8442 Commits (d0f01af913cad9ec5fe2a496794e960e00f2f474)

Author	SHA1	Message	Date
George Joseph	d0f01af913	chan_dahdi: Address gcc9 issues ... Fixed format-truncation issues in chan_dahdi.c and sig_analog.c. Since they're related to fields provided by dahdi-tools we can't change the buffer sizes so we're just checking the return from snprintf and printing an errior if we overflow. Change-Id: Idc1f3c1565b88a7d145332a0196074b5832864e5	6 years ago
agupta	67841b8f55	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
Friendly Automation	e4d91ab4b2	Merge "pjsip: replace 180 by 183 if SDP negotiation has completed" into 13	6 years ago
Guido Falsi	ac4921c373	chan_dahdi: add missing include. ... After some definitions have been moved to asterisk/mwi.h the files channels/chan_dahdi.h channels/sig_pri.c are missing this new include. ASTERISK-28427 #close Change-Id: Ia8cc595eeda653324643f40dcd9799d4c3f0ac91	6 years ago
Alexei Gradinari	595d60846a	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
George Joseph	4337895aee	Fixes for GCC 9 ... Various fixes for issues caught by gcc 9. Mostly snprintf trying to copy to a buffer potentially too small. ASTERISK-28412 Change-Id: I9e85a60f3c81d46df16cfdd1c329ce63432cf32e	7 years ago
Kevin Harwell	4ea20c9c85	mwi core: Move core MWI functionality into its own files ... There is enough MWI functionality to warrant it having its own 'c' and header files. This patch moves all current core MWI data structures, and functions into the following files: main/mwi.h main/mwi.c Note, code was simply moved, and not modified. However, this patch is also in preparation for core MWI changes, and additions to come. Change-Id: I9dde8bfae1e7ec254fa63166e090f77e4d3097e0	7 years ago
George Joseph	1a4fbaa52e	Merge "chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info" into 13	7 years ago
Salah Ahmed	a9a0303544	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
Ben Ford	4853fc2218	build: Fix compiler warnings/errors. ... The compiler complained about a couple of variables that weren't initialized but were being used. Initializing them to NULL resolves the warnings/errors. ASTERISK-28362 #close Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b	7 years ago
Alexei Gradinari	6e20e071a9	pjsip: restrict function PJSIP_PARSE_URI to parse only SIP/SIPS URIs ... The next usage of PJSIP_PARSE_URI will crash asterisk ${PJSIP_PARSE_URI(tel:+1234567890,host)} or ${PJSIP_PARSE_URI(192.168.1.1:5060,host)} The function pjsip_parse_uri successfully parses then, but returns struct pjsip_other_uri *. This patch restricts parsing only SIP/SIPS URIs. Change-Id: I16f255c2b86a80a67e9f9604b94b129a381dd25e	7 years ago
Sean Bright	53aa750839	chan_sip: Ensure 'qualifygap' isn't negative ... Passing negative intervals to the scheduler rips a hole in the space-time continuum. ASTERISK-25792 #close Reported by: Paul Sandys Change-Id: Ie706f21cee05f76ffb6f7d89e9c867930ee7bcd7	7 years ago
Kevin Harwell	9717d1672c	Merge "chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function" into 13	7 years ago
Friendly Automation	4f6daa0aff	Merge "chan_pjsip: add a flag to ignore 183 responses if no SDP present" into 13	7 years ago
Torrey Searle	cbc704c5ec	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
Sean Bright	1cb6466268	Replace calls to strtok() with strtok_r() ... strtok() uses a static buffer, making it not thread safe. Change-Id: Icce265153e1e65adafa8849334438ab6d190e541	7 years ago
cirillor	5065f31fca	chan_dahdi: Add logical group at DAHDIChannel event and CHANNEL function ... Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function. ASTERISK-28317 Change-Id: Ic1f834cd53982a9707a9748395ee746d6575086a	7 years ago
sungtae kim	f6689547ae	chan_pjsip: Changed to continued after invalid media for pjsip show channelstats ... Currently, the pjsip show channelstats cli does not show channel's stats after hits the invalid channel info. This makes hard to use this cli. Changed to keep iterate after hits the invalid channel info. ASTERISK-28292 Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c	7 years ago
Giuseppe Sucameli	f4afd097af	chan_sip: Fix leak using contact ACL ... Free old peer's contactacl before overwrite it within build_peer. ASTERISK-28194 Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c	7 years ago
Corey Farrell	0a9904e1c6	astobj2: Eliminate usage of legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. ao2_container_alloc is now restricted to modules only and is being removed from Asterisk 17. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	b7b581f209	Merge "stasis: Add internal filtering of messages." into 13	7 years ago
Alexei Gradinari	b6d0fbda9d	pjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI ... New dialplan function PJSIP_PARSE_URI added to parse an URI and return a specified part of the URI. This is useful when need to get part of the URI instead of cutting it using a CUT function. For example to get 'user' part of Remote URI ${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)} ASTERISK-28144 #close Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a	7 years ago
Joshua Colp	d748ed4147	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
George Joseph	8b965c386a	Merge "chan_sip: Attempt ast_do_pickup in handle_invite_replaces" into 13	7 years ago
Jasper Hafkenscheid	cf193d53ad	chan_sip: Attempt ast_do_pickup in handle_invite_replaces ... When a call pickup is performed using and invite with replaces header the ast_do_pickup method is attempted and a PICKUP stasis message is sent. ASTERISK-28081 #close Reported-by: Luit van Drongelen Change-Id: Ieb1442027a3ce6ae55faca47bc095e53972f947a	7 years ago
Alexei Gradinari	bfe3821800	pjsip: new endpoint's options to control Connected Line updates ... This patch adds new options 'trust_connected_line' and 'send_connected_line' to the endpoint. The option 'trust_connected_line' is to control if connected line updates are accepted from this endpoint. The option 'send_connected_line' is to control if connected line updates can be sent to this endpoint. The default value is 'yes' for both options. Change-Id: I16af967815efd904597ec2f033337e4333d097cd	7 years ago
Corey Farrell	54a1fbe428	astobj2: Eliminate usage of legacy container allocation macros. ... These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are still available for use but only in modules. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
Corey Farrell	c4b979a841	chan_sip: Tell module loader that chan_sip is extended support. ... Change-Id: I33508c134b1be888b8884f5dcfee19087634e415	7 years ago
pk16208	84c574bb8b	chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI ... With tls and udp enabled asterisk generates a warning about sending message via udp instead of tls. sip notify command via cli works as expected and without warning. asterisk has to set the connection information accordingly to connection and not on presumption ASTERISK-28057 #close Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e	7 years ago
Walter Doekes	d226458c5b	optional_api: Remove unused nonoptreq fields ... As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	7 years ago
Jaco Kroon	2a5d408733	chan_sip: improved ip:port finding of peers for non-UDP transports. ... Also remove function peer_ipcmp_cb since it's not used (according to rmudgett). Prior to b2c4e8660a (ASTERISK_27457) insecure=port was the defacto standard. That commit also prevented insecure=port from being applied for sip/tcp or sip/tls. Into consideration there are three sets of behaviour: 1. "previous" - before the above commit. 2. "current" - post above commit, pre this one. 3. "new" - post this commit. The problem that the above commit tried to address was guests over TCP. It succeeded in doing that but broke transport!=udp with host!=dynamic. This commit attempts to restore sane behaviour with respect to transport!=udp for host!=dynamic whilst still retaining the guest users over tcp. It should be noted that when looking for a peer, two passes are made, the first pass doesn't have SIP_INSECURE_PORT set for the searched-for peer, thus looking for full matches (IP + Port), the second pass sets SIP_INSECURE_PORT, thus expecting matches on IP only where the matched peer allows for that (in the author's opinion: UDP with insecure=port, or any TCP based, non-dynamic host). In previous behaviour there was special handling for transport=tcp|tls whereby a peer would match during the first pass if the utilized transport was TCP|TLS (and the peer allowed that specific transport). This behaviour was wrong, or dubious at best. Consider two dynamic tcp peers, both registering from the same IP (NAT), in this case either peer could match for connections from an IP. It's also this behaviour that prevented SIP guests over tcp. The above referenced commit removed this behaviour, but kept applying the SIP_INSECURE_PORT only to WS|WSS|UDP. Since WS and WSS is also TCP based, the logic here should fall into the TCP category. This patch updates things such that the previously non-explicit (TCP behaviour) transport test gets performed explicitly (ie, matched peer must allow for the used transport), as well as the indeterministic source-port nature of the TCP protocol is taken into account. The new match algorithm now looks like: 1. As per previous behaviour, IP address is matched first. 2. Explicit filter with respect to transport protocol, previous behaviour was semi-implied in the test for TCP pure IP match - this now made explicit. 3. During first pass (without SIP_INSECURE_PORT), always match on port. 4. If doing UDP, match if matched against peer also has SIP_INSECURE_PORT, else don't match. 5. Match if not a dynamic host (for non-UDP protocols) 6. Don't match if this is WS|WSS, or we can't trust the Contact address (presumably due to NAT) 7. Match (we have a valid Contact thus if the IP matches we have no choice, this will likely only apply to non-NAT). To logic-test this we need a few different scenarios. Towards this end, I work with a set number of peers defined in sip.conf: [peer1] host=1.1.1.1 transport=tcp [peer2] host=1.1.1.1 transport=udp [peer3] host=1.1.1.1 port=5061 insecure=port transport=udp [peer4] host=1.1.1.2 transport=udp,tcp [peer5] host=dynamic transport=udp,tcp Test cases for UDP: 1 - incoming UDP request from 1.1.1.1: - previous: - pass 1: * peer1 or peer2 if from port 5060 (indeterminate, depends on peer ordering) * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 - current: as per previous. - new: - pass 1: * peer2 if from port 5060 * peer3 if from port 5061 * peer5 if registered from 1.1.1.1 and source port matches - pass 2: * peer3 2 - incoming UDP request from 1.1.1.2: - previous: - pass 1: * peer5 if registered from 1.1.1.2 and port matches * peer4 if source port is 5060 - pass 2: * no match (guest) - current: as previous. - new as previous (with the variation that if peer5 didn't have udp as allowed transport it would not match peer5 whereas previous and current code could). 3 - incoming UDP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address and source port matches. - pass 2: * peer5 if insecure=port is additionally set. * no match (guest) - current - as per previous - new - as per previous Test cases for TCP based transports: 4 - incoming TCP request from 1.1.1.1 - previous: - pass 1 (indeterministic, depends on ordering of peers in memory): * peer1; or * peer5 if peer5 registered from 1.1.1.1 (irrespective of source port); or * peer2 if the source port happens to be 5060; or * peer3 if the source port happens to be 5061. - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer1 or peer2 if from source port 5060 * peer3 if from source port 5060 * peer5 if registered as 1.1.1.1 and source port matches - pass 2: * no match (guest) - new: - pass 1: * peer 1 if from port 5060 * peer 5 if registered and source port matches - pass 2: * peer 1 5 - incoming TCP request from 1.1.1.2 - previous (indeterminate, depends on ordering): - pass 1: * peer4; or * peer5 if peer5 registered from 1.1.1.2 - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * no match (guest). - new: - pass 1: * peer4 if source port is 5060 * peer5 if peer5 registered as 1.1.1.2 and source port matches - pass 2: * peer4 6 - incoming TCP request from anywhere else: - previous: - pass 1: * peer5 if registered from that address - pass 2: cannot happen since pass 1 will always find a peer. - current: - pass 1: * peer5 if registered from that address and port matches. - pass 2: * no match (guest) - new: as per current. It should be noted the test cases don't make explicit mention of TLS, WS or WSS. WS and WSS previously followed UDP semantics, they will now enforce source port matching. TLS follow TCP semantics. The previous commit specifically tried to address test-case 6, but broke test-cases 4 and 5 in the process. ASTERISK-27881 #close Change-Id: I61a9804e4feba9c7224c481f7a10bf7eb7c7f2a2	7 years ago
alecdavis	6964bc37e0	chan_sip: remove unnecessary ast_channel_unlock(peer) as RAII looks after it ... Otherwise console output (get_refer_info): mutex 'peer' freed more times than we've locked! (get_refer_info): Error releasing mutex: Operation not permitted or (get_refer_info): attempted unlock mutex 'peer' without owning it! (__ast_read): 'peer' was locked here. ...dump_backtrace (get_refer_info): Error releasing mutex: Operation not permitted (__ast_read): mutex 'chan' freed more times than we've locked! ASTERISK-28011 #close Change-Id: I6e45f2764ba4f3273a943300f91ac9b461ac2893	7 years ago
