar 2008) | 22 lines

Merged revisions 110336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines

Merged revisions 110335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines

Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix.  The dnsmgr is not appropriate here.  The dnsmgr takes a pointer to an address
structure that a background thread continuously updates.  However, in these cases,
a stack variable was passed.  That means that the dnsmgr thread would be continuously
writing to bogus memory.

........

................

................
r110339 | russell | 2008-03-20 18:02:20 -0400 (Thu, 20 Mar 2008) | 3 lines

Use the correct buffer for g722tolin16_sample.  This shouldn't have caused any
problems, but Qwell noticed the typo here.

................
r110396 | russell | 2008-03-20 19:14:13 -0400 (Thu, 20 Mar 2008) | 17 lines

Merged revisions 110395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue ASTERISK-11689, reported by dimas)

........

................
r110444 | tilghman | 2008-03-20 21:44:38 -0400 (Thu, 20 Mar 2008) | 2 lines

Add note of the added Directory options, from commit 110237 (closes issue ASTERISK-6965)

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=113924
By: Digium Subversion (svnbot) 2008-04-09 15:18:58

Repository: asterisk
Revision: 113925

U   team/group/NoLossCDR-Redux2/channels/chan_h323.c
U   team/group/NoLossCDR-Redux2/channels/chan_usbradio.c
U   team/group/NoLossCDR-Redux2/channels/misdn_config.c

------------------------------------------------------------------------
r113925 | juggie | 2008-04-09 15:18:55 -0500 (Wed, 09 Apr 2008) | 211 lines

Merged revisions 110020,110023,110036,110084,110087,110132,110161,110164,110211,110237,110268,110270,110272,110303,110337,110339,110396,110444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r110020 | file | 2008-03-19 14:25:33 -0400 (Wed, 19 Mar 2008) | 14 lines

Merged revisions 110019 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines

Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue ASTERISK-10940)
Reported by: sperreault
Patches:
     11429-frametype.diff uploaded by qwell (license 4)

........

................
r110023 | russell | 2008-03-19 14:57:16 -0400 (Wed, 19 Mar 2008) | 2 lines

remove svnmerge-blocked property that is not supposed to be here

................
r110036 | file | 2008-03-19 15:13:39 -0400 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines

Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior.
(closes issue ASTERISK-11136)
Reported by: junky

........

................
r110084 | mmichelson | 2008-03-19 16:34:13 -0400 (Wed, 19 Mar 2008) | 12 lines

Merged revisions 110083 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines

Add a missing unlock in the case that memory allocation fails in app_chanspy.
Thanks to Russell for confirming that this was an issue.


........

................
r110087 | jpeeler | 2008-03-19 17:05:24 -0400 (Wed, 19 Mar 2008) | 2 lines

This change adds DNS manager support for registrations not referencing a peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries.

................
r110132 | qwell | 2008-03-19 17:56:15 -0400 (Wed, 19 Mar 2008) | 1 line

Rename very poorly named function to reflect what it actually does.  This was causing quite a bit of confusion for me...
................
r110161 | qwell | 2008-03-19 18:25:34 -0400 (Wed, 19 Mar 2008) | 5 lines

Rename DSP_FEATURE_DTMF_DETECT, because we are *NOT* only detecting DTMF digits.
This was very misleading.

Early cleanup for issue ASTERISK-11413

................
r110164 | russell | 2008-03-19 18:58:33 -0400 (Wed, 19 Mar 2008) | 13 lines

Merged revisions 110163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines

Fix a bug where when calls on the trunk side hang up while on hold, the state
is not properly reflected.

(closes issue ASTERISK-11434, reported by anakaoka, patched by me)

........

................
r110211 | tilghman | 2008-03-19 23:14:59 -0400 (Wed, 19 Mar 2008) | 2 lines

Fix recent trunk breakage

................
r110237 | tilghman | 2008-03-20 01:06:12 -0400 (Thu, 20 Mar 2008) | 5 lines

Upgrade the sounds version; add several directory enhancements:
1) Number of digits to enter can now be configured
2) The digits can now match on both first AND last name, instead of only one or the other
(Closes issue ASTERISK-6965)

................
r110268 | russell | 2008-03-20 13:41:22 -0400 (Thu, 20 Mar 2008) | 27 lines

Add some fixes that I made in regards to wideband codec handling to get
G.722 music on hold working for me.

(issue ASTERISK-11594, reported by milazzo and jsmith, patches by me)

res/res_musiconhold.c:
- I moved a single line so that the sample queue update happened before
  ast_write().  The reason that this was a bug is that the G.722 frame
  originally says it has 320 samples in it (which is correct).  However,
  when the frame is written to a channel that uses RTP, main/rtp.c modifies
  the frame to cut the number of samples in half before it sends it on
  the wire.  This is to account for the stupid incorrect G.722 spec that
  makes it so we have to lie about the number of samples with RTP.  I should
  probably go and re-work the RTP code so it doesn't modify the frame so
  that a bug like this won't happen in the future.  However, this change to
  MOH is harmless.

main/channel.c:
- I made two fixes in regards to generator timing.  Generators use samples
  for timing.  However, this code assumed 8 kHz samples.  In one case, it was
  a hard coded 160 samples, that is now written as the sample rate / 50.  The
  other place was dealing with timing a generator based on frames coming from
  the other direction.  However, that would have only worked if the sample
  rates for the formats in both directions were the same.  The code now takes
  into account that the sample rates may differ, and scales the generator
  samples accordingly.

................
r110270 | russell | 2008-03-20 13:45:29 -0400 (Thu, 20 Mar 2008) | 2 lines

Remove astobj.h from some places where it wasn't needed

................
r110272 | mmichelson | 2008-03-20 14:01:36 -0400 (Thu, 20 Mar 2008) | 3 lines

Add missing unlock


................
r110303 | russell | 2008-03-20 16:08:26 -0400 (Thu, 20 Mar 2008) | 8 lines

Fix a bug when using zaptel timing for playing back files that have a sample rate
other than 8 kHz.  The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame.  However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.

(another part of issue ASTERISK-11594, reported by milazzo and jsmith, patch by me)

................
r110337 | russell | 2008-03-20 17:55:50 -0400 (Thu, 20 Mar 2008) | 22 lines

Merged revisions 110336 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

................
r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines

Merged revisions 110335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

........
r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines

Fix some very broken code that was introduced in 1.2.26 as a part of the security
fix.  The dnsmgr is not appropriate here.  The dnsmgr takes a pointer to an address
structure that a background thread continuously updates.  However, in these cases,
a stack variable was passed.  That means that the dnsmgr thread would be continuously
writing to bogus memory.

........

................

................
r110339 | russell | 2008-03-20 18:02:20 -0400 (Thu, 20 Mar 2008) | 3 lines

Use the correct buffer for g722tolin16_sample.  This shouldn't have caused any
problems, but Qwell noticed the typo here.

................
r110396 | russell | 2008-03-20 19:14:13 -0400 (Thu, 20 Mar 2008) | 17 lines

Merged revisions 110395 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines

Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread.
This really should not make a difference except in very rare cases.  That case would
be that all of the channels in autoservice are not generating any frames.  In that
case, this change reduces the potential amount of time that a thread waits in
ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning
of its loop.

(closes issue ASTERISK-11689, reported by dimas)

........

................
r110444 | tilghman | 2008-03-20 21:44:38 -0400 (Thu, 20 Mar 2008) | 2 lines

Add note of the added Directory options, from commit 110237 (closes issue ASTERISK-6965)

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=113925