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Summary:ASTERISK-10551: hints display 'unreachable' peers still as 'idle'
Reporter:pj (pj)Labels:
Date Opened:2007-10-17 02:12:48Date Closed:2007-11-09 10:34:33.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Subscriptions
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:if peer becomes 'unreachable' hints are not updated (even after waiting several minutes)
I think this is wrong behaviour, hints should always immediatelly reflect unreachable peers and send notify about state change to subscribers...


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

*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
743/743                    85.160.86.91     D   N      5060     UNREACHABLE


core show hints
ipbx*CLI>
   -= Registered Asterisk Dial Plan Hints =-

                   743@linestates          : SIP/743               State:Idle            Watchers  1
Comments:By: pj (pj) 2007-10-17 02:45:29

it seems, that notify about state change is send to subscribers after peer registration expires, it really should be changed, that notify will be send immediatelly after 'sip qualify' discovers, that peer is death.

[Oct 17 09:12:52] NOTICE[10732]: chan_sip.c:16691 sip_poke_noanswer: Peer '743' is now UNREACHABLE!  Last qualify: 212
[Oct 17 09:47:00]   == Extension Changed 743 new state Unavailable for Notify User 324
By: Digium Subversion (svnbot) 2007-11-06 11:06:12.000-0600

Repository: asterisk
Revision: 89032

U   branches/1.4/channels/chan_sip.c

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r89032 | file | 2007-11-06 11:06:11 -0600 (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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By: Digium Subversion (svnbot) 2007-11-06 11:08:10.000-0600

Repository: asterisk
Revision: 89034

_U  trunk/
U   trunk/channels/chan_sip.c

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r89034 | file | 2007-11-06 11:08:09 -0600 (Tue, 06 Nov 2007) | 12 lines

Merged revisions 89032 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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By: Digium Subversion (svnbot) 2007-11-06 15:31:34.000-0600

Repository: asterisk
Revision: 89066

_U  team/murf/fast-ast2/
U   team/murf/fast-ast2/apps/app_amd.c
U   team/murf/fast-ast2/apps/app_chanisavail.c
U   team/murf/fast-ast2/apps/app_chanspy.c
U   team/murf/fast-ast2/apps/app_directed_pickup.c
U   team/murf/fast-ast2/apps/app_exec.c
U   team/murf/fast-ast2/apps/app_festival.c
U   team/murf/fast-ast2/apps/app_followme.c
U   team/murf/fast-ast2/apps/app_forkcdr.c
U   team/murf/fast-ast2/apps/app_getcpeid.c
U   team/murf/fast-ast2/apps/app_macro.c
U   team/murf/fast-ast2/apps/app_minivm.c
U   team/murf/fast-ast2/apps/app_mixmonitor.c
U   team/murf/fast-ast2/apps/app_morsecode.c
U   team/murf/fast-ast2/apps/app_mp3.c
U   team/murf/fast-ast2/apps/app_nbscat.c
U   team/murf/fast-ast2/apps/app_playback.c
U   team/murf/fast-ast2/apps/app_readfile.c
U   team/murf/fast-ast2/apps/app_sayunixtime.c
U   team/murf/fast-ast2/apps/app_sms.c
U   team/murf/fast-ast2/apps/app_softhangup.c
U   team/murf/fast-ast2/apps/app_speech_utils.c
U   team/murf/fast-ast2/apps/app_stack.c
U   team/murf/fast-ast2/apps/app_test.c
U   team/murf/fast-ast2/apps/app_waitforring.c
U   team/murf/fast-ast2/apps/app_waitforsilence.c
U   team/murf/fast-ast2/apps/app_while.c
U   team/murf/fast-ast2/channels/chan_agent.c
U   team/murf/fast-ast2/channels/chan_gtalk.c
U   team/murf/fast-ast2/channels/chan_jingle.c
U   team/murf/fast-ast2/channels/chan_sip.c
U   team/murf/fast-ast2/codecs/codec_zap.c
U   team/murf/fast-ast2/include/asterisk/jabber.h
U   team/murf/fast-ast2/include/asterisk/lock.h
U   team/murf/fast-ast2/include/asterisk/tdd.h
U   team/murf/fast-ast2/main/ast_expr2.fl
U   team/murf/fast-ast2/main/ast_expr2f.c
U   team/murf/fast-ast2/main/astmm.c
U   team/murf/fast-ast2/main/channel.c
U   team/murf/fast-ast2/main/config.c
U   team/murf/fast-ast2/main/fskmodem.c
U   team/murf/fast-ast2/main/loader.c
U   team/murf/fast-ast2/main/pbx.c
U   team/murf/fast-ast2/main/tdd.c
U   team/murf/fast-ast2/res/res_features.c
U   team/murf/fast-ast2/res/res_indications.c
U   team/murf/fast-ast2/res/res_jabber.c
U   team/murf/fast-ast2/res/res_monitor.c
U   team/murf/fast-ast2/res/res_musiconhold.c