Salah Ahmed	4aa91c6f11	dialplan_functions: wrong srtp use status report of a dialplan function ... If asterisk offer an endpoint with SRTP and that endpoint respond with non srtp, in that case channel(rtp,secure,audio) reply wrong status. Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key: Currently this flag has being set redundantly. In either case identical or different remote_key this flag has being set. So if we don't set it while we receive identical remote_key or non SRTP SDP response then we can take decision of srtp use by using that flag. ASTERISK-27999 Change-Id: I29dc2843cf4e5ae2604301cb4ff258f1822dc2d7	7 years ago
Alexander Traud	78855e13e8	chan_oss: Compile in Solaris 11. ... M_READ existed already and was conflicting in name. Change-Id: I02108e07ae7d2dc314fe1e6c706c17731095a3e4	7 years ago
Richard Mudgett	f94a310ca0	channel: Fix some more unprotected channel flag setting. ... Change-Id: I34c3b1201b1de539945bcfdcb264fff30332d48c	7 years ago
ktyerman	c6116a3234	chan_iax2: better handling for timeout and EINTR ... The iax2 module is not handling timeout and EINTR case properly. Mainly when there is an interupt to the kernel thread. In case of ast_io_wait recieves a signal, or timeout it can be an error or return 0 which eventually escapes the thread loop, so that it cant recieve any data. This then causes the modules receive queue to build up on the kernel and stop any communications via iax in asterisk. The proposed patch is for the iax module, so that timeout and EINTR does not exit the thread. ASTERISK-27705 Reported-by: Kirsty Tyerman Change-Id: Ib4c32562f69335869adc1783608e940c3535fbfb	7 years ago
Joshua Colp	7609249a22	Merge "crypto.h: Repair ./configure --with-ssl=PATH." into 13	7 years ago
Alexander Traud	2c3ad1e40d	crypto.h: Repair ./configure --with-ssl=PATH. ... ASTERISK-27908 Change-Id: Iac49d9f82faeb8a4611c6805906bd6d650b1b1d8	7 years ago
George Joseph	98da1971e3	chan_pjsip: Register for "BEFORE_MEDIA" responses ... chan_pjsip wasn't registering for "BEFORE_MEDIA" responses which meant it was not updating HANGUPCAUSE for 4XX responses. If the remote end sent a "180 Ringing", then a "486 Busy", the hangup cause was left at "180 Normal Clearing". * Removed chan_pjsip_incoming_response from the original session supplement (which was handling only "AFTER MEDIA") and added it to a new session supplement which accepts both "BEFORE_MEDIA" and "AFTER_MEDIA". * Also cleaned up some cleanup code in load module. ASTERISK-27902 Change-Id: If9b860541887aca8ac2c9f2ed51ceb0550fb007a	7 years ago
Corey Farrell	d893e57c90	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	8 years ago
George Joseph	42abc9c430	Merge "BuildSystem: Add DragonFly BSD." into 13	8 years ago
George Joseph	c5d19565fe	Merge "bridge_softmix: Forward TEXT frames" into 13	8 years ago
Alexander Traud	7b219311eb	BuildSystem: Add DragonFly BSD. ... ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557	8 years ago
George Joseph	be7d4faed5	bridge_softmix: Forward TEXT frames ... Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	8 years ago
Alexander Traud	fb9634bcb3	chan_vpb: Avoid GNU old-style field designator extension. ... clang 6.0 warned about this. Beside that, this change removes the used variable 'desc'. ASTERISK-27808 Change-Id: Ia26bdcc0a562c058151814511cfcf70ecafa595b	8 years ago
Richard Mudgett	12aa25b2e1	res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ... ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3	8 years ago
Jenkins2	0edc4ade93	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge" into 13	8 years ago
Richard Mudgett	72b16ee400	res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge ... There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82	8 years ago
Richard Mudgett	ea055386e0	chan_sip.c: Fix INVITE with replaces channel ref leak. ... Given the below call scenario: A -> Ast1 -> B C <- Ast2 <- B 1) A calls B through Ast1 2) B calls C through Ast2 3) B transfers A to C When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of the channel between Ast1 and Ast2. Channel ref leaks are easily seen in the CLI "core show channels" output. The leaked channels appear in the output but you can do nothing with them and they never go away unless you restart Asterisk. * Properly account for the channel refs when imparting a channel into a bridge when handling an INVITE with replaces in handle_invite_replaces(). The ast_bridge_impart() function steals a channel ref but the code didn't account for how many refs were held by the code at the time and which ref was stolen. * Eliminated RAII_VAR in handle_invite_replaces(). ASTERISK-27740 Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4	8 years ago
Jenkins2	b9eb86b56e	Merge "chan_sip: Peers with distinct source ports don't match, regardless of transport." into 13	8 years ago