mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
5692 Commits (d13e236488984e24c044dc93589c0ec1e88ca118)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Sean Bright
|
e1f941d7f6 |
Mostly just whitespace, but also convert 'CVS' to 'SVN' in a couple
places and fix a few typos I found in the CODING_GUIDELINES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167061 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
4412a4ba24 |
Change some incorrect syntax for pri set debug and correct
an off-by-one error in ss7 set debug command git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167021 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
7cb7920e19 |
Merged revisions 166953 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166953 | tilghman | 2008-12-31 13:20:35 -0600 (Wed, 31 Dec 2008) | 5 lines Also inherit the musiconhold class. (Closes #14153) Reported by: Jerry Geis, via the users list. Patch by: me (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
e697dfb43f |
Merged revisions 166772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166772 | russell | 2008-12-28 09:13:48 -0600 (Sun, 28 Dec 2008) | 4 lines Use strncat() instead of an sprintf() in which source and target buffers overlap http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
|
8664b9111a |
There is no section 22.2.2 in rfc 3261. I believe 26.2.2 is what was meant:
Note that in the SIPS URI scheme, transport is independent of TLS,
and thus "sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
17 years ago |
|
Tilghman Lesher
|
8c2030b489 |
Allow semicolons and extended characters in user-specified SIP headers.
(closes issue #14110) Reported by: gork Patches: 20081222__bug14110__2.diff.txt uploaded by Corydon76 (license 14) Tested by: gork, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
61c0d20d20 |
Merged revisions 166380 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r166380 | mmichelson | 2008-12-22 14:56:29 -0600 (Mon, 22 Dec 2008) | 36 lines Fix a deadlock relating to channel locks and autoservice It has been discovered that if a channel is locked prior to a call to ast_autoservice_stop, then it is likely that a deadlock will occur. The reason is that the call to ast_autoservice_stop has a check built into it to be sure that the thread running autoservice is not currently trying to manipulate the channel we are about to pull out of autoservice. The autoservice thread, however, cannot advance beyond where it currently is, though, because it is trying to acquire the lock of the channel for which autoservice is attempting to be stopped. The gist of all this is that a channel MUST NOT be locked when attempting to stop autoservice on the channel. In this particular case, the channel was locked by a call to ast_read. A call to ast_exists_extension led to autoservice being started and stopped due to the existence of dialplan switches. It may be that there are future commits which handle the same symptoms but in a different location, but based on my looks through the code, it is very rare to see a construct such as this one. (closes issue #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded by putnopvut (license 60) Tested by: rtrauntvein Review: http://reviewboard.digium.com/r/107/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Fredrickson
|
775033301a |
Add configuration support for half_full DAHDI buffer policy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
654ea55a65 |
Numerous documentation updates.
(closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
4534957e81 |
Call proxy_update so that the IP address gets populated. Sending stuff to 0.0.0.0 is silly!
(closes issue #14055) Reported by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Nicholson
|
91192e30c5 |
This patch adds a new 'ignoresdpversion' option to sip.conf. When this is
enabled (either globally or for a specific peer), chan_sip will treat any SDP data it receives as new data and update the media stream accordingly. By default, Asterisk will only modify the media stream if the SDP session version received is different from the current SDP session version. This option is required to interoperate with devices that have non-standard SDP session version implementations (observed by toc on the bug tracker with Microsoft OCS which always uses 0 as the session version). http://reviewboard.digium.com/r/94/ (closes issue #13958) Reported by: toc Tested by: toc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165180 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
1d2b4e7a02 |
Merged revisions 164977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164977 | mmichelson | 2008-12-16 17:04:27 -0600 (Tue, 16 Dec 2008) | 7 lines After looking through SIP registration code most of the day, this is one of the few things I could find that was just plain wrong. Even though it probably isn't possible for it to happen, it seems weird to have code that checks if a pointer is NULL and then immediately dereferences that pointer if it was NULL. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Terry Wilson
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2e59fce6d8 |
Make a note of the feature request in bug #11157 as per the reporter and oej, and suspend the bug since no one seems to be keen on implementing it any time soon.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164941 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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fd62012a31 |
Qualify trumps poke per lmadsen.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164814 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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92a4edc593 |
Add configuration options for finer control over how Asterisk handles having to poke all peers at seemingly the same time.
(closes issue #13217) Reported by: cervajs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164809 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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36b1d08dc0 |
Merged revisions 164672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164672 | russell | 2008-12-16 09:56:37 -0600 (Tue, 16 Dec 2008) | 11 lines Fix a memory leak related to the use of the "setvar" configuration option. The problem was that these variables were being appended to the list of vars on the sip_pvt every time a re-registration or re-subscription came in. Since it's just a waste of memory to put them there unless the request was an INVITE, then the fix is to check the request type before copying the vars. (closes issue #14037) Reported by: marvinek Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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ec6e4d2f60 |
When using externhost make sure the port gets set to the bindaddr port if one was not specified in the externhost value itself.
(closes issue #13634) Reported by: performer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164659 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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f546396c0f |
Fix usage of the DAHDI_VMWI ioctl.
(closes issue #14090) Reported by: alecdavis Patches: chan_dahdi.VMWI_ioctl.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164602 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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9e65283794 |
Open a timer before loading configuration so that the trunking configuration option
will take effect. (closes issue #14082) Reported by: seandarcy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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3ef07d4fd4 |
Fix log message to refer to the generic timing interface, not DAHDI specifically
(inspired by issue #14082) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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42e26ee700 |
Revert ast_str opacity in chan_sip for now, since something wasn't quite right
in the merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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ae30bbf43d |
Merged revisions 164350 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164350 | file | 2008-12-15 14:11:21 -0400 (Mon, 15 Dec 2008) | 6 lines Do not try to unlock a non-existant channel if the transfer fails. (closes issue #13800) Reported by: dwagner Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety (license 608) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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808a5fda59 |
Fix a couple more build issues related to ast_str_opaque
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164309 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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c8223fc957 |
Merge ast_str_opaque branch (discontinue usage of ast_str internals)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Steve Murphy
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0692660afc |
demote always-appearing debug message (for certain boards) to ast_debug lev 3 msg instead
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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90e65dc7d3 |
Rename a number of tcptls_session variables. There are no functional changes here.
The name "ser" was used in a lot of places. However, it is a relic from when the struct was a server_instance, not a session_instance. It was renamed since it represents both a server or client connection. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163670 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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4dde380315 |
Fix a small race condition in sip_tcp_locate().
We must increase the reference count on the tcptls_session _before_ unlocking the thread list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163667 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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4295303c56 |
Resolve crashes when using SIP TCP/TLS with qualify.
The problem was a reference count error on the tcptls_session structure. (closes issue #13989) Reported by: Nugget git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163642 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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44b93b6859 |
When a device registers we need to unlink them (if linked) from the peers_by_ip container and link them back in since their IP address has changed. This would have manifested itself if you configured a new device (as type=peer), registered, and then tried to place a call from the device. Since the peer was not linked into the peers_by_ip container it would have never been found.
(closes issue #13811) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163629 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Michiel van Baak
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8c3a7a28bf |
Fix codec capability setup in chan_skinny
Behaviour now is that general codec config flows to default_line and default_device. [devices] stuff amends default_device and similar for [lines]. These are copied to individual device and line as they are created. Added confcapability and confprefs for the configured stuff which doesn't change as device and so on are connected. prefs are based on line prefs if they exist, else the device prefs are used (prefs identifies codec order). (closes issue #13806) Reported by: pj Patches: codecs.diff uploaded by wedhorn (license 30) Tested by: pj and me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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035a7552d6 |
Since chan_sip is callback devicestate driven do not pass in actual states, pass in unknown so we get asked. Additionally do not pass in an actual device state value in ast_setstate since the channel may be callback driven.
(closes issue #13525) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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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 |
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Joshua Colp
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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 |
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Joshua Colp
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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 |
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Mark Michelson
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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 |
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Joshua Colp
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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 |
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Joshua Colp
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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 |
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Joshua Colp
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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 |
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Matthew Nicholson
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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 |
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Terry Wilson
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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
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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
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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 |
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Michiel van Baak
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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 |
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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 |
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Tilghman Lesher
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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
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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 |
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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 |
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Tilghman Lesher
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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 |
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Tilghman Lesher
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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 |
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Sean Bright
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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 |