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Summary:ASTERISK-10085: Use SIP_TRANS_TIMEOUT instead of 32000
Reporter:Gregory Hinton Nietsky (irroot)Labels:
Date Opened:2007-08-15 02:36:44Date Closed:2007-08-15 15:43:13
Priority:TrivialRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip_timeout.patch
Description:if there was a realy trivial option i would have selected it :)

PS there is no 1.4.10.1 option on the asterisk version drop down
Comments:By: Digium Subversion (svnbot) 2007-08-15 09:00:59

Repository: asterisk
Revision: 79523

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r79523 | file | 2007-08-15 09:00:57 -0500 (Wed, 15 Aug 2007) | 6 lines

(closes issue ASTERISK-10085)
Reported by: irroot
Patches:
     sip_timeout.patch uploaded by irroot (license 52)
Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten.

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By: Digium Subversion (svnbot) 2007-08-15 09:02:48

Repository: asterisk
Revision: 79524

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r79524 | file | 2007-08-15 09:02:47 -0500 (Wed, 15 Aug 2007) | 14 lines

Merged revisions 79523 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6 lines

(closes issue ASTERISK-10085)
Reported by: irroot
Patches:
     sip_timeout.patch uploaded by irroot (license 52)
Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten.

........

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By: Digium Subversion (svnbot) 2007-08-15 15:43:13

Repository: asterisk
Revision: 79622

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r79622 | tilghman | 2007-08-15 15:43:12 -0500 (Wed, 15 Aug 2007) | 89 lines

Merged revisions 79471,79507,79524,79529,79558,79595,79611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r79471 | russell | 2007-08-14 13:50:12 -0500 (Tue, 14 Aug 2007) | 10 lines

Merged revisions 79470 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r79470 | russell | 2007-08-14 13:49:10 -0500 (Tue, 14 Aug 2007) | 2 lines

Fix another spot where an iax2_peer would be leaked if realtime was in use.

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r79507 | crichter | 2007-08-15 06:27:51 -0500 (Wed, 15 Aug 2007) | 9 lines

Merged revisions 78936 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r78936 | crichter | 2007-08-10 15:24:03 +0200 (Fr, 10 Aug 2007) | 1 line

fixed a bug with the useruser information element. We send them now also in the disconnect message.
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r79524 | file | 2007-08-15 09:20:36 -0500 (Wed, 15 Aug 2007) | 14 lines

Merged revisions 79523 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r79523 | file | 2007-08-15 11:18:44 -0300 (Wed, 15 Aug 2007) | 6 lines

(closes issue ASTERISK-10085)
Reported by: irroot
Patches:
     sip_timeout.patch uploaded by irroot (license 52)
Change hardcoded timer value to defined value. I'm doing this in 1.4 as well so if it needs to be changed in the future this place would not have been forgotten.

........

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r79529 | mmichelson | 2007-08-15 09:27:35 -0500 (Wed, 15 Aug 2007) | 13 lines

Merged revisions 79527 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r79527 | mmichelson | 2007-08-15 09:26:40 -0500 (Wed, 15 Aug 2007) | 5 lines

Fixed an error in the Russian language voicemail intro.

(issue ASTERISK-10087, reported and patched by Oleh)


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r79558 | file | 2007-08-15 09:42:49 -0500 (Wed, 15 Aug 2007) | 14 lines

Merged revisions 79553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r79553 | file | 2007-08-15 11:40:23 -0300 (Wed, 15 Aug 2007) | 6 lines

(closes issue ASTERISK-10073)
Reported by: irroot
(closes issue ZAP-217)
Reported by: flo_turc
Increase maximum timestamp skew to 120. 20 was apparently far too low.

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r79595 | murf | 2007-08-15 14:21:27 -0500 (Wed, 15 Aug 2007) | 1 line

This commit closes bug 7605, and half-closes 7638. The AEL code has been redistributed/repartitioned to allow code re-use both inside and outside of Asterisk. This commit introduces the utils/conf2ael program, and an external config-file reader, for both normal config files, and for extensions.conf (context, exten, prio); It provides an API for programs outside of asterisk to use to play with the dialplan and config files.
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r79611 | file | 2007-08-15 14:34:42 -0500 (Wed, 15 Aug 2007) | 2 lines

Remove properties that appeared from Steve's last branch merge. Automerge has already run so everyone's branches based off of trunk are probably toast by now.

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