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2009-03-05Merge phase 1 support for the new bridging architecture.Joshua Colp
This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04Resolve object matching issues related to the removal of the sip_user object.Russell Bryant
Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04Allow for "magic" pickups to work when we wish to ignore the contextMark Michelson
When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Please prefix default values with DEFAULTOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02Do not try to remove a registration scheduled item if the scheduler context ↵Joshua Colp
has already been destroyed. (closes issue #14580) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-01Properly free memory and remove scheduler entries when a transmission ↵Mark Michelson
failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27Add reload support to chan_skinny.Michiel van Baak
Special thanks goes to DEA who had to redo this patch twice because we first put unload/load support in and later redid the way we configure devices and lines. (closes issue #10297) Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA (license 3) With mods by me based on feedback from wedhorn and Russell and seanbright Tested by: DEA, mvanbaak, pj Review: http://reviewboard.digium.com/r/130/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26IAX2 prune realtime, minor tweak to last fixDavid Vossel
A return statement was missing which caused unexpected cli output. issue #14479 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26IAX2 prune realtime fixDavid Vossel
Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24Allows manager command to see if IAX link is trunked and encrypted. Displays ↵David Vossel
what kind of encryption is enabled as well. Manager command "iaxpeers" now shows if a link is trunked and encrypted. Instead of encryption saying simply "yes" or "no", it now displays what type of encryption is enabled and if keyrotation is on or not. (closes issue #14427) Reported by: snuffy Patches: iax_show_trunks.diff uploaded by snuffy (license 35) 2009022200_iax2_show_trunkencryption.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, dvossel, snuffy Review: http://reviewboard.digium.com/r/173/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24Merged revisions 178205 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178205 | file | 2009-02-24 11:16:07 -0400 (Tue, 24 Feb 2009) | 9 lines Skip check for extension when subscribing for MWI. Since the remote side is not actually subscribing to a specific extension when subscribing for MWI just skip the check to see if the extension exists. They can't use it to specify the mailbox either since we require configuration of that in sip.conf (closes issue #14531) Reported by: festr ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178213 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-23update the new manager commands in chan_skinny to matchMichiel van Baak
chan_sip's headers. requested by oej. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-23Changes the way keyrotation is enabled by defaultDavid Vossel
Key rotation was enabled by default by setting the global encryption method to IAX_ENCRYPT_KEYROTATE. the problem with this is that if encryption is not enabled, and the encryption method is set to anything except 0, the peer appears to have encryption enabled when issuing a "iax2 show peers". Rather than have the key rotation bit always set by default, it is now only set when an encryption method is enabled. (closes issue #14523) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-22Add a couple of manager commands to chan_skinnyMichiel van Baak
Added: SKINNYdevices SKINNYshowdevice SKINNYlines SKINNYshowline (closes issue #14521) Reported by: mvanbaak Review: http://reviewboard.digium.com/r/170/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21On update, test against the existence of sipregs.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21make chan_sip.c compile on OpenBSD again.Michiel van Baak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177849 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20Set sip_request ast_str data to NULL so ast_str_copy allocates space properlyJeff Peeler
in copy_request (issue #14478) Reported by: erik_dedecker git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18Modify h323 to build against PTLib as well as the older PWLibJeff Peeler
Several changes in PTLib have occurred requiring build time detection. Changes accounted for include the library name change, config option change, install location change, and a boolean type change which is handled by ast_ptlib.h. Also, the sed check has been modified to properly work with autoconf >= 2.62. (closes issue #14224) Reported by: bergolth Patches: asterisk-autoconf-sed.patch uploaded by bergolth (license 661) asterisk-pwlib-v3.patch uploaded by bergolth (license 661) Tested by: jpeeler git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177162 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18Fix ordering of output for a ChannelUpdate manager event.Joshua Colp
(closes issue #14497) Reported by: vinsik Patches: chan_update_fix-chan_sip.c.diff uploaded by vinsik (license 623) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-18T38 faxdetect should jump to the 'fax' extension for incoming calls onlyDwayne M. Hubbard
The previous implementation of T38 faxdetect resulted in both sides of the call jumping to a fax extension when both sides had 't38pt_udptl=yes' and 'faxdetect=yes' in sip.conf and a 'fax' extension in the current context. This revision will jump to a 'fax' extension on incoming calls only. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17create a UDPTL structure in create_addr_from_peer() if it does not already ↵Dwayne M. Hubbard
exist for T38 This is required to create a UDPTL structure in create_addr_from_peer() to handle the scenario where 't38pt_udptl=yes' is not defined in the [general] section of sip.conf but is defined the peer's context. I tested this patch by enabling t38pt_udptl in the [general] section on one system and only enabling t38pt_udptl in a peer's context on the system sending a fax. Without the patch, the sending system will fail to initiate T38 negotiation with the warning message, "No way to add SDP without an UDPTL structure". When this patch is applied the sending side will successfully initiate T38 negotiation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17Prior to masquerade, move the group definitions to the channel performing theTilghman Lesher
