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path: root/channels/chan_sip.c
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2010-02-09This commit removes an extra newline in T.38 generated SDP packets. This bug ↵Matthew Nicholson
was caused by the fix introduced in r243860. (closes issue #16766) Reported by: raivisr Patches: t38-sdp-newline-fix1.diff uploaded by mnicholson (license 96) Tested by: raivisr git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-07Remove parsing of constantssrc from reload_config.Mark Michelson
This config option is already handled by the function handle_common_options and it is unnecessary to parse the value again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245230 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-06Remove useless sip options related to hash table size.Mark Michelson
First off, these options weren't actually doing anything. By the time the options were parsed, the peer and dialog containers had already been allocated with their default values. Second, hash table size is something that doesn't really make sense to change in a config file. If a user is that interested in changing the hashtable size, he can modify the source itself. I have removed the parsing of the hash_peer, hash_user, and hash_dialog options. I have removed the hash_user_size variable altogether since it is not used at all. I also changed hash_peer_size and hash_dialog_size to be constant, and have changed the symbols to be in all caps as constants typically are. I have also removed the entire section in sip.conf.sample regarding configurable hashtable sizes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-05fixes issue with sip registry not having correct default expiryDavid Vossel
default expiry was not being set correctly for a registry object. Thanks to ebroad for reporting the issue and testing the patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-04parse_moved_contact tries to parse contact_name twiceDavid Vossel
parse_moved_contact attempts to remove a quoted string twice, and the first try wasn't even being done correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-03-----Changes -----David Vossel
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
2010-02-02fixes crash during T.38 negotiation caused by invalid or missing ↵David Vossel
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
2010-01-28Add a missing line terminator for T.38 SDP.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-28Merged revisions 243779 via svnmerge from Russell Bryant
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
2010-01-26RFC compliant uri and display-name encode/decodeDavid Vossel
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
2010-01-22Merged revisions 242226 via svnmerge from Olle Johansson
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
2010-01-15Convert a few places to use ast_calloc_with_stringfields where applicable.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-13SIP Show channelstats fix - use float division to show proper statsOlle Johansson
(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
2010-01-12fixes text support in sdp answerDavid Vossel
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
2010-01-07Change in sip show channels display format allowing more digits for CIDDavid Vossel
(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
2010-01-06Whoa, duplicate setting (dead code).Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-02global_contact_ha was renamed in trunkTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-02Merged revisions 237135 via svnmerge from Olle Johansson
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
2009-12-30Fix compiling with LOW_MEMORY.Jeff Peeler
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
2009-12-29Shut down the SIP session timers more gracefully, in order to prevent a ↵Tilghman Lesher
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
2009-12-22Merged revisions 236062 via svnmerge from David Vossel
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
2009-12-17Remove some old code for going to the 'fax' extension when a T.38 switchover ↵Joshua Colp
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
2009-12-15reverse minor sip registration regressionDavid Vossel
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
2009-12-14Merged revisions 234492 via svnmerge from Olle Johansson
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
2009-12-10Merged revisions 234095 via svnmerge from Tilghman Lesher
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
2009-12-07Merged revisions 233471 via svnmerge from David Vossel
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
2009-12-07Do not reject SDP packets describing only non audio streams.Matthew Nicholson
(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
2009-12-02Do not change the exten string field or rebuild the contact headerMark Michelson
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
2009-12-02Add support for handling the 415 Unsupported media type response like we do ↵Joshua Colp
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
2009-12-02Fix a bug where a scheduled item ID would get retained on registrations in a ↵Joshua Colp
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
2009-12-01More 32->64 bit codec conversions.Tilghman Lesher
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
2009-11-30Another round of UDPTL stack fixes/improvements:Kevin P. Fleming
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
2009-11-30When receiving SDP that matches the version of the last one do not treat it ↵Joshua Colp
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
2009-11-30Fix a bug where an immediate masquerade would cause a queued unhold frame to ↵Joshua Colp
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
2009-11-23Change fax detection in chan_sip so it behaves as one would expect.Joshua Colp
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
2009-11-23Merged revisions 230839 via svnmerge from Kevin P. Fleming
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
2009-11-23Merged revisions 230772 via svnmerge from Kevin P. Fleming
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
2009-11-13Merged revisions 230144 via svnmerge from Joshua Colp
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
2009-11-13Display a list of channel variables in each channel-oriented event.Tilghman Lesher
(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
2009-11-13Fix T.38 negotiation regression introduced with the SDP parser changes.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-10Reverted revision 201717.Matthew Nicholson
(closes issue 0016175) Reported by: paul-tg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-06Merged revisions 228547 via svnmerge from Joshua Colp
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
2009-11-04Modify the SDP parsing code to parse session and media level items separately.Matthew Nicholson
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
2009-11-04Merged revisions 227700 via svnmerge from Joshua Colp
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
2009-11-04Two other trunk build fixes (reported by seanbright on #asterisk-dev)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03Fixed a spelling error in the q850 reason header option in the output of sip ↵Matthew Nicholson
show settings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03user.conf entries in SIP were not having their peer type set.David Vossel
(closes issue #16120) Reported by: jsmith git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-03Merged revisions 227088 via svnmerge from Olle Johansson
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
2009-11-02SIP channel name uniquenessDavid Brooks
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
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