mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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2801 Commits (a9932363a9560dee6b5733a33d789f4ef867d8a5)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David Vossel
|
a9932363a9 |
-----Changes -----
New files - channels/sip/sip.h – A new header for shared #define, enum, and struct definitions. - channels/sip/include/sip_utils.h – sip util functions shared among the all the sip APIs - channels/sip/include/config_parser.h – sip config-parser API - channels/sip/config_parser.c – Contains sip.conf parsing helper functions with unit tests. - channels/sip/include/reqresp_parser.h – sip request response parser API - channels/sip/reqresp_parser.c – Contains sip request and response parsing helper functions with unit tests. New Unit Tests - sip_parse_uri_test - sip_parse_host_test - sip_parse_register_line_test Code Refactoring - All reusable #define, enum, and struct definitions were moved out of chan_sip.c into sip.h. During this process formatting changes were made to comments in both sip.h and chan_sip.c in order to better adhere to the coding guidelines. - The beginnings of three new sip APIs, sip-utils.h, config-parser.h, reqresp-parser.h using existing chan_sip.c functions. - parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c along with unit tests for both functions. - sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to config-parser.c along with unit tests for both functions. Changes to parse_uri() -removal of the options parameter. It was never used and did not behave correctly. -additional check for [?header] field. When this field was present, the transport type was not being set correctly. ----- Overview ----- This patch is introduced with the hope that unit tests for all our sip parsing functions will be written soon. chan_sip is a huge file, and with the addition of each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing we begin refactoring chan_sip, starting with the parsing functions. With each parsing function we move into a separate helper file, a unit test should accompany it. I've attempted to lay down the ground work for this change by creating two new parser helper files (config-parser.c and reqresp-parser.c) and moving all shared structs, enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything in Asterisk using unit tests, but string parsing is one area where unit tests make the most sense. By beginning to restructure the code in this way, chan_sip not only becomes less bloated, but Asterisk as a whole will become more stable. Review: https://reviewboard.asterisk.org/r/477/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
36bbf8f902 |
fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field
AST-2010-001 (closes issue #16634) Reported by: krn (closes issue #16724) Reported by: barthpbx (closes issue #16517) Reported by: bklang (closes issue #16485) Reported by: elsto git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244443 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
|
5766b06ad4 |
Add a missing line terminator for T.38 SDP.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Russell Bryant
|
9ae1efe42c |
Merged revisions 243779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010) | 2 lines Fix a bogus third argument to ast_copy_string(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243780 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
d16b89be17 |
RFC compliant uri and display-name encode/decode
1. URI Encoding
This patch changes ast_uri_encode()'s behavior when doreserved is enabled.
Previously when doreserved was enabled only a small set of reserved
characters were encoded. This set was comprised primarily of the reserved
characters defined in RFC3261 section 25.1, but contained other characters as
well. Rather than only escaping the reserved set, doreserved now escapes
all characters not within the unreserved set as defined by RFC 3261 and
RFC 2396. Also, the 'doreserved' variable has been renamed to 'do_special_char'
in attempts to avoid confusion.
When doreserve is not enabled, the previous logic of only encoding the
characters <= 0X1F and > 0X7f remains, except for the '%' character, which
must always be encoded as it signifies a HEX escaped character during the decode
process.
2. URI Decoding: Break up URI before decode.
In chan_sip.c ast_uri_decode is called on the entire URI instead of it's
individual parts after it is parsed. This is not good as ast_uri_decode
can introduce special characters back into the URI which can mess up parsing.
This patch resolves this by not decoding a URI until parsing is completely
done. There are many instances where we check to see if pedantic checking
is enabled before we decode a URI. In these cases a new macro,
SIP_PEDANTIC_DECODE, is used on the individual parsed segments of the URI
rather than constantly putting if (pedantic) { decode() } checks everywhere
in the code. In the areas where ast_uri_decode is not dependent upon
pedantic checking this macro is not used, but decoding is still moved to
each individual part of the URI. The only behavior that should change from
this patch is the time at which decoding occurs.
Since I had to look over every place URI parsing occurs to create this
patch, I found several places where we use duplicate code for parsing.
To consolidate the code, those areas have updated to use the parse_uri()
function where possible.
3. SIP display-name decoding according to RFC3261 section 25.
To properly decode the display-name portion of a FROM header, chan_sip's
get_calleridname() function required a complete re-write. More information
about this change can be found in the comments at the beginning of this function.
4. Unit Tests.
Unit tests for ast_uri_encode, ast_uri_decode, and get_calleridname() have been
written. This involved the addition of the test_utils.c file for testing the
utils api.
(closes issue #16299)
Reported by: wdoekes
Patches:
astsvn-16299-get_calleridname.diff uploaded by wdoekes (license 717)
get_calleridname_rewrite.diff uploaded by dvossel (license 671)
Tested by: wdoekes, dvossel, Nick_Lewis
Review: https://reviewboard.asterisk.org/r/469/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
16 years ago |
Olle Johansson
|
64b76fa41a |
Merged revisions 242226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3 lines Initialize notify_types to NULL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Sean Bright
|
e612d87695 |
Convert a few places to use ast_calloc_with_stringfields where applicable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Olle Johansson
|
c15a667733 |
SIP Show channelstats fix - use float division to show proper stats
(closes issue #15819) Reported by: klaus3000 Patches: asterisk-sip-show-channelstats-trunk.txt uploaded by klaus3000 (license 65) Tested by: klaus3000, oej This patch is for trunk only and will be blocked in 1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
fa0ef8031d |
fixes text support in sdp answer
The code that handled setting 'm=text' in the sdp was not executing in the correct order. The check to see if text was needed came after the check to add 'm=text' to the sdp, this resulted in 'm=text' always being set to 0 because it looked like text was never required. (closes issue #16457) Reported by: peterj Patches: textportinsdp.diff uploaded by peterj (license 951) issue16457.diff uploaded by dvossel (license 671) Tested by: peterj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
084e235a8c |
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: https://origsvn.digium.com/svn/asterisk/trunk@238405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
|
82f998dcd4 |
Whoa, duplicate setting (dead code).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
0078b3bc5c |
global_contact_ha was renamed in trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Olle Johansson
|
6da31b48d7 |
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: https://origsvn.digium.com/svn/asterisk/trunk@237136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
|
1a219ad725 |
Fix compiling with LOW_MEMORY.
Modified handle_verbose to be LOW_MEMORY aware, removed old RTP related code in chan_sip. (closes issue #16381) Reported by: michael_iedema Patches: ast_complete_source_filename.patch uploaded by michael iedema (license 942) modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236893 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
5d2b47ffb8 |
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: https://origsvn.digium.com/svn/asterisk/trunk@236802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
b74201f2b6 |
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: https://origsvn.digium.com/svn/asterisk/trunk@236063 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Joshua Colp
|
ff0f861383 |
Remove some old code for going to the 'fax' extension when a T.38 switchover occurs. This would have
already happened when we detected the CNG tone so this was basically a noop. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
181f617fd7 |
reverse minor sip registration regression
A registration regression caused by a code tweak in (issue #14331) and a bug fix in (issue #15539) caused some sip registration config entries to be constructed incorrectly. Origially issue #14331 contained the code tweak as well as a bug fix, but since the issue was reported as a tweak the bug fix portion was moved into issue #15539. Both the tweak and the bug fix contained minor incorrect logic that resulted in some SIP registrations to fail. (issue #14331) (issue #15539) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235132 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Olle Johansson
|
2d49d547ed |
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: https://origsvn.digium.com/svn/asterisk/trunk@234526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
84678fc77d |
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: https://origsvn.digium.com/svn/asterisk/trunk@234129 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
86dc66625c |
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: https://origsvn.digium.com/svn/asterisk/trunk@233472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
|
3069bab67c |
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: https://origsvn.digium.com/svn/asterisk/trunk@233394 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
|
74b388ea4a |
Do not change the exten string field or rebuild the contact header
on an inbound sip_pvt if the outbound call is redirected. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Joshua Colp
|
6a7f37e07d |
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: https://origsvn.digium.com/svn/asterisk/trunk@232345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
23781604aa |
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: https://origsvn.digium.com/svn/asterisk/trunk@232230 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
f59fe83c56 |
More 32->64 bit codec conversions.
In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Kevin P. Fleming
|
5ba2b689b2 |
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: https://origsvn.digium.com/svn/asterisk/trunk@231692 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
c899add7f9 |
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: https://origsvn.digium.com/svn/asterisk/trunk@231602 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
28b009b266 |
Fix a bug where an immediate masquerade would cause a queued unhold frame to get lost. Now we just
indicate unhold directly after the masquerade is complete. (issue ABE-2011) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
60e10aba46 |
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: https://origsvn.digium.com/svn/asterisk/trunk@230881 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
|
fe1ebc8d46 |
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: https://origsvn.digium.com/svn/asterisk/trunk@230877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
|
1f759eddfa |
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: https://origsvn.digium.com/svn/asterisk/trunk@230773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
8ba56154bb |
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: https://origsvn.digium.com/svn/asterisk/trunk@230145 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
5e2aa190fe |
Display a list of channel variables in each channel-oriented event.
(Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
b3b6537e71 |
Fix T.38 negotiation regression introduced with the SDP parser changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
|
2cc2bade4b |
Reverted revision 201717.
(closes issue 0016175) Reported by: paul-tg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
c205958f4c |
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: https://origsvn.digium.com/svn/asterisk/trunk@228548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
|
b3bd43366f |
Modify the SDP parsing code to parse session and media level items separately.
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 Tested by: frawd, mnicholson, file Review: https://reviewboard.asterisk.org/r/414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
45f0f0cfef |
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: https://origsvn.digium.com/svn/asterisk/trunk@227712 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
2bbda7a7c8 |
Two other trunk build fixes (reported by seanbright on #asterisk-dev)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
d8e0c58437 |
Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
|
4b69c3af69 |
Fixed a spelling error in the q850 reason header option in the output of sip show settings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227298 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
8cd25fc043 |
user.conf entries in SIP were not having their peer type set.
(closes issue #16120) Reported by: jsmith git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227238 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Olle Johansson
|
ede3699c6e |
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: https://origsvn.digium.com/svn/asterisk/trunk@227091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Brooks
|
2c4d3b3168 |
SIP channel name uniqueness
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: https://origsvn.digium.com/svn/asterisk/trunk@226974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
|
93e43578ec |
This patch adds support for a draft proposal for adding Q.850 reason headers to sip messages.
(closes issue #13385) Reported by: adomjan Patches: sip.conf.sample-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) CHANGES-trunk20090929-reason_q850.patch uploaded by adomjan (license 487) chan_sip.c-trunk20090929-reason_q850_atoi_fix.patch uploaded by adomjan (license 487) sip-q850-hangupcause1.diff uploaded by mnicholson (license 96) Tested by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Joshua Colp
|
5825f68e8b |
Add support for receiving unsolicited MWI NOTIFY messages.
This change adds a configuration option to SIP peers, unsolicited_mailbox, which configures a virtual mailbox to use for received new/old MWI information. This virtual mailbox can then be used by any device supporting MWI. (closes issue #13028) Reported by: AsteriskRocks Patches: bug_13028_chan_sip_external_mwi_20090707.patch uploaded by cmaj (license 830) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
|
ea8b54fb9d |
Fix building in REF_DEBUG mode.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
|
ec0a1882c9 |
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: https://origsvn.digium.com/svn/asterisk/trunk@225912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
|
fb0196fce6 |
Improve performance of pedantic mode dialog searching in chan_sip.
This patch changes chan_sip to use the new astobj2 OBJ_MULTIPLE iterator support to make pedantic mode dialog searching in find_call() not require a linear search of all dialogs in the list of dialogs. This patch does *not* change the dialog matching logic (more on that later), just improves the searching performance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |