181 Commits (22.5.0)

Author	SHA1	Message	Date
Mike Bradeen	2c168d5afe	chan_pjsip: Serialize INVITE creation on DTMF attended transfer ... When a call is transfered via DTMF feature code, the Transfer Target and Transferer are bridged immediately. This opens the possibilty of a race condition between the creation of an INVITE and the bridge induced colp update that can result in the set caller ID being over-written with the transferer's default info. Fixes: #1234	1 month ago
Alexei Gradinari	04ed031b62	chan_pjsip: set correct Endpoint Device State on multiple channels ... 1. When one channel is placed on hold, the device state is set to ONHOLD without checking other channels states. In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN to calculate aggregate device state of all active channels. 2. The current implementation incorrectly classifies channels in use. The only channels that has the states: UP, RING and BUSY are considered as "in use". A channel should be considered "in use" if its state is anything other than DOWN or RESERVED. 3. Currently, if the number of channels "in use" is greater than device_state_busy_at, the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate device state. The endpoint device state should be BUSY if the number of channels "in use" is greater than or equal to device_state_busy_at. Fixes: #1181 (cherry picked from commit b82d9b0d5b)	6 months ago
Holger Hans Peter Freyther	fe5ff3021e	ari/pjsip: Make it possible to control transfers through ARI ... Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function. (cherry picked from commit a0d0c47d06)	7 months ago
Maximilian Fridrich	0bd4d3320a	chan_pjsip: Send VIDUPDATE RTP frame for all H.264 streams ... Currently, when a chan_pjsip channel receives a VIDUPDATE indication, an RTP VIDUPDATE frame is only queued on a H.264 stream if WebRTC is enabled on that endpoint. This restriction does not really make sense. Now, a VIDUPDATE RTP frame is written even if WebRTC is not enabled (as is the case with VP8, VP9, and H.265 streams). Resolves: #1013 (cherry picked from commit 45c5d39c58)	9 months ago
Ben Ford	fb0e46e464	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	1 year ago
Sean Bright	25686a57da	pjsip: Add PJSIP_PARSE_URI_FROM dialplan function. ... Various SIP headers permit a URI to be prefaced with a `display-name` production that can include characters (like commas and parentheses) that are problematic for Asterisk's dialplan parser and, specifically in the case of this patch, the PJSIP_PARSE_URI function. This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that behaves identically to `PJSIP_PARSE_URI` except that the first argument is now a variable name and not a literal URI. Fixes #756	1 year ago
George Joseph	f309ffad3d	chan_pjsip: Add PJSIPHangup dialplan app and manager action ... See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range.	2 years ago
Maximilian Fridrich	18f0b6661a	chan_pjsip: Allow topology/session refreshes in early media state (#74) ... With this change, session modifications in the early media state are possible if the SDP was sent reliably and confirmed by a PRACK. For details, see RFC 6337, escpecially section 3.2. Resolves: #73	2 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. ... For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	3 years ago
Igor Goncharovsky	410150235a	res_pjsip_rfc3326: Add SIP causes support for RFC3326 ... Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	3 years ago
George Joseph	7dc8773178	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	3 years ago
George Joseph	3a3d6c7dcb	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" ... This reverts commit e66c5da145. Reason for revert: Issue when transcoding to/from g722 Change-Id: I12853c5b1d3a77f5b9200f41908fd238a17159dc	3 years ago
George Joseph	e66c5da145	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	3 years ago
Ben Ford	31b3addce7	res_pjsip: Add TEL URI support for basic calls. ... This change allows TEL URI requests to come through for basic calls. The allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To headers will now allow TEL URIs, as well as the request URI. Support is only for TEL URIs present in traffic from a remote party. Asterisk does not generate any TEL URIs on its own. ASTERISK-26894 Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a	3 years ago
Maximilian Fridrich	a03b53bb7b	chan_pjsip: Only set default audio stream on hold. ... When a PJSIP channel is set on hold or off hold, all streams were set on/off hold. This is not the desired behaviour and caused issues when there were multiple streams in the topology. Now, only the default audio stream is set on/off hold when a hold is indicated. ASTERISK-30051 Change-Id: I04f1110565fd05fea565f5539b534b54549d4f71	3 years ago
