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Summary:ASTERISK-11571: During page no audio can be heard from Polycom or Aastra phones, Snom phones are OK
Reporter:David Brillert (aragon)Labels:
Date Opened:2008-03-04 15:56:50.000-0600Date Closed:2008-03-11 13:46:19
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Applications/app_page
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) pagingdebug6003to6004.txt
Description:I've made pages thousands of times on various sites.
I recently upgraded to Asterisk 1.4 SVN as of Friday February 29th because my customer was having segfault problems and I applied 1.4 branch to fix bug ID 12038
Paging was working until my upgrade.
Since the upgrade I can no longer make any pages to Aastra or Polycom phones.
Pages sent to Snom phones work fine.
The Aastra phone answers the page and for about a half second I can hear some audio but then the audio is gone.
The incoming page only disconnects when I hangup as is expected

I have attached a debug file

I can easily provide remote access to my deployment server for assistance.

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

In my test 6003 dials *830 to make the page
For simplicity only 6004 is in the paging group.
6004/6004         192.168.30.178 Aastra 480i v 1.4.2.3000  
6003/6003         192.168.30.199 Snom 370 v 7.1.30
Comments:By: Mark Michelson (mmichelson) 2008-03-04 19:38:47.000-0600

I was able to reproduce this by attempting to page Local channels. The problem does not appear to show itself when paging SIP channels. From the debug output, the line that stands out most to me is:

[Mar  4 16:22:57] DEBUG[3766] channel.c: Got a FRAME_CONTROL (-1) frame on channel Local/6004@default-local-paging-ff51,2

Can you tell what the latest revision you were using when paging Local channels worked? If we can get a base revision to start at, we might be able to pinpoint which revision caused the break. Just as a sanity check, I locally reverted the fix for 12038 and that did not fix this problem.

Note: I normally would put this into feedback mode since I am requesting information, but for some reason Mantis doesn't seem to want to allow me to do that. I apologize for that.
By: David Brillert (aragon) 2008-03-04 21:03:16.000-0600

Can you tell what the latest revision you were using when paging Local channels worked?

Asterisk 1.4.18 official release worked OK
By: David Brillert (aragon) 2008-03-04 21:11:22.000-0600

I realize 1.4.18 does not help pinpoint the problem much.
I will go through some history to try and find the exact revision
By: David Brillert (aragon) 2008-03-05 08:46:51.000-0600

The latest known working version for me is Asterisk 1.4.18 official
the broken SVN version I am using is 104868
By: David Brillert (aragon) 2008-03-05 08:50:38.000-0600

Also tested with latest SVN this morning and still broken
By: Joel Vandal (jvandal) 2008-03-05 09:06:09.000-0600

The problem look to be related to patch on ticket ASTERISK-11240

If we revert the change on app_meetme, then Page work.
By: Joel Vandal (jvandal) 2008-03-05 09:08:50.000-0600

doh... error ! related to 11835


If entering a conference with the 'w' option ensure that we can't listen or speak until the marked user appears.
(closes issue ASTERISK-11296)
By: David Brillert (aragon) 2008-03-05 09:15:30.000-0600

Yes if we update to today's SVN and revert patch for 11835 then paging works
By: David Brillert (aragon) 2008-03-06 15:31:58.000-0600

Hi file

Any news?
We can fix the bad behavior by using SVN and removing patch 11835
I noticed 1.4.19rc1 was released but I saw nothing related to fixing this issue.
Of course I'd like to see some fix in time for 1.4.19 official

Cheers
By: Joel Vandal (jvandal) 2008-03-11 12:50:13

Any news about this ticket ?

We use Asterisk SVN (branches/1.4) with a revert of 11835 and no more problems with Paging audio.

IMHO, you must revert patch 11835 from SVN, re-open 11835, add a relationship and  close this ticket.
By: Digium Subversion (svnbot) 2008-03-11 13:43:52

Repository: asterisk
Revision: 107637

U   branches/1.4/apps/app_meetme.c

------------------------------------------------------------------------
r107637 | file | 2008-03-11 13:43:46 -0500 (Tue, 11 Mar 2008) | 4 lines

Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue ASTERISK-11571)
Reported by: aragon

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

http://svn.digium.com/view/asterisk?view=rev&revision=107637
By: Digium Subversion (svnbot) 2008-03-11 13:45:04

Repository: asterisk
Revision: 107638

_U  trunk/
U   trunk/apps/app_meetme.c

------------------------------------------------------------------------
r107638 | file | 2008-03-11 13:45:02 -0500 (Tue, 11 Mar 2008) | 12 lines

Merged revisions 107637 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines

Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue ASTERISK-11571)
Reported by: aragon

........

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

http://svn.digium.com/view/asterisk?view=rev&revision=107638
By: Digium Subversion (svnbot) 2008-03-11 13:46:19

Repository: asterisk
Revision: 107639

_U  branches/1.6.0/
U   branches/1.6.0/apps/app_meetme.c

------------------------------------------------------------------------
r107639 | file | 2008-03-11 13:46:18 -0500 (Tue, 11 Mar 2008) | 20 lines

Merged revisions 107638 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) | 12 lines

Merged revisions 107637 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines

Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened.
(closes issue ASTERISK-11571)
Reported by: aragon

........

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

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

http://svn.digium.com/view/asterisk?view=rev&revision=107639