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${ noResults }
5661 Commits (cda010c3b7923058fefcea56a1a184bcad7073f1)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Joshua Colp
|
a4a9815fe2 |
When a device registers to use it is entirely possible that they may be in use, so tell the core that we don't know the devstate and have it ask us for it.
(closes issue #13525) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
a039a65656 |
Merged revisions 162804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162804 | file | 2008-12-10 15:01:17 -0400 (Wed, 10 Dec 2008) | 6 lines Fix subscription based MWI up a bit. We only want to put sip: at the beginning of the URI if it is not already there and revert code to ignore destination check if subscribing for MWI. (closes issue #12560) Reported by: vsauer Patches: patch001.diff uploaded by ramonpeek (license 266) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162805 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
02ce4faaeb |
Merged revisions 162738 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162738 | file | 2008-12-10 13:50:43 -0400 (Wed, 10 Dec 2008) | 6 lines When a SIP peer unregisters set the expiry time back to 0 so that the 200 OK contains an expires of 0. (closes issue #13599) Reported by: hjourdain Patches: chan_sip.c.diff uploaded by hjourdain (license 583) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162739 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
d659ec3cd2 |
Merged revisions 162663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162663 | mmichelson | 2008-12-10 10:24:56 -0600 (Wed, 10 Dec 2008) | 11 lines Revert fix for issue 13570. It has caused more problems than it helped to fix. (closes issue #13783) Reported by: navkumar (closes issue #14025) Reported by: ffs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162664 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
d8c152f7f0 |
When transmitting a register set the socket port to the local one for the transport being used, not the port for the remote server.
(closes issue #13633) Reported by: performer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
135bb29ba6 |
Finish conversion to using ARRAY_LEN and remove it as a janitor project.
(closes issue #14032) Reported by: bkruse Patches: 14032.patch uploaded by bkruse (license 132) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
ac12d0d4ce |
Merged revisions 161725 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161725 | file | 2008-12-08 13:52:10 -0400 (Mon, 08 Dec 2008) | 6 lines Make the usereqphone option work again. (closes issue #13474) Reported by: mmaguire Patches: 20080912_bug13474.diff uploaded by mmaguire (license 571) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161726 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Nicholson
|
8b77d66a61 |
Fix a crash that can occur on a transfer in chan_sip when attempting to collect
rtp stats. (closes issue #13956) Reported by: chris-mac Tested by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
|
f6dda1e544 |
Add the ability to play a courtesy tone to the transfer target in a native SIP attended transfer by setting the variable ATTENEDED_TRANSFER_COMPLETE_SOUND.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Eliel C. Sardanons
|
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 |
Dwayne M. Hubbard
|
f9b6507796 |
If 'faxdetect=yes' in sip.conf, switch to a 'fax' extension (if it exists) after T38 is negotiated.
Terry Wilson created the original patch for this functionality, which I slightly modified and added the faxdetect=yes|no configuration option. This patch is only for T38 fax detection and does not do anything for G711 over SIP fax detection. By default, this option is disabled. Reviewboard: http://reviewboard.digium.com/r/69/ This functionality is for issue AST-140. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Michiel van Baak
|
e219598843 |
Add debug flag so skinny debug will show information about packets.
We dont want to scare users with this, so we added a devmode compile flag (closes issue #13952) Reported by: wedhorn Patches: packetdebug3.diff uploaded by wedhorn (license 30) Tested by: mvanbaak, wedhorn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160938 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
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d17d9b2e30 |
- iax2-provision was not freeing iax_templates structure when unloading the chan_iax2.so module.
- Move the code to start using the LIST macros. Review: http://reviewboard.digium.com/r/72 (closes issue #13232) Reported by: eliel Patches: iax2-provision.patch.txt uploaded by eliel (license 64) (with minor changes pointed by Mark Michelson on review board) Tested by: eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
c9f471ac77 |
Merged revisions 160480 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160480 | tilghman | 2008-12-03 08:09:35 -0600 (Wed, 03 Dec 2008) | 7 lines Jon Bonilla (Manwe) pointed out on the -dev list: "I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160481 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
41a4bdd1a6 |
remove duplicate comment that I accidentally merged
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Jeff Peeler
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b4d8a5b771 |
(closes issue #13786)
Reported by: tzafrir Readding DAHDI_CHECK_HOOKSTATE define that was removed in r134260 which fixes not being able to make outgoing calls on some FXO adapters: http://lists.digium.com/pipermail/asterisk-users/2008-November/thread.html#221553 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160319 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
f96547b0b9 |
Merged revisions 160297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160297 | tilghman | 2008-12-02 11:42:09 -0600 (Tue, 02 Dec 2008) | 10 lines When the text does not match exactly (e.g. RTP/SAVP), then the %n conversion fails, and the resulting integer is garbage. Thus, we must initialize the integer and check it afterwards for success. (closes issue #14000) Reported by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff uploaded by folke (license 626) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160308 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
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 |
Sean Bright
|
12b7311782 |
Silence a build warning. (chan_phone.c:810: warning: value computed is not used)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
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15431e2948 |
Merged revisions 160003 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160003 | russell | 2008-12-01 11:27:30 -0600 (Mon, 01 Dec 2008) | 6 lines Apply some logic used in iax2_indicate() to iax2_setoption(), as well, since they both have the potential to send control frames in the middle of call setup. We have to wait until we have received a message back from the remote end before we try to send any more frames. Otherwise, the remote end will consider it invalid, and we'll get stuck in an INVAL/VNAK storm. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160004 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Kevin P. Fleming
|
887e28d7aa |
incorporates r159808 from branches/1.4:
------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Kevin P. Fleming
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9a7c28cd5a |
we can now build with -Wformat=2, which found a couple of real bugs
because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159774 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
5769e6ea72 |
Don't allow for configuration options to overwrite options
set via channel variables on a reload. (closes issue #13921) Reported by: davidw Patches: 13921.patch uploaded by putnopvut (license 60) Tested by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Steve Murphy
|
c5e64b2ac4 |
Merged revisions 159316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159316 | murf | 2008-11-25 15:41:10 -0700 (Tue, 25 Nov 2008) | 15 lines (closes issue #12694) Reported by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17) Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX fix. the change to cdr.c allows no-answer to percolate up into CDR's, and feels like the right place to locate this fix; if BUSY is done here, no-answer should be, too. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
bb80c835e0 |
Add an option, waitfordialtone, for UK analog lines which do not end a call
until the originating line hangs up. (closes issue #12382) Reported by: one47 Patches: zap-waitfordialtone-trunk.080901.patch uploaded by one47 (license 23) zap-waitfordialtone-bra-1.4.21.2.patch uploaded by fleed (license 463) Tested by: fleed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
fe2c495db6 |
Merged revisions 159269 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r159269 | tilghman | 2008-11-25 15:56:48 -0600 (Tue, 25 Nov 2008) | 7 lines Don't try to send a response on a NULL pvt. (closes issue #13919) Reported by: barthpbx Patches: chan_iax2.c.patch uploaded by eliel (license 64) Tested by: barthpbx ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159276 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
f41f8858cd |
Merged revisions 159246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r159246 | tilghman | 2008-11-25 15:40:28 -0600 (Tue, 25 Nov 2008) | 14 lines Merged revisions 159245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines Regression fix for last security fix. Set the iseqno correctly. (closes issue #13918) Reported by: ffloimair Patches: 20081119__bug13918.diff.txt uploaded by Corydon76 (license 14) Tested by: ffloimair ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
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 |
|
Terry Wilson
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853f21e90d |
Make chan_usbradio compile under dev mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158992 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
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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 |
|
Sean Bright
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7bd3ce358b |
If you enabled 'notifycid' one of the limitations is that the calling channel
is only found if it dialed the extension that was subscribed to. You can now specify 'ignore-context' for the 'notifycid' option in sip.conf which will, as it's value implies, ignore the current context of the caller when doing the lookup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
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74c112a501 |
No need to use a separate structure for this since we can just pass
our sip_pvt pointer in directly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158754 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Michiel van Baak
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4a68fe383a |
dont send reorder tone after a device is hungup if a dialout is abandoned or failed.
Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box (closes issue #13948) Reported by: wedhorn Patches: switch.diff uploaded by wedhorn (license 30) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158694 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Michiel van Baak
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ced8427b09 |
Add Media Source Update to skinny's control2str
(issue #13948) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Michiel van Baak
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58ff098571 |
fix a very occasional core dump in chan_skinny found by wedhorn.
(issue #13948) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Fredrickson
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8cb6ecdd24 |
Fix for #13963. Make physical channel mapping unconfigured default
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158482 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Doug Bailey
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d68e8b8e02 |
Add fix to prevent crash during reload if there is an outstanding MWI registration message pending.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158315 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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95c416df0b |
Use a more expressive constant for a 64-bit scanned int
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158266 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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bd6586e3d7 |
Use some magic constants to get the right size
for this sscanf statement. Thanks Richard! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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4e67fdd3f9 |
Fix the build for 32-bit systems. %lu is only 32-bits
on 32-bit systems, so we need to use %llu instead. Of course %llu is 128-bits on 64-bit systems, so we have to cast to unsigned long long. No harm, but it's sure annoying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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e8aa0e29ce |
Change the remote user agent session version variable
from an int to a uint64_t. This prevents potential comparison problems from happening if the version string exceeds INT_MAX. This was an apparent problem for one user who could not properly place a call on hold since the version in the SDP of the re-INVITE to place the call on hold greatly exceeded INT_MAX. This also aligns with RFC 2327 better since it recommends using an NTP timestamp for the version (which is a 64-bit number). (closes issue #13531) Reported by: sgofferj Patches: 13531.patch uploaded by putnopvut (license 60) Tested by: sgofferj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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3a9c27459e |
Merged revisions 158072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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2d4e3b21ee |
Merged revisions 158071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158071 | mmichelson | 2008-11-20 11:48:42 -0600 (Thu, 20 Nov 2008) | 16 lines We don't handle 4XX responses to BYE well. According to section 15 of RFC 3261, we should terminate a dialog if we receive a 481 or 408 in response to our BYE. Since I am aware of at least one phone manufacturer who may sometimes send a 404 as well, I am being liberal and saying that any 4XX response to a BYE should result in a terminated dialog. (closes issue #12994) Reported by: pabelanger Patches: 12994.patch uploaded by putnopvut (license 60) Closes AST-129 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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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 |
|
Kevin P. Fleming
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8d5deb312b |
Merged revisions 157859 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems. with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course). while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Terry Wilson
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d66a8cd264 |
Fix checking for CONFIG_STATUS_FILEINVALID so that modules don't crash upon trying to parse an invalid config
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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1a4fc71415 |
Merged revisions 157503 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157503 | mmichelson | 2008-11-18 16:47:57 -0600 (Tue, 18 Nov 2008) | 13 lines Add some missing invite state changes necessary in the sip_write function. Not setting the invite state correctly on the call was resulting in the Record application leaving empty files. I also have updated the doxygen comment next to the declaration of the INV_EARLY_MEDIA constant to reflect that we also use this state when we *send* a 18X response to an INVITE. (closes issue #13878) Reported by: nahuelgreco Patches: sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162) Tested by: putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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2ede9a603f |
Based on Russell's advice on the asterisk-dev list, I have
changed from using a global lock in update_call_counter to using the locks within the sip_pvt and sip_peer structures instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157496 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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16efb5c4dd |
* Add a lock to be used in the update_call_counter function.
* Revert logic to mirror 1.4's in the sense that it will not allow the call counter to dip below 0. These two measures prevent potential races that could cause a SIP peer to appear to be busy forever. (closes issue #13668) Reported by: mjc Patches: hintfix_trunk_rev152649.patch uploaded by wolfelectronic (license 586) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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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 |