mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
13671 Commits (9e773ebd33ed86ef2e8ea72bbeaa75bd4b5e613c)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
9e773ebd33 |
Reverting merge 219520. This change was not necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219720 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
78e6e5b017 |
Really stop the stream, when ast_closestream() is called.
(closes issue #15129) Reported by: bmh Patches: 20090918__issue15129.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/372/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
1e24571def |
Make sure the iax_pvt exists before dereferencing it.
This fixes the latest crash posted on issue 15609. (issue #15609) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219586 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
0a3504f74b |
iax2 frame double free
The iax frame's retrans sched id was written over right before iax2_frame_free was called. In iax2_frame_free that retrans id is used to delete the sched item. By writing over the retrans field before the sched item could be deleted, it was possible for a retransmit to occur on a freed frame. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
66fff128f0 |
via-header branches not updated correctly on INVITE
INVITE requests must always contain a new unique branch id. When a new branch id is created for an INVITE, the dialog's invite_branch variable must be updated so CANCEL requests use the correct branch id. (closes issue #15262) Reported by: maniax Patches: asterisk-1.6.1.0-sip-branch.patch uploaded by tweety (license 608) invite_new_branch_trunk.diff uploaded by dvossel (license 671) Tested by: maniax, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
e2dabd44a3 |
Send a 100 Trying response when we detect a spiral.
This was problematic during spiral tests at SIPit... along with some other things as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
7e0f2c802f |
INVITE w/Replaces deadlock fix
This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read(). The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function. This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls. (closes issue #15151) Reported by: irroot Patches: invite_w_replaces_1.4.diff uploaded by dvossel (license 671) Tested by: irroot, dvossel Review: https://reviewboard.asterisk.org/r/371/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
ea3f81a68f |
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
d28e69ad5d |
Properly deal with quotes in the arguments of '#exec' includes.
(closes issue #15583) Reported by: pkempgen Patches: 20090726__issue15583.diff.txt uploaded by tilghman (license 14) 20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169) Tested by: pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@219023 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
6913d06d19 |
Fixes CID pattern matching behavior to mirror that of extension pattern matching.
Pattern matching for extensions uses a type of scoring system, giving values for specificity to each character in the pattern. Unfortunately, this is done character by character, in order. This does lead to some less specific patterns being first in line for matching, but it will usually get the job done. This patch merely brings CID matching to the same level as extension matching. This patch does not attempt to tackle the problem shared by extension matching. (closes issue #14708) Reported by: klaus3000 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
03d3a47ca7 |
Remove the IAXy firmware from Asterisk.
The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218798 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
cefd4b7b03 |
If the user enters the same password as before, don't signal an error when the change does nothing.
(closes issue #15492) Reported by: cbbs70a Patches: 20090713__issue15492.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218730 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
434dcbf847 |
Fix small memory leak in handle_init_event by always destroying the pthread
attr before returning. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
ca41240806 |
Send request contact header field with response to registrer queries instead of the address of record.
(closes issue #14438) Reported by: ravindrad Patches: regquerypatch uploaded by ravindrad (license 684) Tested by: ravindrad git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218578 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3913dcbc40 |
Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are accountcode and musicclass, and the inheritance of datastores. (closes issue #15372) Reported by: Romik Patches: 20090828__issue15372.diff.txt uploaded by tilghman (license 14) Tested by: cervajs git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218577 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b36f0b9340 |
revert accidental commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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2c027162a2 |
Use proper hostname for downloading sound files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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395e431ab6 |
Fix handling of DAHDI_EVENT_REMOVED event to prevent crash in do_monitor.
After talking to rmudgett about some of his recent iflist locking changes, it was determined that the only place that would destroy a channel without being explicitly to do so was in handle_init_event. The loop to walk the interface list has been modified to wait to destroy the channel until the dahdi_pvt of the channel to be destroyed is no longer needed. (closes issue #15378) Reported by: samy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218401 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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12f92cf9c6 |
Don't say "Please try again" if we don't give the user another chance to try again.
(issue #15055, SWP-129) Reported by: jthurman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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903460b848 |
Ensure we don't pickup ourselves when doing pickup by exten.
(closes issue #15100) Reported by: lmsteffan Patches: (modified) pickup.patch uploaded by lmsteffan (license 779) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@218223 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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269854a47d |
Don't ring another channel, if there's not enough time for a queue member to answer.
(Fixes AST-228) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217989 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fb591b9f93 |
Backport realtime fix to 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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92acf5ac29 |
IAX2 encryption regression
The IAX2 Call Token security patch inadvertently broke the use of encryption due to the reorganization of code in the socket_process() function. When encryption is used, an incoming full frame must first be decrypted before the information elements can be parsed. The security release mistakenly moved IE parsing before decryption in order to process the new Call Token IE. To resolve this, decryption of full frames is once again done before looking into the frame. This involves searching for an existing callno, checking the pvt to see if encryption is turned on, and decrypting the packet before the internal fields of the full frame are accessed. associated with AST-2009-006 (closes issue #15834) Reported by: karesmakro Patches: iax2_encryption_fix_1.4.diff uploaded by dvossel (license 671) Tested by: dvossel, karesmakro Review: https://reviewboard.asterisk.org/r/355/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217806 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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74d16bb0df |
Reinstate muted that was removed by mistake.
muted doesn't compile any more on os/x, so I have to disable it in order to testcompile other code... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217735 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b546a14a99 |
Remove harmful code that causes endless loops.
Remove code that causes loops in registrations. We have agreed that the patch that this code was part of was bad. I am ripping out the code that causes the issue. putnopvut needs to check the rest of the patch, if it needs to be changed as well. This solves the issue reported in #15540, but needs more work before we close it (as described above). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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b10343fd19 |
When MOH is playing on the channel, announcements sent through the conference are not heard.
(closes issue #14588) Reported by: voipas Patches: 20090716__issue14588__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen, twisted, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@217156 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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8edfe07e6d |
make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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da349b0e75 |
make chan_sip compile under devmode again
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216432 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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05899c19a1 |
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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17 years ago |
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d5cc6f0879 |
Make sure 'start' is always initialized.
This is the same as rev 216222 in trunk but 1.4 is affected as well git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3ce5ac3c0c |
Merged revisions 216262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216262 | russell | 2009-09-04 05:47:37 -0500 (Fri, 04 Sep 2009) | 2 lines Add a plain text version of the IAX2 security document. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216263 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ff3097a592 |
Blocked revisions 216087 via svnmerge
........ r216087 | russell | 2009-09-03 14:37:05 -0500 (Thu, 03 Sep 2009) | 2 lines Fix a typo. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e982c31095 |
Merged revisions 216080 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216085 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5694fcdb14 |
Merged revisions 216005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216005 | russell | 2009-09-03 13:42:24 -0500 (Thu, 03 Sep 2009) | 2 lines Add IAX2 security document related to AST-2009-006. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ed1951d895 |
Merge code associated with AST-2009-006
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216000 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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82b1e162e1 |
Re-send non-100 provisional responses to prevent cancellation
From section 13.3.1.1 of RFC 3261: If the UAS desires an extended period of time to answer the INVITE, it will need to ask for an "extension" in order to prevent proxies from canceling the transaction. A proxy has the option of canceling a transaction when there is a gap of 3 minutes between responses in a transaction. To prevent cancellation, the UAS MUST send a non-100 provisional response at every minute, to handle the possibility of lost provisional responses. (closes issue #11157) Reported by: rjain Tested by: twilson Review: https://reviewboard.asterisk.org/r/315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@215682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fad010ebe7 |
Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names
In general channel names are in the form Foo/Bar-Z, but the channel name could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to truncate the channel name at the last hyphen. (closes issue #15810) Reported by: dhubbard Patches: dw-softhangup-1.4.patch uploaded by dhubbard (license 733) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@215270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fa27e8dffa |
Also unlock the "other" channel, when returning, due to glare.
(closes issue #15787) Reported by: tim_ringenbach Patches: chan_local.diff uploaded by tim ringenbach (license 540) Tested by: tim_ringenbach git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214940 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fafa668b04 |
Modify comment to be a bit more accurate.
We have kept this comment around long enough, that it's pretty clear that we're keeping the code, because changing the code would require a pretty fundamental architectural shift. We've also taken criticism in some quarters, because it was believed that it was referring to the code being nasty. No, the code isn't nasty, just the operation itself is rather odd. Fixed for eternity (probably not). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214701 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4b133920a5 |
Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before.
(closes issue #15714) Reported by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded by tilghman (license 14) Tested by: pprindeville git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7215954ccf |
One more build system change, to make the descriptions look better, if we have better information.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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60fd401064 |
Make autoheader descriptions render correctly in our autoconfig.h file.
(Figured out while working with issue #14906) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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899272c2cc |
ast_write() ignores ast_audiohook_write() results
In ast_write(), if a channel has a list of audiohooks, those lists are written to and the resulting frame is what ast_write() should continue with. The problem was the returned audiohook frame was not being handled at all, and the original frame passed into it did not contain the mixed audio, so essentially audio was being lost. One result of this was chan_spy's whisper mode no longer worked. To complicate the issue, frames passed into ast_write may either be a single frame, or a list of frames. So, as the list of frames is processed in the audiohook_write, the returned frames had to be added to a new list. (closes issue #15660) Reported by: corruptor Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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a45f5e5be0 |
I should always compile before committing...
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214069 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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10a972f98e |
Fix pronunciation of German dates.
(closes issue #15273) Reported by: Benjamin Kluck Patches: say_c.patch uploaded by Benjamin Kluck (license 803) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@214068 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0055593341 |
Improve error message by informing user exactly which function is missing a parethesis.
(closes issue #15242) Reported by: Nick_Lewis Patches: pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790) pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213970 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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679cefec98 |
Use the default runlevels for Debian derivatives, instead of making up our own.
(closes issue #14730) Reported by: pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213899 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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79221dad8d |
Ensure that T.38 INVITEs generated by Asterisk properly result in T.38 being enabled.
(closes issue #15373) Reported by: dcolombo Patches: chan_sip.patch uploaded by mbrancaleoni (license 342) Tested by: dcolombo, mbrancaleoni git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1ea4af21ca |
Permit DEBUG_FD_LEAKS to be used with C++ source files.
(closes issue #15698) Reported by: slavon Patches: 20090817__issue15698.diff.txt uploaded by tilghman (license 14) Tested by: slavon, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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3300ac5b40 |
Clarify queues.conf comments to specify that variables should be set in the dialplan.
(closes issue #15755) Reported by: trendboy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |