asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	cacc356bbe	When a session timer expires during a T.38 call, re-invite with correct SDP When a session timer expires during a dialog that has re-negotiated to T.38 and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP containing audio media only. This causes some hilarity with the poor fax session under weigh. This patch corrects that by sending T.38 parameters if we are in the middle of a T.38 session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches: dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418) ........ Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383125 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d1d66c3878	Fix processing of call files when using KQueue on OS X In certain situations, call files are not processed when using KQueue with pbx_spool. Asterisk was sending an invalid timeout value when the spool directory is empty, causing the call to kevent to error immediately. This can create a tight loop, increasing the CPU load on the system. (closes issue ASTERISK-21176) Reported by: Carlton O'Riley patches: kqueue_osx.patch uploaded by coriley (License 6473) ........ Merged revisions 383120 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383121 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	7a952f1841	Fix whitespace in AST_EXT_LIB_CHECK macro. ........ Merged revisions 383061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 383062 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	95849b1a83	Always set the RTP instance data in the RTP engine Not informing the RTP engine of the instance data creates shrapnel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Andrew Latham	e29737179a	Update Doxygen Push some cleanups upstream before testing another ticket. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	ba7ee9bfc9	Fix Sorting Order For Parking Lots Stored In Static Realtime When retrieving the parking lots from a MySQL database table, the current order is "filename, cat_metric desc, var_metric asc, category". If there are multiple parking lots with the same cat_metric but different categories, everything is being sorted on cat_metric first resulting in errors when loading the parking lots. This patch fixes the problem by sorting on the category field first, then the cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young (license 5026) ........ Merged revisions 382942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382943 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	6a57a36d28	Update Contributed Realtime Schema Files - IPv6 Addresses This commit updates some fields in the contributed realtime schema files to handle IPv6 addresses. (closes issue ASTERISK-21173) Reported by: Torrey Searle Patches: realtime_sql.patch Torrey Searle (license 5334) asterisk-21173-update-ip-fields.diff Michael L. Young (license 5026) ........ Merged revisions 382939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382940 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	9a992c6cba	Fix a crash when res_xmpp is configured using a username without a domain. (closes issue ASTERISK-21156) Reported by: amsoft2001 ........ Merged revisions 382923 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	1cb917096b	Switch to using external pjproject libraries. ICE/STUN/TURN support in res_rtp_asterisk is also now optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	00e9ffb907	Include the Username field in SIP Registry events when Status is registered In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed to include the Username field. Somehow, one of the events was missed. This patch corrects that - the Username field should be included in all AMI Registry events involving SIP registrations. (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by: Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) ........ Merged revisions 382847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382848 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	ef64b29f8b	Fix core dump on CLI usage Fix issue with 'unistim show info' CLI command when device connected not configured ........ Merged revisions 382827 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	09ecb25e08	Added an option to disallow music on hold Added an option "discard_remote_hold_retrieval" (default "no") that if set does not trigger the music on hold event. This essentially stops telling the peer to start music on hold. (issue ABE-2899) Reported by: Denis Alberto Martinez Review: https://reviewboard.asterisk.org/r/2336/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	761465d642	confbridge: Rename items for clarity and consistency. struct conference_bridge_user -> struct confbridge_user struct conference_bridge -> struct confbridge_conference struct conference_state -> struct confbridge_state struct conference_bridge_user conference_bridge_user -> struct confbridge_user user struct conference_bridge_user cbu -> struct confbridge_user user struct conference_bridge conference_bridge -> struct confbridge_conference conference The names are now generally shorter, consistently used, and don't conflict with the struct names. This patch handles the renaming part of the issue. (issue ASTERISK-20776) Reported by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b4a010e958	chan_sip: Update the via header when relaying SMS MESSAGE Prior to this change, certain conditions for sending the message would result in an address of '(null)' being used in the via header of the SIP message because a NULl value of pvt->ourip was used when initially generating the via header. This is fixed by adding a call to build_via when the address is set before sending the message. (closes issue ASTERISK-21148) Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475) ........ Merged revisions 382739 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	91eba7dc13	Stasis documentation updates. (issue ASTERISK-20887) (issue ASTERISK-20959) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c0e2ed1fe9	Ensure dummy channels get a stasis topic. Fixes test failure introduced in r382685. (issue ASTERISK-20887) (issue ASTERISK-20959) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	c6b06e40dc	Add message dump capability to stasis cache layer The cache dump mechanism allows the developer to retreive multiple items of a given type (or of all types) from the cache residing in a stasis caching topic in addition to the existing single-item cache retreival mechanism. This also adds to the caching unit tests to ensure that the new cache dump mechanism is functioning properly. Review: https://reviewboard.asterisk.org/r/2367/ (issue ASTERISK-21097) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382705 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4edd8be35c	This patch adds a new message bus API to Asterisk. For the initial use of this bus, I took some work kmoore did creating channel snapshots. So rather than create AMI events directly in the channel code, this patch generates Stasis events, which manager.c uses to then publish the AMI event. This message bus provides a generic publish/subscribe mechanism within Asterisk. This message bus is: - Loosely coupled; new message types can be added in seperate modules. - Easy to use; publishing and subscribing are straightforward operations. In addition to basic publish/subscribe, the patch also provides mechanisms for message forwarding, and for message caching. (issue ASTERISK-20887) (closes issue ASTERISK-20959) Review: https://reviewboard.asterisk.org/r/2339/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f6f6bc7b59	Remove unused function After r382670, get_ip_and_port_from_sdp was no longer used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	12748bc735	Don't reset the RTP address on a glare re-INVITE Originally, way back in r201583, we added the alternate RTP address so that the RTP engine would expect to receive audio from a new source when a glare re-INVITE occurred. In r382589, we remove the alternate RTP source, as the 'secret' probation mode allows for switching to a new RTP source when a previous source stops sending RTP. At the time, it seemed appropriate to set the RTP source based on the information in the glared re-INVITE. Unfortunately, that doesn't work so well - in a glared re-INVITE that occurs with no SDP - such as in a connected line update that glances - we'll set the RTP source to an invalid address. In subsequent re-INVITE requests from this Asterisk instance, we'll then send an invalid media address, which will result in the remote side sending a 488. Whoops. There isn't any need to reset the RTP source - if we're using strictrtp, we'll simply synchronize to a new source when we stop getting packets from the old one. If we aren't using strictrtp, then again there shouldn't be a problem. Note that the Asterisk Test Suite's connectedline test caught this error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	3f0ea90ce6	Changing log level of "Not changing threadpool size" from notice to debug. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	c55592a324	Load sorcery modules earlier, so they can actually be used. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b0fc2032ff	Let vm_mailbox_snapshot combine "Urgent" when no folder is specified r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot that Urgent also "counts" as new messages. This fixed the problem when any of the three folders was specified and the combine option was used. It missed the case where the folder isn't specified and we build a snapshot of all folders. This patch corrects that. ........ Merged revisions 382617 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	dd867daac9	Fix a memory leak in xmldoc Another instance of attribute retrieval not being freed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	675f43f24f	Resolve more memory leaks in xmldoc Many places that allocated to pull out an attribute are now freed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	80b8c2349c	Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source Often, Asterisk may realize that a change in the source of an RTP stream is about to occur and ask that the RTP engine reset it's lock on the current RTP source. In certain scenarios, it may take awhile for the new remote system to send RTP packets, while the old remote system may continue providing RTP during that time period. This causes Asterisk to re-lock onto the old source, thereby rejecting the new source when the old source stops sending RTP and the new source begins. This patch prevents that by having a constant secondary, 'secret' probation mode enabled when an RTP source has been chosen. RTP packets from other sources are always considered, but never chosen unless the current RTP source stops sending RTP. Review: https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124) Reported by: John Bigelow Tested by: John Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested by: John Bigelow ........ Merged revisions 382573 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a3a2b99519	Fix minor memory leak in xmldoc Strings retrieved via ast_xml_get_text() must be freed with ast_xml_free_text(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	e6b5e3a62a	Ensure that logmsgs are freed properly Messages sent while the logger thread is shutting down will now have their associated callid freed properly. ........ Merged revisions 382574 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0366f7ca77	Fix ref leak in threadpool.c If ast_threadpool_set_size with a size equal to the current size, a reference to a set_size_data structure would be leaked. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d182115fcb	Resolve a ref leak in threadpool.c Ownership of the listener reference is not transferred because the listener is reffed when placed into the taskprocessor. Ensure that the listener is dereffed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8d5c36c9bb	Add RFC 3327 Path header support to chan_sip This patch adds support for RFC 3327 "Path" headers. This can be enabled in sip.conf using the 'supportpath' setting, either on a global basis or on a peer basis. This setting enables Asterisk to route outgoing out-of-dialog requests via a set of proxies by using a pre-loaded route-set defined by the Path headers in the REGISTER request. This patch also adds Realtime support for dynamically updating the Path information for a peer. A huge thank-you to Klaus Darillion and Olle E Johansson for their efforts in writing this patch. Review: https://reviewboard.asterisk.org/r/2235/ Review: https://reviewboard.asterisk.org/r/991/ (closes issue ASTERISK-16884) Reported by: klaus3000 Tested by: klaus3000, oej, mjordan patches: path-1.8.0-patch.txt uploaded by klaus3000 (License 5054) oolong-path-support-trunk in team branch by oej (License 5267) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	469ca1c71d	Fix several unreleased mutex locks that cause problem with processing calls	12 years ago
Richard Mudgett	736f4e9420	Fixup some bridge and format capabilities comments and whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	22d58fbea8	Fix comparison of presence state in event subsystem. Several new IEs were not given types (or names), causing the comparison function to improperly succeed. This adds those. (closes issue AST-1128) ........ Merged revisions 382390 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	933800754f	Confbridge CLI new record file name check. This fix checks to make sure that if a confbridge record start command is issued from the CLI it will always use the file name given on the CLI even if it changes between start/stop records for a conference. Previously it had been reusing the same file between start/stops even if a new filename was given. (issue AST-1088) Reported by: John Bigelow ........ Merged revisions 382385 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	a4f45a2c95	Add support for registering a sorcery handler which supports multiple fields using a regex. Review: https://reviewboard.asterisk.org/r/2332/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382340 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	a3ad8b28e6	Fix / Clean Up Some Items To Handle The New auto_* NAT Options The original report had to do with a realtime peer behind NAT being pruned and the peer's private address being used instead of its external address. Upon debugging, it was discovered that this was being caused by the addition of the auto_force_rport and auto_comedia settings. This patch does the following: * Adds a missing note to the CHANGES file indicating that the default global nat setting is auto_force_rport * Constify the 'req' parameter for check_via() * Add calls to check_via() in a couple of places in order for the auto_* settings to do their job in attempting to determine if NAT is involved * Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_* settings are in use where it was needed * Moves the copying of peer flags up in build_peer() to before they are used; this fixes the realtime prune issue * Update the contrib/realtime schemas to allow the nat column to handle the different nat setting combinations we have This patch received a review and "Ship It!" on the issue itself. (closes issue ASTERISK-20904) Reported by: JoshE Tested by: JoshE, Michael L. Young Patches: asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026) ........ Merged revisions 382322 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382323 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3a8caa351e	While the ICE negotiation is occurring leave strictrtp in an open state, media can and will come from different places. ........ Merged revisions 382298 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	50a74cbd2a	Fix a bug with ICE and strictrtp where media could get dropped. If the end result of the ICE negotiation resulted in the path for media changing it was possible for the strictrtp code to discard the RTP packets. This change causes strictrtp to enter learning mode once again when the ICE negotiation has completed successfully. ........ Merged revisions 382296 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	855bb334c8	threadpool: Make ast_threadpool_push() return -1 if shutting_down git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e2832f18bc	threadpool: Whitespace and comment corrections. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	6acc9ceb76	Don't undefine bzero()/bcopy(). This was causing build failures against external libraries that happened to use them, unless silly hacks were added to the modules that used those headers. Review: https://reviewboard.asterisk.org/r/2359/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382292 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1a34b465bc	Prevent deadlock in chan_iax2 when attempting to set caller ID A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it already holds the iax2 private lock and improperly fails to obtain the channel lock before calling ast_set_callerid. By not safely obtaining the channel lock, a locking inversion can take place, causing a deadlock. This patch solves this by calling the required deadlock avoidance functions that obtain the channel lock before setting the caller ID. Thanks to Pavel for fixing my syntax errors and testing this patch out. (closes issue ASTERISK-21128) Reported by: Pavel Troller Tested by: Pavel Troller patches: ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283) ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302) ........ Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382234 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	62f7acfac6	Let channels joining a MeetMe conference opt out of the denoiser For some channel drivers, specifically those that have a varying rate in the number of audio samples, the audio quality for a MeetMe conference can be exceedingly poor. This is due to a unilateral application of the DENOISE function in func_speex to channels joining the conference. The denoiser function in the speex library is initialized with the number of audio samples in each sample that will be provided to it. If the number of audio samples changes, the denoiser has to be thrown away and re-initialized. While this could be worked around by removing func_speex, that doesn't help if you actually use the denoiser with other channels on the system. This patches does the following: * Checks for the presence of func_speex as opposed to codec_speex when determining if the DENOISE function is present (which is where the function is actually implemented) * Adds an option to MeetMe 'n' that causes the denoiser to not be applied to a channel when it joins. This keeps the current behavior the default, but let's users disable the denoiser if it causes problems on their system. Review: https://reviewboard.asterisk.org/r/2358 (closes issue AST-1062) Reported by: Thomas Arimont ........ Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382230 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382232 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	de90681293	More places to eliminate the cast to argv but were not giving warnings. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382204 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	31f08344ee	Fix compiler warning by eliminating the need for a cast. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e0b49e7331	Relax dialog checking in get_sip_pvt_byid_locked so it works when the dialog is forked. (closes issue ASTERISK-20638) Reported by: eelcob Patches: pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442) ........ Merged revisions 382171 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382174 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	d6a6422880	Consider linux-gnuspe as linux-gnu * The powerpcspe Linux port uses linux-gnuspe as the OS string. * Our build system shouldn't really care for that, so just call it linux-gnu. * Original report: Roland Stigge , http://bugs.debian.org/701505 Review: https://reviewboard.asterisk.org/r/2357/ ........ Merged revisions 382110 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382111 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	d33d9c1781	Correct RPID parsing for unquoted display-name. Parsing Remote-Party-ID will now succeed if display-name is of the *(token LWS) kind and not just the quoted-string kind. Review: https://reviewboard.asterisk.org/r/2341/ ........ Merged revisions 382107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382108 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	4951ab9ed1	Remove unneeded linux-gnueabi* As of r380522 the configure scripts converts the value of linux-gnueabi* of OSARCH to "linux-gnu". So no point in testing for those values. ........ Merged revisions 382087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 382096 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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