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Author	SHA1	Message	Date
Matthew Jordan	81598fa082	doc: Add a manpage for the aelparse utility ... This patch adds a manpage for the aelparse utility. Note that this is one of the patches the Debian distro applies for the Asterisk project, as per ASTERISK-24191. Review: https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy Laine (License 6561) ........ Merged revisions 422371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422372 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422373 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422374 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	2df2d785b7	The assertion that peer was not found on final event ... message was being triggered on configuration reload. This patch changes that case to just return instead. Review: https://reviewboard.asterisk.org/r/3953/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3194892ea2	LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP ... The UniMRCP project distributes Asterisk modules that integrate Asterisk with UniMRCP, and other Asterisk users use the UniMRCP library as well. Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software Foundation, is not a compatible license with the GPLv2. "Please note that this license is not compatible with GPL version 2, because it has some requirements that are not in that GPL version. These include certain patent termination and indemnification provisions. The patent termination provision is a good thing, which is why we recommend the Apache 2.0 license for substantial programs over other lax permissive licenses." On the other hand, UniMRCP is a great project and we'd like to let people use it with Asterisk. This patch updates the LICENSE text to allow users to link Asterisk with UniMRCP and distribute the resulting binaries. ........ Merged revisions 422293 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422294 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422295 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422296 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	c5916fb39f	chan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure ... The reporter on the issue found some issues when upgrading from version 10 to 11 on 55 hosts. Two situations that can occur with dynamic registrations. 1. With dnsmgr disabled, if the host is not resolvable we are not trying to resolve the host again when it is time to attempt to register again. This results in never registering to the host. 2. With dnsmgr enabled, when the host is temporarily not resolvable the address is set to 0.0.0.0:0 and then when the host is resolvable the port is not being restored and stays set to 0. This patch resolves these two issues by: * Storing the hostname so that it can be used for resolving with DNS. * Resolve the hostname on the next scheduled attempt to register. * Storing the port used to reach the host so that when the hostname is resolvable again, we can set the port again if the port is still unset after looking up the host. ASTERISK-23767 #close Reported by: David Herselman Tested by: David Herselman, Michael L. Young Patches: asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3856/ ........ Merged revisions 422274 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422275 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422276 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4e750a26fd	Added ConfBridge AMI event note to UPGRADE.txt. ... ........ Merged revisions 422255 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422256 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Paul Belanger	ef28cc0d43	chan_sip.c: Add 'rtpbindaddr' setting ... Users now have the ability to bind the rtpengine instance to a specific IP address. For example, you want chan_sip (call control) on eth0 but rtp (media) on eth1. ASTERISK-24280 #close Reported by: Paul Belanger Tested by: Paul Belanger Review: https://reviewboard.asterisk.org/r/3952/ Patches: rtpengine.diff uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	327d67270f	Fix bug that did not allow for multiple batched RLS notifications to be sent. ... A misunderstanding of how the scheduler worked caused further batched notifications beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after the batched notification is sent. This way, further notifications can be scheduled when they arise. ........ Merged revisions 422239 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	94e1b4a8a4	res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage. ... * Fix off nominal ref leak in find_or_create_contact_status(). * Add missing NULL check of status in update_contact_status() and init_start_time(). ........ Merged revisions 422214 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422215 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4728c05957	sched: Fix typo and whitespace change. ... ........ Merged revisions 422200 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	7c1a22fba7	confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events ... Currently there's no way to tell if a user is an admin or not when receiving the join, leave, mute, unmute and talking events. This patch adds that capability. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3950/ ........ Merged revisions 422176 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422177 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	bf85018107	CallerID: Fix parsing of malformed callerid ... This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ Review: https://reviewboard.asterisk.org/r/3933/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422114 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422154 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d199536a04	confbridge: Make kick, mute and unmute handle channel targets consistently. ... Kick, mute and unmute were a little inconsistent in their handling of channel targets. This patch cleans that up by insuring they all handle the 'all' target consistently and adds the 'participants' target which acts on non-admins. Documentation for kick was also cleaned up as it never supported partial channel names. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3944/ ........ Merged revisions 422090 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422091 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	c5ab4adf17	Fix race condition in the scheduler when deleting a running entry. ... When scheduled tasks run, they are removed from the heap (or hashtab). When a scheduled task is deleted, if the task can't be found in the heap (or hashtab), an assertion is triggered. If DO_CRASH is enabled, this assertion causes a crash. The problem is, sometimes it just so happens that someone attempts to delete a scheduled task at the time that it is running, leading to a crash. This change corrects the issue by tracking which task is currently running. If that task is attempted to be deleted, then we mark the task, and then wait for the task to complete. This way, we can be sure to coordinate task deletion and memory freeing. ASTERISK-24212 Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3927 ........ Merged revisions 422070 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422071 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	fefa6fba82	res_musiconhold.c: Release any format refs before memset(). ... * Clear the channel music_state pointer before destroying the music_state object for safety. ........ Merged revisions 422037 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2b19d94a71	res_musiconhold: Fix MOH restarting where it left off from the last hold. ... Restore code removed by https://reviewboard.asterisk.org/r/3536/ that introduced a regression that prevents MOH from restarting were it left off the last time. ASTERISK-24019 #close Reported by: Jason Richards Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3928/ ........ Merged revisions 421976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421977 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421978 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421979 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	497a92d079	res_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs. ... In order to alter the Contact header on in-dialog requests and responses the Websocket module must be attached on outgoing INVITEs. The Contact header is modified so that the PJSIP transport layer can find and use the existing Websocket connection based on the source IP address, port, and transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov ........ Merged revisions 421955 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421956 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	477e2e6edb	res_pjsip_transport_websocket: Fix a progressive memory growth. ... The packet structure used to receive messages was using the transport pool. This meant that for each parsing the pool would grow accordingly. Since memory can not be reclaimed without resetting it this would cause the memory pool to grow and grow. This change uses a specific memory pool for the packet structure and resets it to a fresh state after the message has been received and handled. ........ Merged revisions 421939 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421945 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	2c0cbf8e64	res_pjsip_transport_websocket: Ensure secure Websocket clients can be called. ... This change enforces the transport in the Contact header for Websocket clients. Previously a client may provide a transport of 'ws' when it is actually using a transport of 'wss'. This would cause outgoing calls to fail as the existing connection could not be found. ........ Merged revisions 421931 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421932 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	cee660dadf	chan_sip: Use the server reflexive ICE candidate RTCP port as provided. ... This code originally worked around an issue within res_rtp_asterisk itself. The wrong socket was being used for the STUN check for RTCP, causing the port to be the same as RTP. This was subsequently fixed and the RTCP port provided for the ICE candidate is correct and does not need to be incremented. ASTERISK-23997 #close Reported by: Badalian Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav (license 5249) ........ Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421911 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcfffce66d	Fix a locking inversion in MixMonitor. ... We need to unlock the audiohook before trying to lock the channel, since the correct locking order is channel then audiohook. ........ Merged revisions 421882 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	33835e17a0	ARI: Fix a crash caused by hanging during playback to a channel in a bridge ... ASTERISK-24147 #close Reported by: Edvin Vidmar Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged revisions 421879 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421880 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1498ae0830	main/message: Add a new-line to a DEBUG message ... ........ Merged revisions 421859 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421860 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f8c4fc1121	res_musiconhold.c: Remove obsolete REF_DEBUG code. ... Remove unneeded code that writes to the wrong file location in an obsolete format. ........ Merged revisions 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421800 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421801 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421802 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	644e693645	Switch from hostname to an IP address in the SDP origin line. ... Using the hostname in the SDP origin line may not satisfy the requirement of RFC 4566 that we use a FQDN or IP address. This change has us use the same information from the SDP connection line if possible. If not possible, we'll use the configured media address. And if that's not possible, we use the result of a PJLIB call to get the IP address of ourself. ASTERISK-23994 #close Reported by Private Name Review: https://reviewboard.asterisk.org/r/3925 ........ Merged revisions 421796 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421797 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	56a1d4930a	Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge. ... Because of the departable state of channels that enter Stasis bridges, Stasis has to take responsibility for directing the channel to its intended after-bridge destination if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures that when such a move occurs, when the channel leaves the bridging system, any after bridge gotos are honored. Review: https://reviewboard.asterisk.org/r/3920 ........ Merged revisions 421792 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421794 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	4946981646	res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set ... Due to a faulty function for debugging reference decrementing, it was possible to reduce the refcount on the wrong object if two moh classes of the same name were in the moh class container. (closes issue ASTERISK-22252) Reported by: Walter Doekes Patches: 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182) ........ Merged revisions 398937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421777 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421779 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421788 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	12d34bb12f	Let's try checking the name and number, instead of the name twice. ... ........ Merged revisions 421789 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421790 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2150daf748	Improve consistency of party ID privacy usage. ... Prior to this change, the Remote-Party-ID header took the position of "If caller name and number are not explicitly allowed, then they are private" and P-Asserted-Identity took the position of "Caller name and number are only private if marked explicitly so" Now both mechanisms of conveying party identification use the former approach. ........ Merged revisions 421778 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421783 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	77ddc5b713	chan_sip: Don't use port derived from fromdomain if it isn't set ... If a user does not provide a port in the fromdomain setting, chan_sip will set the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will then get used unilaterally in certain places. This causes issues with TLS, where the default port is expected to be 5061. This patch modifies chan_sip such that fromdomainport is only used if it is not the standard SIP port; otherwise, the port from the SIP pvt's recorded self IP address is used. Review: https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close Reported by: Elazar Broad patches: fromdomainport_fix.diff uploaded by Elazar Broad (License 5835) ........ Merged revisions 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421718 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421719 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421720 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f3a525e9a6	ARI: Fix implicit answer when playback is initiated on unanswered channel ... When issuing a POST /channels/{channel_id}/play on a channel that is not yet answered, ARI is supposed to: * Queue up an AST_CONTROL_PROGRESS on the channel * Start up the playback of the media Instead, we sneak an answer on the channel right before starting playing media. This is due to ARI's usage of control_streamfile. This function implicitly answers the channel (and doesn't give ARI the option to stop it). The answering of the channel here is probably unnecessary: * app_voicemail, by far the biggest consumer of this function, always answers the channels anyway * control stream file (in res_agi) and ControlPlayback probably shouldn't be implicitly answering the channel. Answering should not be tied directly to playing back media. As it turns out, the answering of the channel here is pretty old: 356042 twilson if (ast_channel_state(chan) != AST_STATE_UP) { 3087 anthm res = ast_answer(chan); 180259 tilghman } (As in, ancient?) Note that others ran into this problem and commented about it on various mailing lists. Review: https://reviewboard.asterisk.org/r/3907/ ASTERISK-24229 #close Reported by: Matt Jordan ........ Merged revisions 421695 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421696 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	085d5a2629	Clean up files that do not end with newlines ... Trivial patch to add new lines to several files missing them. This fixes warnings when compiling with gcc 4.1.2 on CentOS 5. ASTERISK-24245 #close Reported by: Shaun Ruffell patches: 0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417) ........ Merged revisions 421677 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421678 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	da91946df7	uri: Quiet warning about type qualifiers ignored on function return type ... This patch fixes gcc warnings that occur due to the type qualifier 'const' being ignored on a return type of int. ASTERISK-24246 #close Reported by: Shaun Ruffell patches: 0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417) ........ Merged revisions 421675 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421676 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b7f98c3da4	chan_pjsip: Update media translation paths when new SDP negotiated. ... On a SIP reinvite that changes media strams, the PJSIP channel driver was flooding the log with "Asked to transmit frame type %s, while native formats is %s" warnings. * Fixes PJSIP not setting up translation paths when the formats change on a reinvite. AFS-63 was effectively reintroduced because of the media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the unexpected frame format WARNING message to include more information. * Added protective locking while altering formats on a channel. Reworked set_format() to simplify and protect the formats under manipulation. * Restored some code that got lost in the media_formats work. (channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3906/ ........ Merged revisions 421645 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4672c139dd	cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled. ... filename_completion_function() returns memory that was not allocated by the MALLOC_DEBUG allocation tracker so the memory must be freed by ast_std_free(). ........ Merged revisions 421600 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421602 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421608 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421616 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	49f8bd4ad4	Set the role for inbound subscriptions correctly. ... This was causing the AMI show_subscriptions test in the testsuite to fail since all subscriptions were being seen as subscribers instead of notifiers. ........ Merged revisions 421585 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	d0640ad7df	Move evaluation of set_var options in pjsip to the end of channel initialization. ... This allows for set_var to override certain defaults such as caller ID and codec values. This also fixes a test suite regression. The "set_var" test suite test attempted to use set_var to override caller ID, but a recent change caused that to no longer work. ........ Merged revisions 421565 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421566 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	36f4bff943	Stasis: Add information to blind transfer event ... When a blind transfer occurs that is forced to create a local channel pair to satisfy the transfer request, information about the local channel pair is not published. This adds a field to describe that channel to the blind transfer message struct so that this information is conveyed properly to consumers of the blind transfer message. This also fixes a bug in which Stasis() was unable to properly identify the channel that was replacing an existing Stasis-controlled channel due to a blind transfer. Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3921/ ........ Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421538 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	01f1ff1f77	AMI: Add AllVariables parameter to Status ... This adds the AllVariables parameter to the Status AMI action such that if defined and set to "true", all channel variables will be reported in the subsequent Status event(s). This parameter does not negate the functionality of the "Variables" parameter so that global variables and dialplan functions can be requested. Review: https://reviewboard.asterisk.org/r/3915/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	76290adf50	Alter documentation for callerid_privacy to use correct values. ... ........ Merged revisions 421485 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421488 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	28a89e7685	Fix compilation error on certain versions of GCC. ... ........ Merged revisions 421447 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421448 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	a85a483fcd	AMI Docs: Fix Status channel parameter optionality ... ........ Merged revisions 421442 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421443 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421444 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421445 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	222b5cd036	ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX ... If /channels/{channelID}/continue is called on a channel that was originated without a PBX (such as the ARI command POST channel with a stasis application argument), the channel will not start dialplan execution. This patch will now run the PBX out of the stasis execution if the channel doesn't currently have an active PBX upon continuing. ASTERISK-24043 #close Reported by: Krandon Bruse Review: https://reviewboard.asterisk.org/r/3917/ Patches: stasis-continue.diff submitted by Krandon Bruse (license 6631) ........ Merged revisions 421416 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421423 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	83a9b91da9	chan_pjsip: Fix attended transfer connected line name update. ... A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Damien Wedhorn	c4c9d4ad6c	Skinny: Fixup compile warning for non dev-mode. ... ........ Merged revisions 421376 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	1de8b8035e	func_config: Change 'Not Found' message from ERROR to DEBUG ... When you call the CONFIG dialplan function with the name of a variable that doesn't exist in the target context you get an ERROR. This does nothing but clutter up the logs with messages that may be perfectly acceptable. Just because a variable wasn't in the context doesn't mean it's an error. Maybei t's optional or just needs to be defaulted or ignored. This patch changes the log level from ERROR to DEBUG. If a dialplan developer wants to debug their dialplan they still canby setting the console debug level as needed. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........ Merged revisions 421327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421328 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421329 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421337 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bb494067a5	Multiple revisions 421311-421312 ... ........ r421311 | mjordan | 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) | 9 lines res/ari/resource_channels: Don't return allocation failure on failed function If a function fails to execute, it is most likely due to one of two reasons: (1) The function doesn't exist or can't be read from (2) The function is dangerous and is restricted based on the user's permissions Currently we return allocation failure, which is incorrect. This updates the reason code to more accurately reflect why the request failed. ASTERISK-24215 ........ r421312 | mjordan | 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) | 4 lines res/ari/resource_channels: Fix compilation issue Forgot a parameter. Whoops. ........ Merged revisions 421311-421312 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ba5d5da60b	Improve call forwarding reporting, especially with regards to ARI. ... This patch addresses a few issues: 1) The order of Dial events have been changed when performing a call forward. The order has now been altered to 1) Dial begins dialing channel A. 2) When A forwards the call to B, we issue the dial end event to channel A, indicating the dial is being canceled due to a forward to B. 3) When the call to channel B occurs, we then issue a new dial begin to channel B. 2) Call forwards are now reported on the calling channel, not the peer channel. 3) AMI DialEnd events have been altered to display the extension the call is being forwarded to when relevant. 4) You can now get the values of channel variables for channels that are not currently in the Stasis application. This brings the retrieval of channel variables more in line with the rest of channel read operations since they may be performed on channels not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan ASTERISK-24138 #close Reported by Matt Jordan Patches: forward-shenanigans.diff uploaded by Matt Jordan (License #6283) Review: https://reviewboard.asterisk.org/r/3899 ........ Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6525f374db	apps/app_meetme: Fix crash when publishing MeetMe messages with no channel ... The same function, meetme_stasis_generate_msg, handles creating and publishing Stasis message both when there are channels in the MeetMe conference and when there are no channels in the conference. When the performance improvement was made to use cached snapshots, this created a situation where Asterisk would crash: obtaining a cached snapshot is not NULL tolerant. This patch restores the previous implementation, which used a NULL safe set of routines to produce a blob containing the channel snapshot (if available) and information about the MeetMe conference. ASTERISK-24234 #close Reported by: Shaun Ruffell Tested by: Shaun Ruffell ........ Merged revisions 421270 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421273 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	44fc6ea6ff	apps/app_dial: Fix Dial 'z' option ... The 'z' option is supposed to disable the dial timeout in the case of a call forward. Unfortunately, the wrong timeout timer was passed to the do_forward function, resulting in the option not working. ASTERISK-24225 #close Reported by: dimitripietro Tested by: dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621) ........ Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421233 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421235 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421236 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	98ca5c0b5f	configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc ... Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is executed with optimization. This "help" unfortunately results in re-definition warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning. Review: https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close Reported by: Kilburn Tested by: Kilburn, wdoekes patches: 1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by cloos (License 5956) 11.diff uploaded by cloos (License 5956) 12.diff uploaded by cloos (License 5956) 13.diff uploaded by cloos (License 5956) ........ Merged revisions 421227 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421228 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421229 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421230 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago