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${ noResults }
21821 Commits (ca0f2acab77e52464dc126f76c212de528edb1a5)
Author | SHA1 | Message | Date |
---|---|---|---|
|
ca0f2acab7 |
Set 'core' support level for test_format_api.c
........ Merged revisions 341146 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341147 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
2ffea6ddc3 |
Multiple revisions 341108,341112
........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged revisions 341108,341112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341122 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
a79c41ee66 |
Add information about limitations of new codec support in channel drivers.
(issue ASTERISK-18680) ........ Merged revisions 341094 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341096 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
2cb5178d29 |
Don't try to remove peers without IPs from peers_by_ip
(closes issue ASTERISK-18696) ........ Merged revisions 341088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341089 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c292e39cdc |
Change the internal name of the menuselect options that are used to control
whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. ........ Merged revisions 341022 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341023 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341024 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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899df042f5 |
Fix simple switch to not progress a call when call already progressed.
If a simple switch was started on a device and then a specific call made (such as redial or speed dial), on timeout of the simple switch the call would be attempted again. This patch only allows the simple switch to make a call if the substate is still in the collecting digits mode. Also added small debug message to dialAndAactivate sub. Tested by snuff and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340973 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4b9546abdf |
Merged revisions 340971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340971 | kmoore | 2011-10-14 15:50:37 -0500 (Fri, 14 Oct 2011) | 15 lines Merged revisions 340970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340972 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e77f1a6ae1 |
Some additional module documentation changes for 10 for the menuselect change.
(issue ASTERISK-18268) ........ Merged revisions 340931 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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19d3e269f6 |
Avoid unnecessary WARNING message
Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban ........ Merged revisions 340878 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340879 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cabf08b511 |
Fix DTMF blind transfer continuing to execute dialplan after transfer.
Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 340809 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340810 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d0ab521e61 |
Update 10 merged property.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340812 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ec93d933c0 |
Restore branch 10 merge properties.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3ee015db7a |
Merged revisions 339463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c48bee8e82 |
Merged revisions 340718 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340718 | schmidts | 2011-10-13 06:59:50 +0000 (Thu, 13 Oct 2011) | 9 lines Merged revisions 340717 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 Oct 2011) | 3 lines storing the route-set also on a 181 response not only on 180,182 or 183. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5c77498afd |
Initialize ast_sockaddr before calling ast_sockaddr_resolve
Avoid possible jump based on unitialized value ........ Merged revisions 340715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340716 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9d83162d55 |
Don't skip the query field on a realtime multi query
There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. ........ Merged revisions 340662 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340663 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340665 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ee8844782c |
Merged revisions 340577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340577 | schmidts | 2011-10-12 20:33:37 +0000 (Mit, 12 Okt 2011) | 9 lines Merged revisions 340576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 Okt 2011) | 3 lines Store route-set from provisional SIP responses so early-dialog requests can be routed properly ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340626 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e7ebf7d5ab |
Merged revisions 340578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340578 | twilson | 2011-10-12 13:57:19 -0700 (Wed, 12 Oct 2011) | 16 lines Merged revisions 340534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) | 9 lines Update SIP realtime fullcontact regardless of caching We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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3bc3e9bbb7 |
Initialize the PRI channel alarms properly on startup.
The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen ........ Merged revisions 340522 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340523 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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796ed62f47 |
Update MeetMe p and X option documentation when interacting with the s option.
ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 ........ Merged revisions 340470 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340471 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f2cc666a99 |
Fix verbose messages when IPv6 logic was added
(closes issue ASTERISK-18612) Reported by: Tim Osman ........ Merged revisions 340418 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340419 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340420 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9abab10b66 |
Add protection for SS7 channel allocation and better glare handling.
* Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 340365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340366 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340367 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b63c1cc545 |
Fix some potential deadlocks pointed out by helgrind.
* Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. ........ Merged revisions 340284 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340310 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1ec8a9d896 |
Update SHA1 code to RFC 6234
RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340283 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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067250f74c |
Convert registered AMI actions to ao2 objects.
* Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340281 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340282 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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15fd1e375c |
Return error when no rows are deleted for AMI DBDelTree
(closes issue AST-654) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340224 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cf8db24132 |
Merged revisions 340222 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011) | 8 lines On astdb conversion, also warn about permissions requirements The user running Asterisk must have permission to the directory the Asterisk database resides in since SQLite 3 needs to be able to create a journal file. (closes issue ASTERISK-18174) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6708ee76a0 |
Merged revisions 340219-340220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10 Oct 2011) | 8 lines Add astdb conversion utility for Berkeley to SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 they can use the astdb2bdb utility to convert the database back to the Berkeley format that Asterisk 1.8 uses. Review: https://reviewboard.asterisk.org/r/1502/ ........ r340220 | twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines Add a missing file for the astdb2bdb conversion utility ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340221 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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4ec8d57454 |
Merged revisions 340165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340165 | mjordan | 2011-10-10 15:30:18 -0500 (Mon, 10 Oct 2011) | 20 lines Merged revisions 340164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) | 13 lines Updated chan_sip to place calls on hold if SDP address in INVITE is ANY This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bb07ca66a1 |
Merged revisions 340109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) | 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7cb2ac8664 |
Add skinny version 17 protocol support.
Added some data to skinny packet structures to make compatible with v17. Added protocolversion to device, set on registration based on the version provided by device. v17 includes some increased ip space for ip6. This patch increases ip space in the packets but still only uses ip4. Some packet structures duplicated (ip4 and ip6 types). ip4 type used unless version is greater or equal to 17. Tested by snuff and myself on 7961 with recent 8.5 firmware. Also tested compatible with old 7960 and older 30VIPs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340071 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c5546e2bd7 |
Increase SKINNY_MAX_PACKET and add some logging.
Increase SKINNY_MAX_PACKET to 2000 bytes to handle some messages in v17 that are greater than the old 1000 bytes. Also add some useful logging regarding packet and session handling. A device (with protocol v17) was sending a packet with length greater than 1000 which resulted in the TCP session being destroyed and registration being retryed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340070 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0ac40dc255 |
Merged revisions 340031 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) | 8 lines Return -1 to skinny_session if register rejected. If device registration is rejected, return -1 so that the session is destroyed immediately. Previously, a segfault would occur on a graceful shutdown if a register is rejected and the skinny_session has not yet timed out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340032 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b90964eda5 |
Merged revisions 339992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) | 9 lines Remove log message on traverse session list. On destroying a session, a list of sessions is traversed to find the matching session. For each session not matching, skinny erroneously logged that the session was not matched. While technically correct the message was misleading, and tended to indicate errors that were not there. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7e5ce2ac49 |
Merged revisions 339942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339942 | igorg | 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) | 12 lines Merged revisions 339938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines Fix compilation issue, caused by missed session structure (closes issue ASTERISK-18694) Reported by: alex70 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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326c3a39d5 |
Merged revisions 339885 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339885 | igorg | 2011-10-08 22:46:27 +0700 (Сбт, 08 Окт 2011) | 13 lines Merged revisions 339884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines Fix segfault in Unistim channel (closes issue ASTERISK-18638) Reported by: jonnt ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a01b34f488 |
Merged revisions 339831 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339831 | igorg | 2011-10-08 22:01:35 +0700 (Сбт, 08 Окт 2011) | 14 lines Merged revisions 339830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines Fix char array cast as short array in send_client() function (for ARM platform) (closes issue ASTERISK-17314) Reported by: jjoshua ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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56c9f288d6 |
Merged revisions 339777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339777 | rmudgett | 2011-10-07 14:36:24 -0500 (Fri, 07 Oct 2011) | 12 lines Merged revisions 339776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) | 5 lines Initialize option flags for SendURL application. (closes issue ASTERISK-18574) Reported by: marcelloceschia ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339778 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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363e3feffe |
Recorded merge of revisions 339681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011) | 10 lines Fixed segfault on core stop gracefully. There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff (in trunk) and myself. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6e5f97df77 |
Merged revisions 339720 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339720 | rmudgett | 2011-10-06 17:58:40 -0500 (Thu, 06 Oct 2011) | 27 lines Merged revisions 339719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines Fix regression in configure script for libpri capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix the unexpected requirement that libpri *must* have that feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI optional features required. Unfortunately, I thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and deleted those lines for libpri. The result was the HAVE_PRI_xxx defines that control the ability to use optional libpri features were also deleted. * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional features in a library that the source code could take advantage of if the code supports the feature. (closes issue ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d59bb319fc |
Fixed segfault on core stop gracefully.
There was an issue that the cap and confcap pointers for each line and device were being memcpy'd so they all pointed to the same ast_format_cap. On destroying, a segfault occured on the second call to the same struct. skinny reload now works again as well. Tested by snuff and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2f82296096 |
Merged revisions 339626 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339626 | rmudgett | 2011-10-06 12:53:00 -0500 (Thu, 06 Oct 2011) | 25 lines Merged revisions 339625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) | 18 lines Fix debugging messages generated by 'udptl debug'. * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339627 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7b75de509b |
Merged revisions 339586 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339586 | lmadsen | 2011-10-06 08:43:21 -0500 (Thu, 06 Oct 2011) | 16 lines Merged revisions 339566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011) | 8 lines Update prep_tarball script to download pre-exported documentation. I've updated the prep_tarball script to now download the pre-exported documentation from the Asterisk wiki. This will give us more control over what is being included in the tarball releases, and will make both the PDF and HTML exported documentation look much better (especially when viewing from a console). (Closes issue ASTERISK-18677) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339587 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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e4b07e2d38 |
Merged revisions 339512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339512 | rmudgett | 2011-10-05 12:01:46 -0500 (Wed, 05 Oct 2011) | 9 lines Merged revisions 339511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 Oct 2011) | 1 line Fix Dial F option notes formatting. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5d5db050f8 |
Merged revisions 339508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339508 | rmudgett | 2011-10-05 11:35:02 -0500 (Wed, 05 Oct 2011) | 18 lines Merged revisions 339504,339506 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011) | 7 lines Add missing documentation of required AMI action Challenge AuthType header. (closes issue ASTERISK-18554) Reported by: Vlad Povorozniuc Patches: __20110919-manager-challenge-docs.patch.txt (license #4999) patch uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI action Challenge. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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07133b3a96 |
Merged revisions 339507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339507 | mnicholson | 2011-10-05 11:32:59 -0500 (Wed, 05 Oct 2011) | 10 lines Merged revisions 339505 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct 2011) | 3 lines The app name in the documentation must match what we register the application as. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b698038995 |
Add generic faxdetect framehook to res_fax
Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan faxdetect allowing more flexibility. as soon as a fax tone is detected the framehook is removed. there is a penalty involved in running this framehook on non G711 channels as they will be transcoded. CNG tone is suppresed using the SQUELCH flag to allow WaitForNoise to be run on the channel to detect Voice. (Closes issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew Nicholson, Kevin Fleming Review: https://reviewboard.asterisk.org/r/1116/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1b3bd7ddb4 |
Merged revisions 339463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | 9 lines Only change the capabilities on the gateway when the session is been destroyed there is still a race condition that ends in a segfault. if the caps are changed the logic in res_fax_spandsp will run T30 code not gateway code to end the session. this has been experienced on a "slower" under spec system. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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f1440edfa7 |
Merged revisions 339407 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339407 | rmudgett | 2011-10-04 17:56:25 -0500 (Tue, 04 Oct 2011) | 15 lines Merged revisions 339406 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011) | 8 lines Make always create the MOH directory (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael Keuter ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339408 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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29b4c1a9f1 |
Merged revisions 339353 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339353 | jrose | 2011-10-04 14:44:02 -0500 (Tue, 04 Oct 2011) | 18 lines Merged revisions 339352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) | 12 lines Removes improper use of sound 'and' in German language mode from application saynumber Asterisk would say 'Five hundert und sechs und zwanzig' instead of 'Five hundert sechs und zwanzig'... which is both weird sounding and wrong. This patch makes sure Asterisk will only say the 'and' word between the single digit and double digit places. (closes issue ASTERISK-18212) Reported By: Lionel Elie Mamane Patches: upstream_germand_no_and.diff (License #5402) uploaded by Lionel Elie Mamane ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339354 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |