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Summary:ASTERISK-09843: Incorrect notify handling when hints contain multiple devices
Reporter:Nic Bellamy (nic_bellamy)Labels:
Date Opened:2007-07-09 21:48:36Date Closed:2007-11-09 10:34:34.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Subscriptions
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip-hinting-svn-branch-1.2.patch
( 1) sip-hinting-svn-branch-1.4.patch
Description:There's a logic flaw in transmit_state_notify():

When sending a state notification, the hints for the extension are checked to see if the device is offline; if so it overrides the state notification to say "not online".

The problem: if your hint contains multiple devices, _any_ of these being offline will override the state. The state should only be overridden if _all_ the hinted devices are offline.

This breaks BLFs for Polycom devices (and anything else using pidf/xpidf notifications) when you have multiple devices hinted.

This was introduced by the fix for ASTERISK-9259.

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

Patch attached.
Comments:By: Nic Bellamy (nic_bellamy) 2007-07-09 21:54:51

Well, ok - patch _would_ be attached, but it's asking me to sign the license agreement. There is already one on file under Vadacom Ltd. Signing again requires going through all the corporate procedures, at which point your name will be mud with my CEO - "What, they *lost* the agreement!?"

I'll happy upload the patch when this is fixed.


By: Russell Bryant (russell) 2007-07-09 22:18:01

We have a new system in place.  New uploads require agreeing to the license through the web site.  If you have any questions or concerns, feel free to talk to the person at Digium in charge of legal matters - Michelle Petrone (mpetrone@digium.com).
By: Nic Bellamy (nic_bellamy) 2007-07-10 22:39:07

Hi Russell,

I am aware of the new system - and think it's a great idea for those who don't already have a disclaimer - makes things easier for both sides.

I work for a company. As in most companies, we're not in the business of giving extremely liberal (in favor of Digium) copyright licenses away for free, so getting the original disclaimer done was quite an involved process, involving CEO, legal, etc. This took me about two weeks to get organized and signed-off on the first time around.

Suggesting I go through this entire process again is just ludicrous - I'm trying to _give you code_!
By: Jörn Frenzel (frenzel) 2007-07-19 11:05:24

Hi Russel, hi Nic,

did you aleready master the issue about the license? I´m really hungry for the patch written by Nic. I´ve got the same problem here with some "snom 300" and multiple hints to them. It would be a great favour for me!

btw: Does this bug still exist in asterisk version 1.4?

Thanks for your work!


By: Nic Bellamy (nic_bellamy) 2007-07-19 15:58:13

Nearly there with the license - discussed it with KPF @ Digium and my management, and there some actual changes in the v3.0 license that make legal sense. Going through the process of getting it approved internally at the moment. I was hoping to have it sorted in time to get the patch into 1.2.22, but that got fast tracked due to a number of security issues.

And yes, from reading the code, the bug does also affect 1.4 and trunk.
By: Joshua C. Colp (jcolp) 2007-10-29 11:53:02

Hola! Did you get the licensing situation sorted out?
By: Nic Bellamy (nic_bellamy) 2007-10-29 18:09:14

Thanks for the reminder - I just prodded my boss again, and have just gotten approval, so I've submitted my request.

Awaiting confirmation that it's accepted.
By: Nic Bellamy (nic_bellamy) 2007-11-01 15:37:38

License accepted, and a patch for the latest 1.2 and 1.4 branches attached.


By: Digium Subversion (svnbot) 2007-11-07 19:27:00.000-0600

Repository: asterisk
Revision: 89099

U   branches/1.4/channels/chan_sip.c

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r89099 | file | 2007-11-07 19:26:59 -0600 (Wed, 07 Nov 2007) | 6 lines

Improve the devicestate logic for multiple devices. If any are available then the extension is considered available.
(closes issue ASTERISK-9843)
Reported by: nic_bellamy
Patches:
     sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)

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By: Digium Subversion (svnbot) 2007-11-07 19:28:32.000-0600

Repository: asterisk
Revision: 89100

_U  trunk/
U   trunk/channels/chan_sip.c

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r89100 | file | 2007-11-07 19:28:32 -0600 (Wed, 07 Nov 2007) | 14 lines

Merged revisions 89099 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines

Improve the devicestate logic for multiple devices. If any are available then the extension is considered available.
(closes issue ASTERISK-9843)
Reported by: nic_bellamy
Patches:
     sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)

........

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By: Digium Subversion (svnbot) 2007-11-09 10:34:34.000-0600

Repository: asterisk
Revision: 89131

_U  team/file/t38fun/
U   team/file/t38fun/apps/app_queue.c
U   team/file/t38fun/apps/app_voicemail.c
U   team/file/t38fun/cdr/cdr_tds.c
U   team/file/t38fun/channels/chan_sip.c
U   team/file/t38fun/codecs/codec_zap.c
U   team/file/t38fun/configs/extensions.ael.sample
U   team/file/t38fun/configs/res_odbc.conf.sample
U   team/file/t38fun/doc/valgrind.txt
U   team/file/t38fun/include/asterisk/lock.h
U   team/file/t38fun/main/config.c
U   team/file/t38fun/main/manager.c
U   team/file/t38fun/main/say.c
U   team/file/t38fun/main/srv.c
U   team/file/t38fun/main/tdd.c
U   team/file/t38fun/pbx/pbx_ael.c
U   team/file/t38fun/res/res_jabber.c
U   team/file/t38fun/res/res_musiconhold.c