Moritz Fain	4bf2473ac4	ari: expose channel driver's unique id to ARI channel resource ... This change exposes the channel driver's unique id (i.e. the Call-ID for chan_sip/chan_pjsip based channels) to ARI channel resources as `protocol_id`. ASTERISK-30027 Reported by: Moritz Fain Tested by: Moritz Fain Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87	3 years ago
Naveen Albert	193b7a81fe	chan_pjsip: Add ability to send flash events. ... PJSIP currently is capable of receiving flash events and converting them to FLASH control frames, but it currently lacks support for doing the reverse: taking a FLASH control frame and converting it into a flash event in the SIP domain. This adds the ability for PJSIP to process flash control frames by converting them into the appropriate SIP INFO message, which can then be sent to the peer. This allows, for example, flash events to be sent between Asterisk systems using PJSIP. ASTERISK-29941 #close Change-Id: I1590221a4d238597f79672fa5825dd4a920c94dd	4 years ago
Mark Petersen	1cdaeb8161	chan_pjsip: add allow_sending_180_after_183 option ... added new global config option "allow_sending_180_after_183" that if enabled will preserve 180 after a 183 ASTERISK-29842 Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18	4 years ago
Boris P. Korzun	6624e34580	res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity ... Change RTP timer behavior for detecting RTP only after two-way SDP channel establishment. Ignore detecting after receiving 183 with SDP or while direct media is used. Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout and rtpholdtimeout options in chan_sip. ASTERISK-26689 #close ASTERISK-29929 #close Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb	4 years ago
Alexander Traud	726d6dd166	channels: Fix for Doxygen. ... ASTERISK-29762 Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7	4 years ago
Naveen Albert	5b5c358e4b	res_pjsip_caller_id: Add ANI2/OLI parsing ... Adds parsing of ANI II digits (Originating Line Information) to PJSIP, on par with what currently exists in chan_sip. ASTERISK-29472 Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847	4 years ago
Sean Bright	1b41629447	chan_pjsip: Correct misleading trace message ... ASTERISK-29358 #close Change-Id: I050daff67066873df4e8fc7f4bd977c1ca06e647	5 years ago
Holger Hans Peter Freyther	48ed4f670f	pjsip: Generate progress (once) when receiving a 180 with a SDP ... ASTERISK-29105 Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38	5 years ago
Dan Cropp	55891227e8	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable ... When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	5 years ago
Ben Ford	948ceb1228	chan_pjsip.c: Add parameters to frame in indicate. ... There are a couple of parameters (datalen and data) that do not get set in chan_pjsip_indicate which could cause an Invalid message to pop up for things such as fax. This patch adds them to the frame. Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8	5 years ago
Ivan Poddubnyi	f2aa6c7017	chan_pjsip: Assign SIPDOMAIN after creating a channel ... session->channel doesn't exist until chan_pjsip creates it, so intead of setting a channel variable every new incoming call sets one and the same global variable. This patch moves the code to chan_pjsip so that SIPDOMAIN is set on a newly created channel, it also removes a misleading reference to channel->session used to fetch call pickup configuraion. ASTERISK-29240 Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755	5 years ago
Ivan Poddubnyi	134d2e729d	chan_pjsip: Stop queueing control frames twice on outgoing channels ... The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice. This resulted in extra noise in logs (for example, "is making progress" and "is ringing" get logged twice by app_dial), as well as in noise in signalling: one incoming 183 Session Progress results in 2 outgoing 183-s. This change splits the response handler into 2 functions: - one for updating HANGUPCAUSE, which is still called twice, - another that does the rest, which is called only once as before. ASTERISK-28016 Reported-by: Alex Hermann ASTERISK-28549 Reported-by: Gant Liu ASTERISK-28185 Reported-by: Julien Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940	5 years ago
Dan Cropp	ffa87ecade	chan_pjsip: Incorporate channel reference count into transfer_refer(). ... Add channel reference count for PJSIP REFER. The call could be terminated prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY occurred several minutes later, it would attempt to access a session which was no longer valid. Terminate event subscription if pjsip_xfer_initiate() or pjsip_xfer_send_request() fails in transfer_refer(). ASTERISK-29201 #close Reported-by: Dan Cropp Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435	5 years ago
Joshua C. Colp	6475fe3dd7	pjsip: Match lifetime of INVITE session to our session. ... In some circumstances it was possible for an INVITE session to be destroyed while we were still using it. This occurred due to the reference on the INVITE session being released internally as a result of its state changing to DISCONNECTED. This change adds a reference to the INVITE session which is released when our own session is destroyed, ensuring that the INVITE session remains valid for the lifetime of our session. ASTERISK-29022 Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9	5 years ago
