5095 Commits (17)

Author	SHA1	Message	Date
Ben Ford	17f177ce06	AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite. ... When Asterisk sends a reinvite negotiating T38 faxing, it's possible a crash can occur if the response contains a m=image and zero port. The reinvite callback code now checks session_media to see if it is null or not before trying to access the udptl variable on it. ASTERISK-29305 Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9	6 years ago
Kevin Harwell	b4f221dcf3	AST-2021-002: Remote crash possible when negotiating T.38 ... When an endpoint requests to re-negotiate for fax and the incoming re-invite is received prior to Asterisk sending out the 200 OK for the initial invite the re-invite gets delayed. When Asterisk does finally send the re-inivite the SDP includes streams for both audio and T.38. This happens because when the pending topology and active topologies differ (pending stream is not in the active) in the delayed scenario the pending stream is appended to the active topology. However, in the fax case the pending stream should replace the active. This patch makes it so when a delay occurs during fax negotiation, to or from, the audio stream is replaced by the T.38 stream, or vice versa instead of being appended. Further when Asterisk sent the re-invite with both audio and T.38, and the endpoint responded with a declined T.38 stream then Asterisk would crash when attempting to change the T.38 state. This patch also puts in a check that ensures the media state has a valid fax session (associated udptl object) before changing the T.38 state internally. ASTERISK-29203 #close Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09	6 years ago
Alexander Traud	251e50a132	rtp: Enable srtp replay protection ... Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	6 years ago
Ivan Poddubnyi	3f802a33d9	res_pjsip_diversion: Fix adding more than one histinfo to Supported ... New responses sent within a PJSIP sessions are based on those that were sent before. Therefore, adding/modifying a header once causes it to be sent on all responses that follow. Sending 181 Call Is Being Forwarded many times first adds "histinfo" duplicated more and more, and eventually overflows past the array boundary. This commit adds a check preventing adding "histinfo" more than once, and skipping it if there is no more space in the header. Similar overflow situations can also occur in res_pjsip_path and res_pjsip_outbound_registration so those were also modified to check the bounds and suppress duplicate Supported values. ASTERISK-29227 Reported by: Ivan Poddubny Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322	6 years ago
Sean Bright	8fb9da2fd9	res_rtp_asterisk.c: Fix signed mismatch that leads to overflow ... ASTERISK-29205 #close Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea	6 years ago
Torrey Searle	cc22e3bc1c	res/res_pjsip_diversion: prevent crash on tel: uri in History-Info ... Add a check to see if the URI is a Tel URI and prevent crashing on trying to retrieve the reason parameter. ASTERISK-29191 ASTERISK-29219 Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37	6 years ago
Kevin Harwell	fab1ef83b2	AST-2020-001 - res_pjsip: Return dialog locked and referenced ... pjproject returns the dialog locked and with a reference. However, in Asterisk the method that handles this decrements the reference and removes the lock prior to returning. This makes it possible, under some circumstances, for another thread to free said dialog before the thread that created it attempts to use it again. Of course when the thread that created it tries to use a freed dialog a crash can occur. This patch makes it so Asterisk now returns the newly created dialog both locked, and with an added reference. This allows the caller to de-reference, and unlock the dialog when it is safe to do so. In the case of a new SIP Invite the lock, and reference are now held for the entirety of the new invite handling process. Otherwise it's possible for the dialog, or its dependent objects, like the transaction, to disappear. For example if there is a TCP transport error. ASTERISK-29057 #close Change-Id: I5ef645a47829596f402cf383dc02c629c618969e	6 years ago
Ben Ford	32c5ffc354	AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. ... If Asterisk sends out an INVITE and receives a challenge with a different nonce value each time, it will continuously send out INVITEs, even if the call is hung up. The endpoint must be configured for outbound authentication for this to occur. A limit has been set on outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. ASTERISK-29013 Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7	6 years ago
Kevin Harwell	7f53752619	res_pjsip, res_pjsip_session: initialize local variables ... This patch initializes a couple of local variables to some default values. Interestingly, in the 'pj_status_t dlg_status' case the value not being initialized caused memory to grow, and not be recovered, in the off nominal path (at least on my machine). Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7	6 years ago
Nick French	f7835bcf13	res_pjsip_session: Restore calls to ast_sip_message_apply_transport() ... Commit 44bb0858cb ("debugging: Add enough to choke a mule") accidentally removed calls to ast_sip_message_apply_transport when it was attempting to just add debugging code. The kiss of death was saying that there were no functional changes in the commit comment. This makes outbound calls that use the 'flow' transport mechanism fail, since this call is used to relay headers into the outbound INVITE requests. ASTERISK-29124 #close Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a	6 years ago
Jean Aunis	b11f1242c5	resource_endpoints.c: memory leak when providing a 404 response ... When handling a send_message request to a non-existing endpoint, the response's body is overriden and not properly freed. ASTERISK-29108 Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8	6 years ago
Sean Bright	1b11f143d5	pbx.c: On error, ast_add_extension2_lockopt should always free 'data' ... In the event that the desired extension already exists, ast_add_extension2_lockopt() will free the 'data' it is passed before returning an error, so we should not be freeing it ourselves. Additionally, there were two places where ast_add_extension2_lockopt() could return an error without also freeing the 'data' pointer, so we add that. ASTERISK-29097 #close Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae	6 years ago
lvl	9511e8d379	res_musiconhold: Load all realtime entries, not just the first ... ASTERISK-29099 Change-Id: I45636679c0fb5a5f59114c8741626631a604e8a6	6 years ago
Holger Hans Peter Freyther	0c1d9d4730	res_pjsip_sdp_rtp: Fix accidentally native bridging calls ... Stop advertising RFC2833 support on the rtp_engine when DTMF mode is auto but no tel_event was found inside SDP file. On an incoming call create_rtp will be called and when session->dtmf is set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without looking at the SDP file. Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND but continued to advertise RFC2833 support. This meant the native_rtp bridge would falsely consider the two channels as compatible. In addition to changing the DTMF mode we now set or remove the AST_RTP_PROPERTY_DTMF. The property is checked in ast_rtp_dtmf_compatible and called by native_rtp_bridge_compatible. ASTERISK-29051 #close Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287	6 years ago
Torrey Searle	1746703d25	res_pjsip_diversion: fix double 181 ... Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice, resulting in to 181 being generated. Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab	6 years ago
Sean Bright	a68e33d91b	res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs ... Change-Id: I41e77a04e4a523f4ed61a7a20b738ffd42be441e	6 years ago
Joshua C. Colp	26873338d7	res_pjsip_session: Fix stream name memory leak. ... When constructing a stream name based on the media type and position the allocated name was not being freed causing a leak. Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de	6 years ago
Sean Bright	6ba11fed7e	res_musiconhold: Start playlist after initial announcement ... Only track our sample offset if we are playing a non-announcement file, otherwise we will skip that number of samples when we start playing the first MoH file. ASTERISK-24329 #close Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc	6 years ago
Joshua C. Colp	1bf404e15d	res_pjsip_session: Fix session reference leak. ... The ast_sip_dialog_get_session function returns the session with reference count increased. This was not taken into account and was causing sessions to remain around when they should not be. ASTERISK-29089 Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8	6 years ago
Michal Hajek	4128bdfdb9	res_stasis.c: Add compare function for bridges moh container ... Sometimes not play MOH on bridge. ASTERISK-29081 Reported-by: Michal Hajek <michal.hajek@daktela.com> Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232	6 years ago
Sean Bright	617f62b6ab	res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined ... Change-Id: Id4852c26e9c412af8e37b5dd3c15da9453ad3276	6 years ago
Torrey Searle	71820ea114	res_pjsip_diversion: implement support for History-Info ... Implemention of History-Info capable of interworking with Diversion Header following RFC7544 ASTERISK-29027 #close Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6	6 years ago
George Joseph	a92119306f	res_pjsip_session: Fix issue with COLP and 491 ... The recent 491 changes introduced a check to determine if the active and pending topologies were equal and to suppress the re-invite if they were. When a re-invite is sent for a COLP-only change, the pending topology is NULL so that check doesn't happen and the re-invite is correctly sent. Of course, sending the re-invite sets the pending topology. If a 491 is received, when we resend the re-invite, the pending topology is set and since we didn't request a change to the topology in the first place, pending and active topologies are equal so the topologies-equal check causes the re-invite to be erroneously suppressed. This change checks if the topologies are equal before we run the media state resolver (which recreates the pending topology) so that when we do the final topologies-equal check we know if this was a topology change request. If it wasn't a change request, we don't suppress the re-invite even though the topologies are equal. ASTERISK-29014 Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314	6 years ago
George Joseph	1cd184250f	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	6 years ago
George Joseph	d3c67d585f	res_pjsip_session: Handle multi-stream re-invites better ... When both Asterisk and a UA send re-invites at the same time, both send 491 "Transaction in progress" responses to each other and back off a specified amount of time before retrying. When Asterisk prepares to send its re-invite, it sets up the session's pending media state with the new topology it wants, then sends the re-invite. Unfortunately, when it received the re-invite from the UA, it partially processed the media in the re-invite and reset the pending media state before sending the 491 losing the state it set in its own re-invite. Asterisk also was not tracking re-invites received while an existing re-invite was queued resulting in sending stale SDP with missing or duplicated streams, or no re-invite at all because we erroneously determined that a re-invite wasn't needed. There was also an issue in bridge_softmix where we were using a stream from the wrong topology to determine if a stream was added. This also caused us to erroneously determine that a re-invite wasn't needed. Regardless of how the delayed re-invite was triggered, we need to reconcile the topology that was active at the time the delayed request was queued, the pending topology of the queued request, and the topology currently active on the session. To do this we need a topology resolver AND we need to make stream named unique so we can accurately tell what a stream has been added or removed and if we can re-use a slot in the topology. Summary of changes: * bridge_softmix: * We no longer reset the stream name to "removed" in remove_all_original_streams(). That was causing multiple streams to have the same name and wrecked the checks for duplicate streams. * softmix_bridge_stream_sources_update() was checking the old_stream to see if it had the softmix prefix and not considering the stream as "new" if it did. If the stream in that slot has something in it because another re-invite happened, then that slot in old might have a softmix stream but the same stream in new might actually be a new one. Now we check the new_stream's name instead of the old_stream's. * stream: * Instead of using plain media type name ("audio", "video", etc) as the default stream name, we now append the stream position to it to make it unique. We need to do this so we can distinguish multiple streams of the same type from each other. * When we set a stream's state to REMOVED, we no longer reset its name to "removed" or destroy its metadata. Again, we need to do this so we can distinguish multiple streams of the same type from each other. * res_pjsip_session: * Added resolve_refresh_media_states() that takes in 3 media states and creates an up-to-date pending media state that includes the changes that might have happened while a delayed session refresh was in the delayed queue. * Added is_media_state_valid() that checks the consistency of a media state and returns a true/false value. A valid state has: * The same number of stream entries as media session entries. Some media session entries can be NULL however. * No duplicate streams. * A valid stream for each non-NULL media session. * A stream that matches each media session's stream_num and media type. * Updated handle_incoming_sdp() to set the stream name to include the stream position number in the name to make it unique. * Updated the ast_sip_session_delayed_request structure to include both the pending and active media states and updated the associated delay functions to process them. * Updated sip_session_refresh() to accept both the pending and active media states that were in effect when the request was originally queued and to pass them on should the request need to be delayed again. * Updated sip_session_refresh() to call resolve_refresh_media_states() and substitute its results for the pending state passed in. * Updated sip_session_refresh() with additional debugging. * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE to pjproject if a transaction is in progress. This stops us from creating a partial pending media state that would be invalid later on. * Updated reschedule_reinvite() to clone both the current pending and active media states and pass them to delay_request() so the resolver can tell what the original intention of the re-invite was. * Added a large unit test for the resolver. ASTERISK-29014 Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb	6 years ago
Sungtae Kim	519bb01d5f	res_stasis.c: Added video_single option for bridge creation ... Currently, it was not possible to create bridge with video_mode single. This made hard to put the bridge in a vidoe_single mode. So, added video_single option for Bridge creation using the ARI. This allows create a bridge with video_mode single. ASTERISK-29055 Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae	6 years ago
Patrick Verzele	d2cc36e4a8	res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly ... Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again. Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6	6 years ago
Joshua C. Colp	cc60a0fb6f	parking: Copy parker UUID as well. ... When fixing issues uncovered by GCC10 a copy of the parker UUID was removed accidentally. This change restores it so that the subscription has the data it needs. ASTERISK-29042 Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a	6 years ago
Nickolay Shmyrev	7f5cc9ad28	res_speech: Bump reference on format object ... Properly bump reference on format object to avoid memory corruption on double free ASTERISK-29040 #close Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3	6 years ago
Torrey Searle	25f0588c0f	res_pjsip_diversion: handle 181 ... Adapt the response handler so it also called when 181 is received. In the case 181 is received, also generate the 181 response. ASTERISK-29001 #close Change-Id: I73cfee46a8ca85371280ebdb38674f8fde7510df	6 years ago
Joshua C. Colp	a815f02597	res_pjsip_session: Don't aggressively terminate on failed re-INVITE. ... Per the RFC when an outgoing re-INVITE is done we should only terminate the dialog if a 481 or 408 is received. ASTERISK-29033 Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503	6 years ago
Sean Bright	315b5f3929	res_musiconhold.c: Use ast_file_read_dir to scan MoH directory ... Two changes of note in this patch: * Use ast_file_read_dir instead of opendir/readdir/closedir * If the files list should be sorted, do that at the end rather than as we go which improves performance for large lists Change-Id: Ic7e9c913c0f85754c99c74c9cf6dd3514b1b941f	6 years ago
George Joseph	50a98c37c8	scope_trace: Add/update utilities ... * Added a AST_STREAM_STATE_END sentinel * Add ast_stream_to_str() * Add ast_stream_state_to_str() * Add ast_stream_get_format_count() * Add ast_stream_topology_to_str() * Add ast_stream_topology_get_active_count() * Add ast_format_cap_append_names() * Add ast_sip_session_get_name() Change-Id: I132eb5971ea41509c660f64e9113cda8c9013b0b	6 years ago
Sean Bright	4167cc1be4	res_musiconhold.c: Prevent crash with realtime MoH ... The MoH class internal file vector is potentially being manipulated by multiple threads at the same time without sufficient locking. Switch to a reference counted list and operate on copies where necessary. ASTERISK-28927 #close Change-Id: I479c5dcf88db670956e8cac177b5826c986b0217	6 years ago
George Joseph	5042ebf4ff	res_pjsip_session: Ensure reused streams have correct bundle group ... When a bundled stream is removed, its bundle_group is reset to -1. If that stream is later reused, the bundle parameters on session media need to be reset correctly it could mistakenly be rebundled with a stream that was removed and never reused. Since the removed stream has no rtp instance, a crash will result. Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7	6 years ago
Joshua C. Colp	4ac3d4cdbd	res_pjsip_registrar: Don't specify an expiration for static contacts. ... Statically configured contacts on an AOR don't have an expiration time so when adding them to the resulting 200 OK if an endpoint registers ensure they are marked as such. ASTERISK-28995 Change-Id: I9f0e45eb2ccdedc9a0df5358634a19ccab0ad596	6 years ago
sungtae kim	ade20df66b	stasis_bridge.c: Fixed wrong video_mode shown ... Currently, if the bridge has created by the ARI, the video_mode parameter was not shown in the BridgeCreated event correctly. Fixed it and added video_mode shown in the 'bridge show <bridge id>' cli. ASTERISK-28987 Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295	6 years ago
Joshua C. Colp	a613e07e76	websocket / pjsip: Increase maximum packet size. ... When dealing with a lot of video streams on WebRTC the resulting SDPs can grow to be quite large. This effectively doubles the maximum size to allow more streams to exist. The res_http_websocket module has also been changed to use a buffer on the session for reading in packets to ensure that the stack space usage is not excessive. Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01	6 years ago
Nickolay Shmyrev	c3b3b681fb	res_http_websocket: Avoid reading past end of string ... We read beyond the end of the buffer when copying the string out of the buffer when we used ast_copy_string() because the original string was not null terminated. Instead switch to ast_strndup() which does not exhibit the same behavior. ASTERISK-28975 #close Change-Id: Ib4a75cffeb1eb8cf01136ef30306bd623e531a2a	6 years ago
sungtae kim	7c12303c4c	res_pjsip.c: Added disable_rport option for pjsip.conf ... Currently when the pjsip making an outgoing request, it keep adding the rport parameter in a request message as a default. This causes unexpected rport handle at the other end. Added option for disable this behaviour in the pjsip.conf. This is a system option, but working as a gloabl option. ASTERISK-28959 Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc	6 years ago
Nickolay Shmyrev	b96b4ec438	res_http_websocket.c: Continue reading after ping/pong ... Do not return error if the client received ping frame while looking for a string and just wait for another frame. ASTERISK-28958 #close Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922	6 years ago
Kevin Harwell	bdcaa275da	PJSIP_MEDIA_OFFER: override configuration on refresh ... When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs on refresh unless they had a shared codec between the two. This patch makes it so whatever is set using PJSIP_MEDIA_OFFER is used when creating the SDP for a refresh no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	6 years ago
Joshua C. Colp	f129bbe387	res_pjsip: Apply AOR outbound proxy to static contacts. ... The outbound proxy for an AOR was not being applied to any statically configured Contacts. This resulted in the OPTIONS requests being sent to the wrong target. This change sets the outbound proxy on statically configured contacts once the AOR configuration is done being applied. ASTERISK-28965 Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0	6 years ago
Università di Bologna - CESIA VoIP	6462987a03	res_corosync: Fix crash in huge distributed environment. ... 1) Fix memory-leaks Added code to release ast_events extracted from corosync and stasis messages 2) Clean stasis cache when a member of the corosync cluster leaves the group Added code to remove from the stasis cache of the members remained on the group all the messages with the EID of the left member. If the device states of the left member remain in the stasis cache of other members, they will not be updated anymore and high priority cached values, like BUSY, will take precedence over current device states. 3) Stop corosync event propagation when node is not joined to the group Updated dispatch_thread_handler code to detect when asterisk is not joined to the corosync group and added some condition in publish_event_to_corosync code to send corosync messages only when joined. When a node is not joined its corosync daemon can't send messages: the cpg_mcast_joined function append new messages to the FIFO buffer until it's full and then it blocks indefinitely. In this scenario if the stasis_message_cb callback, registered by res_corosync to handle stasis messages, try to send a corosync messages, the thread of the stasis thread-pool will be blocked until the node join the corosync cluster. ASTERISK-28888 Reported by: Università di Bologna - CESIA VoIP Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2	6 years ago
Moises Silva	6533d02e14	res_http_websocket: Add payload masking to the websocket client ... ASTERISK-28949 Change-Id: Id465030f2b1997b83d408933fdbabe01827469ca	6 years ago
Joshua C. Colp	9589f97c9e	res_pjsip_session: Preserve label on incoming re-INVITE. ... When a re-INVITE is received we create a new set of streams that are then swapped in as the active streams. We did not preserve the SDP label from the previous streams, resulting in the label getting lost. This change ensures that if an SDP label is present on the previous stream then it is set on the new stream. ASTERISK-28953 Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445	6 years ago
Joshua C. Colp	79ebbca3a7	res_sorcery_memory_cache: Disallow per-object expire with full backend. ... The AMI action and CLI command did not take into account the properties of full backend caching. This resulted in an expired object remaining removed until a full backend update occurred, instead of having the object updated when needed. This change makes it so that the AMI action and CLI command for object expire will now fail instead of putting the cache into an undesired state. If full backend caching is enabled then only operations which act on the entire cache are available. ASTERISK-28942 Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4	6 years ago
Walter Doekes	4bc0f24e97	res_pjsip: Include <pjsip_ua.h> instead of internal "pjsua-lib/pjsua.h" ... Change-Id: I24b5453df412232cf7f9a171ea4a34b35ad3ae78	6 years ago
Kevin Harwell	3b99479934	pjproject: Upgrade bundled version to pjproject 2.10 ... This patch makes the usual necessary changes when upgrading to a new version pjproject. For instance, version number bump, patches removed from third-party, new *.md5 file added, etc.. This patch also includes a change to the Asterisk pjproject Makefile to explicitly create the 'source/pjsip-apps/lib' directory. This directory is no longer there by default so needs to be added so the Asterisk malloc debug can be built. This patch also includes some minor changes to Asterisk that were a result of the upgrade. Specifically, there was a backward incompatibility change made in 2.10 that modified the "expires header" variable field from a signed to an unsigned value. This potentially effects comparison. Namely, those check for a value less than zero. This patch modified a few locations in the Asterisk code that may have been affected. Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to check a minimum version of pjproject at compile time. ASTERISK-28899 #close Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81	6 years ago
sungtae kim	37f8adacd1	res_ari: Fix create channel request channelId parameter parsing ... If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly. Fixed it to parse the channelId, other_channel_id parameter correclty. ASTERISK-28948 Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157	6 years ago