masq, so that the group count lingers past the bridge. (closes issue #14275) Reported by: kowalma Patches: 20090216__bug14275.diff.txt uploaded by Corydon76 (license 14) Tested by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17Merge a large set of updates to the Asterisk indications API.Russell Bryant
This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17In this version, we can combine the queries, because we support droppingTilghman Lesher
nonexistent columns. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17Merged revisions 176426 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176426 | tilghman | 2009-02-16 18:49:22 -0600 (Mon, 16 Feb 2009) | 10 lines After a 'sip reload', qualifies for realtime peers weren't immediately restarted, instead waiting until the next registration. We're now caching the qualify across a reload/restart and starting the qualify immediately upon loading the peer. (closes issue #14196) Reported by: pdf Patches: 20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663) Tested by: pdf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176459 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Merged revisions 176354 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176354 | dvossel | 2009-02-16 17:30:52 -0600 (Mon, 16 Feb 2009) | 8 lines Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that. issue #13749 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Use the correct list macros for deleting an item from the middle of a list.Tilghman Lesher
(issue #13777) Reported by: pj Patches: 20090203__bug13777.diff.txt uploaded by Corydon76 (license 14) Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Merged revisions 175597 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/trunk ........ r175597 | dvossel | 2009-02-13 14:11:55 -0600 (Fri, 13 Feb 2009) | 4 lines Fixed iax2 key rotation backwards compatibility Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Can't set debug level 2 (intense debugging) unless the syntax matchesTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Remove chan_features.Russell Bryant
Review: http://reviewboard.digium.com/r/161/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Merged revisions 176029 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176029 | file | 2009-02-16 11:33:53 -0400 (Mon, 16 Feb 2009) | 9 lines Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog. This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the pool was used for the value while the old was left untouched/unused. If the current pool was full a new pool was created. This would cause memory usage to increase steadily. (issue #AA50-2332) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-16Merged revisions 175921 via svnmerge from Michiel van Baak
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175921 | mvanbaak | 2009-02-16 00:37:03 +0100 (Mon, 16 Feb 2009) | 3 lines fix mis-spelling of the word registered. Reported by De_Mon on #asterisk-dev. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15Fix a number of problems with ast_sched_report().Russell Bryant
1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-15Merged revisions 175777 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175777 | oej | 2009-02-15 20:48:38 +0100 (Sön, 15 Feb 2009) | 2 lines Make sure that the debug line is not printed on debug level 0 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13Fixed iax2 key rotation backwards compatibilityDavid Vossel
Turns key rotation back on by default. Added bit into encryption IE to indicate whether or not key rotation is supported or not. If it is not supported then it is not enabled, which insures backwards compatibility. This eliminates the need for the keyrotate option in iax.conf, so it has been removed. Review: http://reviewboard.digium.com/r/159/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13Add basic (passthrough, playback, record) support for ITU G.722.1 and ↵Kevin P. Fleming
G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13Add dynamic fax buffer configuration option to chan_dahdi.confDwayne M. Hubbard
When the 'faxdetect' configuration option is used, one may also want to use the 'faxbuffers' configuration option in chan_dahdi.conf. This option will dynamically use the configured 'faxbuffers' buffer policy on a channel for the life of the call following the detection of fax tones. The faxbuffers buffer policy will be reverted during call teardown. An example use of 'faxbuffers' is below. This example would switch to using 6 buffers with a full buffer policy. faxbuffers=>6,full git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Remove useless string copy, and make sscanf safe againRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Adds force encryption option to iax.confDavid Vossel
This patch adds forceencryption=yes as an iax.conf option. When force encryption is enabled, no unencrypted connections are allowed. This insures all connections are encrypted. This is a new feature, so CHANGES and iax.conf.sample are updated as well. (closes issue #13285) Reported by: sgofferj Tested by: russell Review: http://reviewboard.digium.com/r/150/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Avoid using ast_strdupa() in a loop.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12correct warning message to not refer specifically to DAHDIKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Setting key rotation to be off by defaultDavid Vossel
Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175127 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Issue a warning message if our candidate's IP is the loopback address.Philippe Sultan
(closes issue #13985) Reported by: jcovert Tested by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-12Merged revisions 175029 via svnmerge from Philippe Sultan
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175058 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10Only decrease inringing count if above zero.Joshua Colp
(issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10Set the type for the peer structure to be a peer as the default.Joshua Colp
(closes issue #14447) Reported by: triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-10Make the logic for inuse and inringing manipluation match that of 1.4. The ↵Joshua Colp
old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported. (closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09Fix something I messed up in the merge I just didMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-09Merged revisions 174282 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-06Merged revisions 174082 via svnmerge from Dwayne M. Hubbard
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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