George Joseph	44bb0858cb	debugging: Add enough to choke a mule ... Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	5 years ago
George Joseph	9bd1d686a1	ACN: Add tracing to existing code ... Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	5 years ago
George Joseph	8d1064eaaf	Streams: Add features for Advanced Codec Negotiation ... The Streams API becomes the home for the core ACN capabilities. These include... * Parsing and formatting of codec negotation preferences. * Resolving pending streams and topologies with those configured using configured preferences. * Utility functions for creating string representations of streams, topologies, and negotiation preferences. For codec negotiation preferences: * Added ast_stream_codec_prefs_parse() which takes a string representation of codec negotiation preferences, which may come from a pjsip endpoint for example, and populates a ast_stream_codec_negotiation_prefs structure. * Added ast_stream_codec_prefs_to_str() which does the reverse. * Added many functions to parse individual parameter name and value strings to their respectrive enum values, and the reverse. For streams: * Added ast_stream_create_resolved() which takes a "live" stream and resolves it with a configured stream and the negotiation preferences to create a new stream. * Added ast_stream_to_str() which create a string representation of a stream suitable for debug or display purposes. For topology: * Added ast_stream_topology_create_resolved() which takes a "live" topology and resolves it, stream by stream, with a configured topology stream and the negotiation preferences to create a new topology. * Added ast_stream_topology_to_str() which create a string representation of a topology suitable for debug or display purposes. * Renamed ast_format_caps_from_topology() to ast_stream_topology_get_formats() to be more consistent with the existing ast_stream_get_formats(). Additional changes: * A new function ast_format_cap_append_names() appends the results to the ast_str buffer instead of replacing buffer contents. Change-Id: I2df77dedd0c72c52deb6e329effe057a8e06cd56	5 years ago
Kevin Harwell	8b925fbda3	chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+ ... A patch made a reference to the PJSIP_SC_NULL enumeration value, which was added to pjproject 2.8 and above thus making it so Asterisk would fail to compile with prior versions of pjproject. This patch removes the reference, and instead initializes the value to '0'. ASTERISK-28886 #close Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7	5 years ago
George Joseph	41f3a7da4d	res_fax: Don't start a gateway if either channel is hung up ... When fax_gateway_framehook is called and a gateway hasn't already been started, the framehook gets the t38 state for both the current channel and the peer. That call trickles down to the channel driver which determines the state. If either channel is hung up (or in the process of being hung up), the channel driver's tech_pvt is going to be NULL which, in the case of chan_pjsip, will cause a segfault. * Added a hangup check for both the channel and peer channel before starting a fax gateway. * Added a check for NULL tech_pvt to chan_pjsip_queryoption so we don't attempt to reference a tech_pvt that's already gone. ASTERISK-28923 Reported by: Yury Kirsanov Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c	5 years ago
Joshua C. Colp	1c5e68580a	stream: Enforce formats immutability and ensure formats exist. ... Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
Kevin Harwell	fa3c8f94e0	chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet ... If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a digit begin before media, or rtp has been setup then it's possible the outgoing channel will hear a constant DTMF tone upon answering. This happens because when there is no media, or rtp chan_pjsip notifies the core to initiate inband DTMF. However, upon digit end if media, and rtp become available then chan_pjsip does not notify the core to stop inband DTMF. Thus the tone continues playing. This patch makes it so chan_pjsip only notifies the core to start inband DTMF in only the required cases. Now if there is no media, or rtp availabe upon digit begin chan_pjsip does nothing, but tells the core it handled it. ASTERISK-28817 #close Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5	6 years ago
Michael Neuhauser	5562fb2ea0	chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active ... Do not hang up a PJSIP channel on RTP timeout if that channel is in a direct-media bridge. Also reset the time of the last received RTP packet when direct-media ends (wait full rtp_timeout period before checking first time after audio came back to Asterisk). ASTERISK-28774 Reported-by: Michael Neuhauser Change-Id: I8b62012be7685849e8fb2b1c5dd39d35313ca2d1	6 years ago
Paulo Vicentini	ed2a7e3eaf	chan_pjsip: Check audio frame when remote SSRC changes. ... If the SSRC of a received RTP packet differed from the previous SSRC an SSRC change control frame would be queued ahead of the media frame. In the case of audio this would result in the format of the audio frame not being checked, and if it differed or was not allowed then it could cause the call to drop due to failure to set up a translation path. The chan_pjsip module will now no longer assume the first frame will be the audio frame and instead goes through the complete list to find it. ASTERISK-28759 Change-Id: I6d854cc523f343e299a615636fc65bdbd5f809ec	6 years ago
Sean Bright	313189aae2	chan_pjsip: Ignore RTP that we haven't negotiated ... If chan_pjsip receives an RTP packet whose payload differs from the channel's native format, and asymmetric_rtp_codec is disabled (the default), Asterisk will switch the channel's native format to match that of the incoming packet without regard to the negotiated payloads. We now check that the received frame is in a format we have negotiated before switching payloads which results in these packets being dropped instead of causing the session to terminate. ASTERISK-28139 #close Reported by: Paul Brooks Change-Id: Icc3b85cee1772026cee5dc1b68459bf9431c14a3	6 years ago
lvl	c03f50c1c8	chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel ... ASTERISK-28086 #close Change-Id: Ib3baadc89b9f0477a6f25a63861433812368c5ea	6 years ago
Torrey Searle	b43cdc7f1e	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	6 years ago
Friendly Automation	2205e25073	Merge "core: Add H.265/HEVC passthrough support"	6 years ago
Florian Floimair	c18983207d	core: Add H.265/HEVC passthrough support ... This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2	6 years ago
Joshua Colp	c358da472e	chan_pjsip: Relock correct channel during "fax" redirect. ... When fax detection occurs on an outbound PJSIP channel the redirect operation will result in a masquerade occurring and the underlying channel on the session changing. The code incorrectly relocked the new channel instead of the old channel when returning. This resulted in the new channel being locked indefinitely. The code now always acts on the expected channel. ASTERISK-28538 Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3	6 years ago
Dan Cropp	e52fbae00f	chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS ... Previously, when a Transfer (REFER) was performed, chan_pjsip would set the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not reflect a successful/unsuccessful transfer the way chan_sip did. Added a callback module to process the refer subscription information. Now depends on res_pjsip_pubsub so call transfer progress can be monitored and reported ASTERISK-26968 #close Reported-by: Dan Cropp Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc	6 years ago
agupta	d2f7b22640	chan_pjsip.c: Check for channel and session to not be NULL in hangup ... We have seen some rare case of segmentation fault in hangup function and we could notice that channel pointer was NULL. Debug log shows that there is a 200 OK answer and SIP timeout at the same time. It looks that while the SIP session was being destroyed due to timeout call hangup due to answer event lead to race condition and channel is being destroyed from two different places. The check ensures we check it not to be NULL before freeing it. ASTERISK-25371 Change-Id: I19f6566830640625e08f7b87bfe15758ad33a778	6 years ago
Alexei Gradinari	466a17964f	pjsip: replace 180 by 183 if SDP negotiation has completed ... The caller endpoint hears dead silence if a callee replies 180 (without SDP) and the caller already received 183 (with SDP). It happens because Asterisk sends 180 (WITH SDP) to the caller, there are not incoming RTP packets from the callee and Asterisk does not generate inband ringing, so there are not any outgoing RTP packets to the caller. This patch replaces 180 by 183 if SDP negotiation has completed, as if the caller endpoint is configured with "inband_progress=yes". In this case Asterisk will generate inband ringing untill Asterisk receive incoming RTP packets from the callee. ASTERISK-27994 #close Change-Id: I7450b751083ec30d68d6abffe922215a15ae5a73	7 years ago
Salah Ahmed	5009d6d97a	chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info ... When the dtmf_mode on an endpoint is configured as "auto_info" Asterisk will produce an inband DTMF tone alongside an INFO message when sending DTMF. ASTERISK-28371 Change-Id: I1380b82f006e110a1b83fbb50c9873edd13a5d9a	7 years ago
Torrey Searle	4661c08549	chan_pjsip: add a flag to ignore 183 responses if no SDP present ... chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	7 years ago
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	7 years ago