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r89131 | file | 2007-11-09 10:34:31 -0600 (Fri, 09 Nov 2007) | 168 lines

Merged revisions 89032,89036-89037,89042,89045-89046,89053,89079,89085,89088,89090,89093,89095,89097,89099,89101,89103,89105,89111,89115,89119,89125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines

Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable.
(closes issue ASTERISK-10551)
Reported by: pj

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r89036 | murf | 2007-11-06 13:52:50 -0400 (Tue, 06 Nov 2007) | 1 line

closes issue ASTERISK-8547 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix.
........
r89037 | russell | 2007-11-06 14:20:07 -0400 (Tue, 06 Nov 2007) | 11 lines

If someone were to delete the files used by an existing MOH class, and then
issue a reload, further use of that class could result in a crash due to
dividing by zero.  This set of changes fixes up some places to prevent this
from happening.

(closes issue ASTERISK-10500)
Reported by: jcomellas
Patches:
     res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282)
 Additional changes added by me.

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r89042 | oej | 2007-11-06 14:53:37 -0400 (Tue, 06 Nov 2007) | 2 lines

Bug fixes to tdd support in zaptel.

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r89045 | tilghman | 2007-11-06 15:09:06 -0400 (Tue, 06 Nov 2007) | 2 lines

We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops).

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r89046 | qwell | 2007-11-06 15:09:30 -0400 (Tue, 06 Nov 2007) | 4 lines

Correctly set the total number of channels from a zaptel transcoder board.

SPD-49, patch by Matthew Nicholson.

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r89053 | russell | 2007-11-06 16:18:49 -0400 (Tue, 06 Nov 2007) | 3 lines

Fix init_classes() so that classes that actually do have files loaded aren't
treated as empty, and immediately destroyed ...

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r89079 | tilghman | 2007-11-07 00:07:49 -0400 (Wed, 07 Nov 2007) | 5 lines

Suppress AEL warnings on load.
Reported by: eliel
Patch by: eliel
Closes issue ASTERISK-10703

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r89085 | mmichelson | 2007-11-07 11:56:49 -0400 (Wed, 07 Nov 2007) | 5 lines

Fixing a segfault in the manager "core show channels concise" command.

(closes issue ASTERISK-10708, reported by arnd and patched by ys)


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r89088 | murf | 2007-11-07 17:40:28 -0400 (Wed, 07 Nov 2007) | 1 line

In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho.
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r89090 | mmichelson | 2007-11-07 18:40:35 -0400 (Wed, 07 Nov 2007) | 6 lines

This patch makes it possible for SIP phones to dial extensions defined with '#' characters
in extensions.conf AND maintain their escaped characters when forming URI's

(closes issue ASTERISK-10265, reported by cahen, patched by me, code review by file)


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r89093 | tilghman | 2007-11-07 19:39:37 -0400 (Wed, 07 Nov 2007) | 7 lines

The member refcount must be incremented, to avoid using it after deallocation.
A huge thanks go to lvl- for patiently providing the necessary valgrind output
that was necessary to finding this problem of memory corruption.
Reported by: lvl-
Patch by: tilghman
Closes issue ASTERISK-10699

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r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines

If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan.
(closes issue ASTERISK-10710)
Reported by: spditner

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r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines

Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support.
(closes issue ASTERISK-10498)
Reported by: flefoll
(closes issue ASTERISK-10472)
Reported by: ramonpeek
(closes issue ASTERISK-9288)
Reported by: atca_pres

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r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines

Improve the devicestate logic for multiple devices. If any are available then the extension is considered available.
(closes issue ASTERISK-9843)
Reported by: nic_bellamy
Patches:
     sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299)

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r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines

Do not add a sip: to the beginning of the To URI unless needed.
(closes issue ASTERISK-10331)
Reported by: goestelecom

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r89103 | tilghman | 2007-11-08 00:55:19 -0400 (Thu, 08 Nov 2007) | 2 lines

Typo

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r89105 | kpfleming | 2007-11-08 01:26:47 -0400 (Thu, 08 Nov 2007) | 2 lines

fix a glaring bug in the new SRV record handling that would cause incorrect weight sorting

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r89111 | mmichelson | 2007-11-08 12:47:23 -0400 (Thu, 08 Nov 2007) | 5 lines

I made this same adjustment in trunk to fix a bug, and it makes sense to do it in 1.4 as
well. If an imapfolder is specified in voicemail.conf, don't ever explicitly connect to
INBOX since it may not exist.


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r89115 | qwell | 2007-11-08 14:45:15 -0400 (Thu, 08 Nov 2007) | 4 lines

Avoid warnings on load when using sample configuration files.

Issue 11195, patch by eliel.

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r89119 | mmichelson | 2007-11-08 17:00:08 -0400 (Thu, 08 Nov 2007) | 7 lines

Rework of the commit I made yesterday to use the already built-in
ast_uri_decode function as opposed to my home-rolled one. Also added
comments.

Thanks to oej for pointing me in the right direction


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r89125 | qwell | 2007-11-08 19:52:35 -0400 (Thu, 08 Nov 2007) | 4 lines

Properly say the seconds here..

Issue 11203, fix described by vma.

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