asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	be8639f81b	Merge "res_rtp_asterisk: Allow remote SSRC to change on an RTP instance."	9 years ago
Joshua Colp	8cfff60cee	Merge "res_rtp_asterisk: Only learn a new source in learn state."	9 years ago
Jenkins2	41ab281f1a	Merge "pjsip_message_ip_updater: Fix issue handling "tel" URIs"	9 years ago
George Joseph	f78f5278ff	pjsip_message_ip_updater: Fix issue handling "tel" URIs sanitize_tdata was assuming all URIs were SIP URIs so when a non SIP uri was in the From, To or Contact headers, the unconditional cast of a non-pjsip_sip_uri structure to pjsip_sip_uri caused a segfault when trying to access uri->other_param. * Added PJSIP_URI_SCHEME_IS_SIP(uri) \|\| PJSIP_URI_SCHEME_IS_SIPS(uri) checks before attempting to cast or use the returned uri. ASTERISK-27152 Reported-by: Ross Beer Change-Id: Id380df790e6622c8058a96035f8b8f4aa0b8551f	9 years ago
Corey Farrell	1bf3dfffd7	AST-2017-006: Fix app_minivm application MinivmNotify command injection An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1	9 years ago
Joshua Colp	7f2a60fb38	res_rtp_asterisk: Only learn a new source in learn state. This change moves the logic which learns a new source address for RTP so it only occurs in the learning state. The learning state is entered on initial allocation of RTP or if we are told that the remote address for the media has changed. While in the learning state if we continue to receive media from the original source we restart the learning process. It is only once we receive a sufficient number of RTP packets from the new source that we will switch to it. Once this is done the closed state is entered where all packets that do not originate from the expected source are dropped. The learning process has also been improved to take into account the time between received packets so a flood of them while in the learning state does not cause media to be switched. Finally RTCP now drops packets which are not for the learned SSRC if strict RTP is enabled. ASTERISK-27013 Change-Id: I56a96e993700906355e79bc880ad9d4ad3ab129c	9 years ago
Joshua Colp	5ba82cedc6	res_rtp_asterisk: Allow remote SSRC to change on an RTP instance. When SDP renegotiation occurs it is possible for an RTP instance to be reused for a new stream, resulting in the remote SSRC changing if it is part of a bundle group. This change allows this and updates its mapping in the current bundle group. ASTERISK-27231 Change-Id: I6e3703974f236bc024c5dbe9bd43adae0c6fb490	9 years ago
Torrey Searle	33a648d4c6	res/res_pjsip_session: allow SDP answer to be regenerated If an SDP answer hasn't been sent yet, it's legal to change it. This is required for PJSIP_DTMF_MODE to work correctly, and can also have use in the future for updating codecs too. ASTERISK-27209 #close Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1	9 years ago
Sean Bright	7937d5b8b3	res_smdi: Clean up memory leak Change-Id: I1e33290929e1aa7c5b9cb513f8254f2884974de8	9 years ago
Richard Mudgett	f2c14f00b8	res_pjsip_session.c: Fix crash when declining an active stream. If a previously active stream is declined we could crash because the channel's thread is still using the stream while we are updating the topology in the serializer thread. * Defer removing any declined stream's handler until we have blocked the channel's thread with the channel lock. ASTERISK-27212 Change-Id: I50e1d3ef26f8e41948f4c411ee329aa3b960a420	9 years ago
Jenkins2	c86619bab8	Merge "res_xmpp: fix inverted return code check in OAuth"	9 years ago
Michael Kuron	83b81d1f8d	res_xmpp: fix inverted return code check in OAuth fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon success and -1 if the function is not available. This commit inverts the return code check so that an error is printed if the module is not loaded and not if it is loaded. ASTERISK-27207 #close Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb	9 years ago
Sean Bright	667986d875	res_calendar_icalendar: Properly handle recurring events When looking for recurring events, use the correct end time based on the configured 'timeframe.' ASTERISK-27174 #close Reported by: Mark Thompson Change-Id: Id90c3cfc79d561a5521d79be176683e225f2edef	9 years ago
Richard Mudgett	9e2b2a9837	res_pjsip: Fix prune_on_boot to remove only contacts for the host. * Check that the contact's reg_server matches the host's name before deleting any prune_on_boot contacts. We don't want to delete reliable transport contacts made with other servers if the ps_contacts database table is shared with other servers. Thanks to Ross Beer for pointing out that the original prune logic would delete reliable transport contacts from other servers. ASTERISK-27147 Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0	9 years ago
Andrey Egorov	15fbcc74d8	res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif Add ability to use tokens instead of passwords according to Google OAuth 2.0 protocol. ASTERISK-27169 Reported by: Andrey Egorov Tested by: Andrey Egorov Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db	9 years ago
Richard Mudgett	1bec781cce	res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown. The fix for the issue is broken up into three parts. This is part three which handles the client side of REGISTER requests. The registered contact may no longer be valid on the server when the transport used is reliable and the connection is broken. * Re-REGISTER our contact if the reliable transport is broken after registration completes. We attempt to re-REGISTER immediately to minimize the time we are unreachable. Time may have already passed between the connection being broken and the loss being detected. * Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's are still correct if an allocation failure happens. ASTERISK-27147 Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83	9 years ago
