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1.4.27-rc3
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1.4.29-rc1
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1.4.30
1.4.30-rc1
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1.4.31-rc1
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1.4.32
1.4.32-rc1
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1.4.33-rc1
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1.4.34-rc1
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1.4.35-rc1
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1.4.36-rc1
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1.4.37-rc1
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1.4.40
1.4.40-rc1
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${ noResults }
3282 Commits (390b5bbcd64552aa468c08fea6b5308ad3d566a4)
Author | SHA1 | Message | Date |
---|---|---|---|
|
5e9dc5e1f3 |
Add voicemail related applications and functions XML documentation:
applications: - VoiceMail() - VoiceMailMain() - MailboxExists() - VMAuthenticate() functions: - MAILBOX_EXISTS() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161604 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e9ab875265 |
Introduce SMS() application XML documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161571 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
206fe71680 |
Move Speech* applications and functions documentation to XML.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
07311720f2 |
If the autoloop flag is set on a channel, then we need to
add 1 to the priority when checking if the extension exists. Otherwise, gosubs will fail. This was discovered when investigating an asterisk-users mailing list post made by Gary Hawkins. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
3eee1dbb9b |
Use ast_free() instead of free(), pointed out by eliel on IRC.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161350 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
9d2a8810e6 |
When using IMAP_STORAGE, it's important to convert bare newlines (\n) in
emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed by Mark M. on IRC. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
7d0c1f40fb |
Resolve a compiler warning from buildbot about a NULL format string.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161252 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1e8e12efcf |
Janitor, use ARRAY_LEN() when possible.
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
2afd7a09a7 |
Check the return value of fread/fwrite so the compiler doesn't complain. Only a
problem when IMAP_STORAGE is enabled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161147 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
c42aef2ebb |
Merged revisions 160770 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines Some compilers warn on null format strings; some don't (caught by buildbot) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
a53877b469 |
Add some safety measures when using gosub, especially when using the options
for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160626 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
d635ee0f43 |
- Add <variable /> tags when naming a channel variable.
- Add <filename /> tags when naming a filename. - Simplify the xml formatting putting some enters. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
ac1b520de6 |
When investigating issue #13548, I found that gosub
handling in app_queue was just completely wrong, mostly because the channel operations being performed were being done on the incorrect channel. With this set of changes, a gosub will correctly run on the answering queue member's channel. There are still crash issues which occur if there are dialplan syntax errors, so I cannot yet close the referenced issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
bfe0c6c714 |
- Avoid setting .synopsis and .syntax if we are using XML documentation (or the
xml documentation wont be loaded). - Use <variable></variable> to refer to a dialplan variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160447 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
29502d3bac |
Add LOCAL_PEEK function, as requested by lmadsen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160344 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
3d4c0cd421 |
Merged revisions 160207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160208 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
b323c972b6 |
Allow the '#' sign to exist within an extension (inspired by issue #13330)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159853 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e14dfcbedc |
improve handling of API calls provided by loaded modules through use of some GCC features; this makes app_stack's usage of AGI APIs even cleaner, and will allow it to work 'as expected' either with or without res_agi being loaded
reviewed at http://reviewboard.digium.com/r/62 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
5cf09591b0 |
Add some necessary hangup commands in the case that forwarding
a call fails 1) Hang up the original destination if the local channel cannot be requested. 2) Hang up the local channel (in addition to the original destination) if ast_call fails when calling the newly created local channel. This prevents channels from sticking around forever in the case of a botched call forward (e.g. to an extension which does not exist). (closes issue #13764) Reported by: davidw Patches: 13764_v2.patch uploaded by putnopvut (license 60) Tested by: putnopvut, davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
2d20ab2b07 |
Make the options for the general and profiles more consistent
for the "pls_hold_prompt" option. This does not affect any released version of Asterisk, so there is no need to update the CHANGES file for this. (closes issue #13893) Reported by: eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
4ea49e697e |
Add missing variable declaration for PPC code
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
807566899a |
Copyright clarification; also, have variable set to "t" or "i" on timeout or
invalid extension, respectively. (closes issue #13944) Reported by: chappell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159054 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
ac296a4ad3 |
Merged revisions 159025 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159025 | tilghman | 2008-11-24 22:50:00 -0600 (Mon, 24 Nov 2008) | 3 lines System call ioperm is non-portable, so check for its existence in autoconf. (Closes issue #13863) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159050 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
fd8caa1778 |
This is basically a complete rollback of r155401, as it was determined that
it would be best to maintain API compatibility. Instead, this commit introduces ao2_callback_data() which is functionally identical to ao2_callback() except that it allows you to pass arbitrary data to the callback. Reviewed by Mark Michelson via ReviewBoard: http://reviewboard.digium.com/r/64 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
17ed84ff07 |
Make the Join event from app_queue use CallerIDNum insead of CallerID for
indicating the callerid number just like the rest of asterisk. (closes issue #13883) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
85bbf3ff33 |
This patch adds a new application for sending MWI to phones via Asterisk's event subsystem. Also, the minivm documentation is all converted to use xmldocs.
(closes issue #13946) Reported by: Marquis Patches: minivmmwi_plus_xmldocs.patch uploaded by Marquis (license 32) Tested by: otherwiseguy, Marquis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
7a554a7386 |
Merged revisions 158053 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158053 | mmichelson | 2008-11-20 11:33:06 -0600 (Thu, 20 Nov 2008) | 12 lines Make sure to set the hangup cause on the calling channel in the case that ast_call() fails. For incoming SIP channels, this was causing us to send a 603 instead of a 486 when the call-limit was reached on the destination channel. (closes issue #13867) Reported by: still_nsk Patches: 13867.diff uploaded by putnopvut (license 60) Tested by: blitzrage ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158066 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
9e1283e160 |
Add a space to the output
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
de5ba432da |
Add a RES_NOT_DYNAMIC case for the CLI command
'queue remove member' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157939 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
81a16aa982 |
make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines
also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
6c554e4996 |
Fix the logic for when delete=yes when IMAP storage
is in use so that the message is deleted from both local and IMAP storage. (closes issue #13642) Reported by: jaroth Patches: deleteyes.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
53f3870ed3 |
Merged revisions 157365 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157365 | jpeeler | 2008-11-18 13:13:33 -0600 (Tue, 18 Nov 2008) | 6 lines (closes issue #13899) Reported by: akkornel This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
d91f1df3e0 |
Merged revisions 157305 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines Fix a crash in the end_bridge_callback of app_dial and app_followme which would occur at the end of an attended transfer. The error occurred because we initially stored a pointer to an ast_channel which then was hung up due to a masquerade. This commit adds a "fixup" callback to the bridge_config structure to allow for end_bridge_callback_data to be changed in the case that a new channel pointer is needed for the end_bridge_callback. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
1287486dbf |
Can't use items duplicated off the stack frame in an element returned from
a function: in these cases, we have to use the heap, or garbage will result. (closes issue #13898) Reported by: alecdavis Patches: 20081114__bug13898__2.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
cf6c66de65 |
Fix some refcounting in app_queue.c and change the
hashing used by app_queue.c to be case-insensitive. This is accomplished by adding a new case-insensitive hashing function. This was necessary to prevent bad refcount errors (and potential crashes) which would occur due to the fact that queues were initially read from the config file in a case-sensitive manner. Then, when a user issued a CLI command or manager action, we allowed for case-insensitive input and used that input to directly try to find the queue in the hash table. The result was either that we could not find a queue that was input or worse, we would end up hashing to a completely bogus value based on the input. This commit resolves the problem presented in issue #13703. However, that issue was reported against 1.6.0. Since this fix introduces a behavior change, I am electing to not place this same fix in to the 1.6.0 or 1.6.1 branches, and instead will opt for a change which does not change behavior. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
ea9955dd2a |
Merged revisions 156816 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156816 | mmichelson | 2008-11-14 09:18:59 -0600 (Fri, 14 Nov 2008) | 10 lines If the prompt to reenter a voicemail password timed out, it resulted in the password not being saved, even if the input matched what you gave when first prompted to enter a new password. This is because the return value of ast_readstring was checked, but not checked properly. This bug was discovered by Jared Smith during an Asterisk training course. Thanks for reporting it! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c4fe83046b |
Merged revisions 156755 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156755 | tilghman | 2008-11-13 18:41:37 -0600 (Thu, 13 Nov 2008) | 6 lines ast_waitfordigit() requires that the channel be up, for no good logical reason. This prevents While/EndWhile from working within the "h" extension. Reported by: jgalarneau (for ABE C.2) Fixed by: me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
