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Summary:ASTERISK-09288: Notify sent to a non-existent call
Reporter:atca_pres (atca_pres)Labels:
Date Opened:2007-04-20 08:36:41Date Closed:2007-11-09 10:34:34.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) Bug-NotifyAfterBye.pcap
( 1) patch-10946.txt
( 2) verbosedebug.txt
( 3) verbosedebug-NotifyAfterBye.txt
Description:Sc?nario :
A : Aastra 9113i 10.4.122.235
B : Mediatrix 1104 10.4.121.237
C : Mediatrix 2102 10.4.117.254
* : 10.4.119.245

B Calls A
A Answers
Call OK
A press the xfer key and calls C
A hangs up while C is ringing (blind transfer)
Call redirected BUT :
Look at packet 43 -> A bye the call (call-id ending with abd2) and * OK the bye (packet 44)
Now look at packet 45 * sends a NOTIFY to A same call-id with a 183 progress.
The packet is re-sent at packet 47,54,60,61,64 and then sadly not showing in the capture, * byes the call and is answered call does not exists by A.

I think * should stop send the Notify with 183 after the 200 OK to the BYE (packet 43-44)
Comments:By: Olle Johansson (oej) 2007-04-27 09:19:34

Those are two different transactions, separated by light-years really. The NOTIFY is updating about the progress of the transfer and needs to be there until we have a complete call setup between callee and transfer target.

By: atca_pres (atca_pres) 2007-04-30 13:08:47

I know these are not the same transactions. BUT Asterisk sends NOTIFY even if the transaction worked. The call is redirected and is working. Asteriks still sends NOTIFY with 183 ringing even after the call with the transferor is finished.

The call is established at packet 49 (200Ok) and you can see NOTIFY with 183 after that (packet 54,60,61,63,64, etc)

By: Olle Johansson (oej) 2007-05-02 04:44:19

Thre's no SIP debug in the verbosedebug file. Turn on SIP DEBUG, set core debug to 9 and capture everything. THanks.

By: atca_pres (atca_pres) 2007-05-02 08:40:37

Sorry about that.I've put the sip debug on in sip.conf I shouldn't forget it in the future.

thx

By: Denis Galvao (denisgalvao) 2007-06-30 17:35:51

Hi, is this related to this problem?

1. "A" call "B"
2. Reinvite from "A" to "B"
3. "B" transfer(flash button) "A" to "C"
4. "A" listen MOH
5. "B" hangup
6. "A" stop listen MOH and "C" start ringing
7. "A" doesn't listen anhything until "C" pickup the phone and start the conversation

The same problem occur without reinvite.

By: Joshua C. Colp (jcolp) 2007-11-02 09:53:27

It seems as though the response to the NOTIFY to update the progress of the transfer is being ignored by chan_sip. Please give the patch in 10946 a try to confirm.

By: atca_pres (atca_pres) 2007-11-05 13:46:36.000-0600

The patch in 10946 did indeed remove the NOTIFY. It does not however remove the BYE at the end of the call (that has already been BYE). Is this related ?

I'll upload a debug of what is happening with the patch

Thx

By: Digium Subversion (svnbot) 2007-11-07 19:09:32.000-0600

Repository: asterisk
Revision: 89097

U   branches/1.4/channels/chan_sip.c

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r89097 | file | 2007-11-07 19:09:28 -0600 (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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By: Digium Subversion (svnbot) 2007-11-07 19:12:35.000-0600

Repository: asterisk
Revision: 89098

_U  trunk/
U   trunk/channels/chan_sip.c

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r89098 | file | 2007-11-07 19:12:34 -0600 (Wed, 07 Nov 2007) | 16 lines

Merged revisions 89097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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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.
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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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