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r89066 | murf | 2007-11-06 15:31:33 -0600 (Tue, 06 Nov 2007) | 188 lines

Merged revisions 89031,89034,89038,89041,89043-89044,89047-89052,89054-89055,89057,89062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r89031 | rizzo | 2007-11-06 10:05:13 -0700 (Tue, 06 Nov 2007) | 17 lines

Fix embedding of modules on FreeBSD:
the constructor for the list of modules was run
after the constructors for the embedded modules
(which appended entries to the list).
As a result, the list appeared empty when it was
time to use it.

On linux the order of execution of constructor
was evidently different (it may depend on the
ordering of modules in the ELF file).

This is only a workaround - there may be other
situations where the execution of constructors
causes problems, so if we manage to find a more
general solution this workaround can go away.


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r89034 | file | 2007-11-06 10:10:03 -0700 (Tue, 06 Nov 2007) | 12 lines

Merged revisions 89032 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

........

................
r89038 | russell | 2007-11-06 11:23:36 -0700 (Tue, 06 Nov 2007) | 19 lines

Merged revisions 89037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89037 | russell | 2007-11-06 12:20:07 -0600 (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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r89041 | qwell | 2007-11-06 11:44:19 -0700 (Tue, 06 Nov 2007) | 4 lines

Allow gtalk and jingle to use TLS connections again.

Closes issue ASTERISK-9675

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r89043 | oej | 2007-11-06 12:04:29 -0700 (Tue, 06 Nov 2007) | 12 lines

Merged revisions 89042 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 lines

Bug fixes to tdd support in zaptel.

........

(Small changes for trunk)

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r89044 | mmichelson | 2007-11-06 12:04:45 -0700 (Tue, 06 Nov 2007) | 7 lines

"show application <foo>" changes for clarity.

(closes issue ASTERISK-10696, reported and patched by blitzrage)

Many thanks!


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r89047 | qwell | 2007-11-06 12:10:18 -0700 (Tue, 06 Nov 2007) | 12 lines

Merged revisions 89046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89046 | qwell | 2007-11-06 13:09:30 -0600 (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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r89048 | oej | 2007-11-06 12:10:26 -0700 (Tue, 06 Nov 2007) | 2 lines

Additional TDD changes (preparing for SIP changes - adding TDD support to SIP)

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r89049 | tilghman | 2007-11-06 12:16:02 -0700 (Tue, 06 Nov 2007) | 10 lines

Merged revisions 89045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89045 | tilghman | 2007-11-06 13:09:06 -0600 (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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r89050 | oej | 2007-11-06 12:23:10 -0700 (Tue, 06 Nov 2007) | 2 lines

Formatting. Illegaly using some spare spaces from Russell's space-bucket.

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r89051 | murf | 2007-11-06 12:40:33 -0700 (Tue, 06 Nov 2007) | 1 line

Hoping to avoid a crash in OSX for a problem blitzrage found
................
r89052 | russell | 2007-11-06 12:51:37 -0700 (Tue, 06 Nov 2007) | 4 lines

Fix the memory show allocations CLI command so that it doesn't spew out all
of the current memory allocations when you start Asterisk, when the command's
handler gets called for initialization.

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r89054 | russell | 2007-11-06 13:22:50 -0700 (Tue, 06 Nov 2007) | 11 lines

Merged revisions 89053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89053 | russell | 2007-11-06 14:18:49 -0600 (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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r89055 | mmichelson | 2007-11-06 13:32:49 -0700 (Tue, 06 Nov 2007) | 9 lines

Instead of trying to callback a local channel on a failed attended transfer, call
the device that made the transfer instead. This makes for much smoother calling back
when queues are involved.

(closes issue ASTERISK-10680, reported by IPetrov)

Tremendous thanks to Russell for pulling me out of my block I was having on this one


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r89057 | file | 2007-11-06 13:55:58 -0700 (Tue, 06 Nov 2007) | 4 lines

Remove native bridging check for DTMF based transfers. Thanks to the last batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed.
(closes issue ASTERISK-10697)
Reported by: ibc

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r89062 | murf | 2007-11-06 14:08:38 -0700 (Tue, 06 Nov 2007) | 9 lines

Merged revisions 89036 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89036 | murf | 2007-11-06 10:52:50 -0700 (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.
........

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

------------------------------------------------------------------------
By: Digium Subversion (svnbot) 2007-11-09 10:34:33.000-0600

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

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

........
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

........
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.

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

Bug fixes to tdd support in zaptel.

........
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).

........
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.

........
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 ...

........
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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