a22928b853 |
Introduce XML documentation for:
- MeetMe() - MeetMeCount() - MeetMeChannelAdmin() - MeetMeAdmin() - SLAStation() - SLATrunk() - Add an attribute to optionlist 'hasparams' with the same functionality as the one we have in <parameter> and <argument> (the DTD was updated) - Fix a leak when getting an attribute while parsing an <optionlist>. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156575 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
10afda33c7 |
Merged revisions 156386 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156386 | tilghman | 2008-11-12 15:18:57 -0600 (Wed, 12 Nov 2008) | 5 lines When using call limits under 1 second, infinite call lengths are allowed, instead. (closes issue #13851) Reported by: ruddy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
221998f4d4 |
Merged revisions 156294 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156294 | tilghman | 2008-11-12 13:26:45 -0600 (Wed, 12 Nov 2008) | 6 lines If the SLA thread is not started, then reload causes a memory leak. (closes issue #13889) Reported by: eliel Patches: app_meetme.c.patch uploaded by eliel (license 64) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
611e737463 |
Merged revisions 156289 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156289 | jpeeler | 2008-11-12 13:10:12 -0600 (Wed, 12 Nov 2008) | 3 lines For whatever reason, gcc only warned me about the possible use of an uninitialized variable when compiling 1.6.1. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156290 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
f7aaa011b6 |
Merged revisions 156178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156178 | jpeeler | 2008-11-12 11:53:44 -0600 (Wed, 12 Nov 2008) | 8 lines (closes issue #13173) Reported by: pep This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference. Reviewed by Russell on Review Board: http://reviewboard.digium.com/r/25/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a9e84c1e51 |
Merged revisions 156167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156167 | mmichelson | 2008-11-12 11:38:33 -0600 (Wed, 12 Nov 2008) | 7 lines When doing some tests, I was having a crash at the end of every call if an attended transfer occurred during the call. I traced the cause to the CDR on one of the channels being NULL. murf suggested a check in the end bridge callback to be sure the CDR is non-NULL before proceeding, so that's what I'm adding. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156169 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
48522988ab |
In order to move away from nested function use, some changes to the recently introduced
ast_channel_search_locked need to be made. Specifically, the caller needs to be able to pass arbitrary data which in turn is passed to the callback. This patch addresses all of the nested functions currently in asterisk trunk. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155590 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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9ef09ad1d4 |
Merged revisions 155553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines Use static functions here instead of nested ones. This requires a small change to the ast_bridge_config struct as well. To understand the reason for this change, see the following post: http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155554 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
30d1744ffc |
Add ability to pass arbitrary data to the ao2_callback_fn (called from
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either of these mandates that the passed 'arg' is a hashable object, making searching for an ao2 object based on outside criteria difficult. Reviewed by Russell and Mark M. via ReviewBoard: http://reviewboard.digium.com/r/36/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155401 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
8a25d59bf8 |
Fix some whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154926 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
6ac794074e |
Update a couple places to use the new ast_channel_search_locked API call.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
34e094726e |
Add TestClient() and TestServer() applications XML
documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154801 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5fd54b0fe1 |
Add more [see also] references based on TFOT.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |