Age | Commit message (Collapse) | Author | Files | Lines |
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines
Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files.
(closes issue #16251)
Reported by: asgaroth
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@238497 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r238412 | dvossel | 2010-01-07 14:15:27 -0600 (Thu, 07 Jan 2010) | 16 lines
Merged revisions 238411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines
fixes crash in "scheduled_destroy" in chan_iax
A signed short was used to represent a callnumber. This is makes
it possible to attempt to access the iaxs array with a negative
index.
(closes issue #16565)
Reported by: jensvb
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@238430 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r238405 | dvossel | 2010-01-07 14:00:31 -0600 (Thu, 07 Jan 2010) | 8 lines
Change in sip show channels display format allowing more digits for CID
(closes issue #16459)
Reported by: Rzadzins
Patches:
chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@238407 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@237967 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r237319 | tilghman | 2010-01-04 10:20:03 -0600 (Mon, 04 Jan 2010) | 10 lines
Merged revisions 237318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines
It's also possible for the Local channel to directly execute an Application.
Reviewboard: https://reviewboard.asterisk.org/r/452/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@237321 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r237136 | oej | 2010-01-02 10:54:22 +0100 (Lör, 02 Jan 2010) | 10 lines
Merged revisions 237135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines
Release memory of the contact acl before unloading module
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@237138 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r236982 | tilghman | 2009-12-30 15:59:18 -0600 (Wed, 30 Dec 2009) | 16 lines
Merged revisions 236981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines
Don't queue frames to channels that have no means to process them.
(closes issue #15609)
Reported by: aragon
Patches:
20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14)
Tested by: aragon
Review: https://reviewboard.asterisk.org/r/452/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@236984 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r236802 | tilghman | 2009-12-29 17:05:45 -0600 (Tue, 29 Dec 2009) | 7 lines
Shut down the SIP session timers more gracefully, in order to prevent a possible crash.
(closes issue #16452)
Reported by: corruptor
Patches:
20091221__issue16452.diff.txt uploaded by tilghman (license 14)
Tested by: corruptor
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@236803 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r236063 | dvossel | 2009-12-22 11:00:08 -0600 (Tue, 22 Dec 2009) | 18 lines
Merged revisions 236062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines
fixes issue with p->method incorrectly set to ACK
It is possible for a second ACK to come in for a retransmitted message.
If an ack does not match an unacked message in our queue, restore the previous
p->method as this ACK is completely ignored.
(closes issue #16295)
Reported by: omolenkamp
Patches:
issue16295_v2.diff uploaded by dvossel (license 671)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@236065 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
reverses the changes caused by issue #15539. The
issue reported was expected behavior.
(issue #15539)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@235135 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r234526 | oej | 2009-12-14 11:46:20 +0100 (Mån, 14 Dec 2009) | 16 lines
Merged revisions 234492 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r234492 | oej | 2009-12-14 11:16:00 +0100 (Mån, 14 Dec 2009) | 8 lines
Stop sending 183's after call hangup.
There where still cases where the 183 keep-alive mechanism would not stop
sending 183's even though the Asterisk server had sent a final reply to
the invite.
EDVX-28
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@234533 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r234129 | tilghman | 2009-12-10 10:24:26 -0600 (Thu, 10 Dec 2009) | 16 lines
Merged revisions 234095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r234095 | tilghman | 2009-12-10 10:08:20 -0600 (Thu, 10 Dec 2009) | 9 lines
When we receive no response at all to our INVITE, allow the channel to be destroyed.
(closes issue #15627)
Reported by: falves11
Patches:
20091209__issue15627__1.6.0.diff.txt uploaded by tilghman (license 14)
20091209__issue15627__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: falves11
Review: https://reviewboard.asterisk.org/r/446/
(closes issue #15716)
Reported by: dant
(closes issue #16270)
Reported by: corruptor
(closes issue #15356)
Reported by: falves11
(issue #16382)
Reported by: lftsy
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@234132 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r233472 | dvossel | 2009-12-07 12:08:46 -0600 (Mon, 07 Dec 2009) | 15 lines
Merged revisions 233471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r233471 | dvossel | 2009-12-07 12:07:38 -0600 (Mon, 07 Dec 2009) | 9 lines
fixes missing Contact header angle brackets
(closes issue #16298)
Reported by: mgernoth
Patches:
reg_parse_issue_1.4.diff uploaded by dvossel (license 671)
Tested by: dvossel
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@233474 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r233394 | mnicholson | 2009-12-07 10:14:42 -0600 (Mon, 07 Dec 2009) | 8 lines
Do not reject SDP packets describing only non audio streams.
(closes issue #16387)
Reported by: zalex1953
Patches:
media-level-c-fix1.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, zalex1953
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@233395 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r232345 | file | 2009-12-02 12:40:14 -0400 (Wed, 02 Dec 2009) | 7 lines
Add support for handling the 415 Unsupported media type response like we do for a 488 Not acceptable here response.
(closes issue #16186)
Reported by: atis
Patches:
sip_t38_response_415.patch uploaded by atis (license 242)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@232347 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r232230 | file | 2009-12-02 10:54:28 -0400 (Wed, 02 Dec 2009) | 5 lines
Fix a bug where a scheduled item ID would get retained on registrations in a certain scenario
causing code to execute during reload that should not.
(issue AST-263)
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@232231 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r232091 | jpeeler | 2009-12-01 18:45:18 -0600 (Tue, 01 Dec 2009) | 17 lines
Merged revisions 232090 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r232090 | jpeeler | 2009-12-01 18:42:58 -0600 (Tue, 01 Dec 2009) | 10 lines
Do not modify the gain settings on data calls.
(The digital flag actually represents a data call.)
(closes issue #15972)
Reported by: udosw
Patches:
transcap_digital_fix.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@232093 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines
Another round of UDPTL stack fixes/improvements:
1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL stack can
be 'connected' to the endpoint that caused them to be generated.
2) Improve comments (and process) of calculating the far end's maximum IFP size
when redundancy mode is in use for error correction.
3) When an IFP larger than the calculated 'far max IFP' size is presented for
writing, truncate it rather than putting in the buffer and allowing the buffer
to overflow; this will cause the ends to retrain to a lower bit rate that
produces IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to the one endpoint
supplying a T38FaxMaxDatagram value that is improperly calculated and is
too low to be of use; we have configuration options available to override
this behavior.
4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
needed.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@231694 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r231602 | file | 2009-11-30 16:44:30 -0400 (Mon, 30 Nov 2009) | 5 lines
When receiving SDP that matches the version of the last one do not treat it as a fatal error.
(closes issue #16238)
Reported by: seandarcy
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@231604 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r230881 | file | 2009-11-23 09:45:45 -0600 (Mon, 23 Nov 2009) | 7 lines
Change fax detection in chan_sip so it behaves as one would expect.
Internally the way T.38 is negotiated has changed and the option no longer
reflects a behavior that is valid. It will now look for a CNG tone on
received calls and if present send the call to the 'fax' extension. It is
then up to the application or channel to request the switch over to T.38.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230883 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r230877 | kpfleming | 2009-11-23 09:34:16 -0600 (Mon, 23 Nov 2009) | 9 lines
Merged revisions 230839 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r230839 | kpfleming | 2009-11-23 09:09:24 -0600 (Mon, 23 Nov 2009) | 1 line
Correct fix for issue #16268... the reporter's original patch was very close to correct.
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230879 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r230773 | kpfleming | 2009-11-23 08:15:48 -0600 (Mon, 23 Nov 2009) | 12 lines
Merged revisions 230772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r230772 | kpfleming | 2009-11-23 08:13:56 -0600 (Mon, 23 Nov 2009) | 5 lines
Ensure that SDP parsing does not ignore the last line of the SDP.
(closes issue #16268)
Reported by: sgimeno
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230790 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r230726 | dvossel | 2009-11-20 16:35:54 -0600 (Fri, 20 Nov 2009) | 7 lines
fixes iax2 show cache locking error, thanks alecdavis!
(closes issue #16094)
Reported by: alecdavis
Patches:
bug16094.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis, dvossel
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230728 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r230247 | kpfleming | 2009-11-15 11:23:02 -0600 (Sun, 15 Nov 2009) | 12 lines
Merged revisions 230246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r230246 | kpfleming | 2009-11-15 11:19:06 -0600 (Sun, 15 Nov 2009) | 6 lines
Correct mistaken option name in error message.
The configuration option for allowing hosts to make non-token-based calls
is 'calltokenoptional', not 'calltokenignore'. (reported on asterisk-users)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230249 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r230145 | file | 2009-11-13 16:00:44 -0600 (Fri, 13 Nov 2009) | 15 lines
Merged revisions 230144 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r230144 | file | 2009-11-13 16:00:19 -0600 (Fri, 13 Nov 2009) | 8 lines
Respect the maddr parameter in the Via header.
(closes issue #14446)
Reported by: frawd
Patches:
via_maddr.patch uploaded by frawd (license 610)
Tested by: frawd
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230147 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r230039 | file | 2009-11-13 13:44:53 -0600 (Fri, 13 Nov 2009) | 16 lines
Merged revisions 230038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r230038 | file | 2009-11-13 13:44:07 -0600 (Fri, 13 Nov 2009) | 9 lines
Fix a crash caused by two threads thinking they should both free the
chan_local private structure when only one should.
(closes issue #15314)
Reported by: sroberts
Patches:
Issue15314_Move_Nulling_owner.patch uploaded by davidw (license 780)
Tested by: davidw, lottc
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@230041 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r229912 | file | 2009-11-13 09:56:16 -0600 (Fri, 13 Nov 2009) | 2 lines
Fix T.38 negotiation regression introduced with the SDP parser changes.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229914 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r229750 | qwell | 2009-11-12 17:30:10 -0600 (Thu, 12 Nov 2009) | 1 line
Fix mute toggling on OSS channels.
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229751 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r229168 | dvossel | 2009-11-10 11:16:49 -0600 (Tue, 10 Nov 2009) | 15 lines
Merged revisions 229167 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r229167 | dvossel | 2009-11-10 11:15:57 -0600 (Tue, 10 Nov 2009) | 9 lines
don't crash on log message in solaris
AST-2009-006
(closes issue #16206)
Reported by: bklang
Tested by: bklang
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229233 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #16175)
Reported by: paul-tg
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229099 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r229015 | twilson | 2009-11-09 16:50:22 -0600 (Mon, 09 Nov 2009) | 8 lines
Don't crash when bridge->tech_pvt == NULL
This is a similar solution to what is in place for chan_agent
(closes issue #16003)
Reported by: atis
Tested by: twilson
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229016 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
In sip_hangup we attempted to lock p->owner after we set it to NULL.
Thanks to fhackenberger for reporting the issue and submitting a patch.
(closes issue 0015848)
Reported by: fhackenberger
Patches:
digium_bug_0015848 uploaded by fhackenberger (license 592)
Tested by: fhackenberger, lmadsen, TomS, shin-shoryuken, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@229014 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r228548 | file | 2009-11-06 14:37:59 -0400 (Fri, 06 Nov 2009) | 11 lines
Merged revisions 228547 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r228547 | file | 2009-11-06 14:32:58 -0400 (Fri, 06 Nov 2009) | 4 lines
Don't overwrite caller ID name on a trunk with the configured fullname when using users.conf
(issue ABE-1989)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228550 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r227238 | dvossel | 2009-11-03 11:12:52 -0600 (Tue, 03 Nov 2009) | 5 lines
user.conf entries in SIP were not having their peer type set.
(closes issue #16120)
Reported by: jsmith
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228267 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines
Merged revisions 228078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines
chan_misdn Asterisk 1.4.27-rc2 crash
Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested
by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached
a full bt." This patch zeros out an ast_frame.
(closes issue #16041)
Reported by: francesco_r
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228147 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r228080 | qwell | 2009-11-05 13:16:29 -0600 (Thu, 05 Nov 2009) | 15 lines
Merged revisions 228079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r228079 | qwell | 2009-11-05 13:14:25 -0600 (Thu, 05 Nov 2009) | 8 lines
Fix crash on VPB exception when no hardware is present.
(closes issue #14970)
Reported by: tzafrir
Patches:
vpb_exception.diff uploaded by tzafrir (license 46)
Tested by: markwaters
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@228090 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future.
(closes issue #14994)
Reported by: frawd
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227761 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r227712 | file | 2009-11-04 15:20:46 -0400 (Wed, 04 Nov 2009) | 12 lines
Merged revisions 227700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227700 | file | 2009-11-04 15:17:39 -0400 (Wed, 04 Nov 2009) | 5 lines
Fix a security issue where sending a REGISTER with a differing username in the From
URI and Authorization header would reveal whether it was valid or not.
(AST-2009-008)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227723 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227275 | rmudgett | 2009-11-03 11:55:47 -0600 (Tue, 03 Nov 2009) | 4 lines
Make sure the outgoing flag is cleared if a new channel fails to get created for outgoing calls.
This is the relevant portion of asterisk/trunk -r226648
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227279 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r227167 | file | 2009-11-03 11:37:08 -0400 (Tue, 03 Nov 2009) | 12 lines
Merged revisions 227166 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227166 | file | 2009-11-03 11:36:16 -0400 (Tue, 03 Nov 2009) | 5 lines
Fix a bug where an RPID header could be generated with a blank username in the URI.
(closes issue #15909)
Reported by: kobaz
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227169 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r227091 | oej | 2009-11-03 12:11:15 +0100 (Tis, 03 Nov 2009) | 15 lines
Merged revisions 227088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r227088 | oej | 2009-11-03 11:29:59 +0100 (Tis, 03 Nov 2009) | 7 lines
Use proper response code when violating Contact ACL's.
https://reviewboard.asterisk.org/r/415/
Thanks kpfleming for a quick review.
(EDVX-003)
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@227155 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.
(closes issue #15152)
Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@226977 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r226532 | file | 2009-10-29 15:13:42 -0300 (Thu, 29 Oct 2009) | 13 lines
Merged revisions 226531 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines
Add an option to enabling passing music on hold start and stop requests through instead of
acting on them in chan_local.
(closes issue #14709)
Reported by: dimas
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@226534 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r225912 | jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
ACL check not present for verifying SIP INVITEs
The ACL check in check_peer_ok was missing and has now been restored. The
missing check allowed for calls to be made on prohibited networks where an ACL
was defined in sip.conf and the allowguest option was set to off. See the AST
security advisory below for more information.
Merge code associated with AST-2009-007.
(closes issue #16091)
Reported by: thom4fun
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225913 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r225650 | dvossel | 2009-10-23 09:41:50 -0500 (Fri, 23 Oct 2009) | 3 lines
Fixes an iterator memory leak and uninitialized memory
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225652 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
........
r225445 | dvossel | 2009-10-22 14:55:51 -0500 (Thu, 22 Oct 2009) | 50 lines
SIP TCP/TLS: move client connection setup/write into tcp helper thread, various related locking/memory fixes.
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
........
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225490 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines
Merged revisions 225243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines
IAX2: VNAK loop caused by signaling frames with no destination call number
It is possible for the PBX thread to queue up signaling frames before
a destination call number is received. This can result in signaling
frames being sent out with no destination call number. Since recent
versions of Asterisk require accurate destination callnumbers for all
Full Frames, this can cause a VNAK loop to occur. To resolve this
no signaling frames are sent until a destination callnumber is received,
and destination call numbers are now only required for iax_pvt matching
when the frame is an ACK.
Review: https://reviewboard.asterisk.org/r/413/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225309 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines
Merged revisions 225032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines
IAX/SIP shrinkcallerid option
The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
and '-' from the string. This means values such as 555.5555 and
test-test result in 555555 and testtest. There are instances,
such as Skype integration, where a specific value is passed via
caller id that must be preserved unmodified. This patch makes
the shrinking of caller id optional in chan_sip and chan_iax in
order to support such cases. By default this option is on to
preserve previous expected behavior.
(closes issue #15940)
Reported by: dimas
Patches:
v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/408/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@225062 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@224336 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
https://origsvn.digium.com/svn/asterisk/trunk
................
r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines
Merged revisions 224330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines
Fix stale caller id data from being reported in AMI NewChannel event
The problem here is that chan_dahdi is designed in such a way to set
certain values in the dahdi_pvt only once. One of those such values
is the configured caller id data in chan_dahdi.conf. For PRI, the
configured caller id data could be overwritten during a call. Instead
of saving the data and restoring, it was decided that for all non-analog
channels it was simply best to not set the configured caller id in the
first place and also clear it at the end of the call.
(closes issue #15883)
Reported by: jsmith
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@224333 f38db490-d61c-443f-a65b-d21fe96a405b
|