Richard Mudgett	82f4ade959	res_pjsip: Remove ephemeral registered contacts on transport shutdown. The fix for the issue is broken up into three parts. This is part two which handles the server side of REGISTER requests when rewrite_contact is enabled. Any registered reliable transport contact becomes invalid when the transport connection becomes disconnected. * Monitor the rewrite_contact's reliable transport REGISTER contact for shutdown. If it is shutdown then the contact must be removed because it is no longer valid. Otherwise, when the client attempts to re-REGISTER it may be blocked because the invalid contact is there. Also if we try to send a call to the endpoint using the invalid contact then the endpoint is not likely to see the request. The endpoint either won't be listening on that port for new connections or a NAT/firewall will block it. * Prune any rewrite_contact's registered reliable transport contacts on boot. The reliable transport no longer exists so the contact is invalid. * Websockets always rewrite the REGISTER contact address and the transport needs to be monitored for shutdown. * Made the websocket transport set a unique name since that is what we use as the ao2 container key. Otherwise, we would not know which transport we find when one of them shuts down. The names are also used for PJPROJECT debug logging. * Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state event. Now the global keep_alive_interval option, initially idle shutdown timer, and the server REGISTER contact monitor can work on wetsocket transports. * Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction. Now initially idle websockets will automatically shutdown. ASTERISK-27147 Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4	9 years ago
Richard Mudgett	1dcb92bba8	res_pjsip: PJSIP Transport state monitor refactor. The fix for the issue is broken up into three parts. This is part one which refactors the transport state monitor code to allow more modules to be able to monitor transports. * Pull the management of PJPROJECT's transport state callback code from res_pjsip_transport_management.c into res_pjsip. Now other modules can dynamically add and remove themselves from transport monitoring without worrying about breaking PJPROJECT's callback chain. * Add the ability for other modules to get a callback whenever a specific transport is shutdown. ASTERISK-27147 Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912	9 years ago
Richard Mudgett	ee5edfb050	res_pjsip_transport_management.c: Rename some variables. * Use monitored instead of the misleading keepalive name. Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6	9 years ago
Scott Griepentrog	4ed2733dde	res_pjsip_messaging: IPv6 receive address needs brackets When handling an incoming SIP MESSAGE, PJSIP attaches the IP address that the message was received from to the message in the variable PJSIP_RECVADDR. When the IP address is IPv6 the :PORT appended results in an unparseable mess. By using an additional bit flag on the pj_sockaddr_print call, the conventional use of brackets around the address is achieved. ASTERISK-27193 #close Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9	9 years ago
Jenkins2	08d22bedcc	Merge "res_rtp_asterisk: Make P2P bridge Asymmetric codec aware"	9 years ago
Torrey Searle	d430f718f5	res_rtp_asterisk: enable rtcp & QOS stats on native bridge Asterisk wasn't generating or forwarding RTCP packets when native bridge was activated. Also the stats weren't available via CHANNEL(qos). Now the RTCP stats are always calculated. ASTERISK-27158 #close Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b	9 years ago
Torrey Searle	a2dde59154	res_rtp_asterisk: Make P2P bridge Asymmetric codec aware Introduce a new property to rtp-engine to make it aware of the desire for assymetric codecs or not. If asymmetric codecs is not allowed, the bridge will compare read/write formats and shut down the p2p bridge if needed ASTERISK-26745 #close Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f	9 years ago
Jenkins2	df4bcdda2a	Merge "res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect"	9 years ago
Joshua Colp	62092bc114	res_pjsip_session: Release media resources on session end quicker. A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we are told to terminate the session we immediately release any media sessions associated with it. ASTERISK-27110 Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82	9 years ago
Joshua Colp	dcd846c321	Merge "res_pjsip_nat.c: Remove unnecessary CMP_STOP."	9 years ago
Jenkins2	e0aed61e96	Merge "Support GMIME 3.0"	9 years ago
Kevin Harwell	104a8047a5	res_pjsip_session/_sdp_rtp: Handling of 'msid' is incorrect Currently, the handling of the msid attribute is not quite right. According to the spec the msid's between the offer/answer are not dependent upon one another. Meaning the same msid's given in an offer do not have to be returned in the answer for a given stream. And they probably shouldn't be (copied/reused) since this can potentially cause some browser side confusion. This patch generates new msids when both an offer and answer are sent from Asterisk. However, Asterisk does reuse the original msid it sent out for a reinvite. Also audio+video streams are paired together by sharing the same stream id, but a different track id. ASTERISK-27179 #close Change-Id: Ifaec06dc7e65ad841633a24ebec8c8a9302d6643	9 years ago
Jenkins2	2ba29df200	Merge "alembic/res_pjsip: Add "webrtc" configuration option"	9 years ago
Jenkins2	7af10de1ba	Merge "res_pjsip_transport_websocket.c: Fix serializer ref leak."	9 years ago
Jenkins2	50d842b79a	Merge "res_pjsip_outbound_registration.c: Misc fixes."	9 years ago
Richard Mudgett	842e1414d0	res_pjsip_transport_websocket.c: Fix serializer ref leak. Change-Id: Ib5a19bfd597f63d9021baeb645fc11153b3afa57	9 years ago
Richard Mudgett	615b6a200a	res_pjsip_outbound_registration.c: Misc fixes. * Remove unnecessary CMP_STOP. * In handle_client_registration() use DEBUG_ATLEAST() to only do work needed for the debug log message when the debug log message is needed. * In sip_outbound_registration_state_destroy() check state->registration for NULL. Change-Id: I656d0fa11dda0b00048103efb1558e67a426fd80	9 years ago
Richard Mudgett	564927c5ed	res_pjsip_nat.c: Remove unnecessary CMP_STOP. Change-Id: I6279b0d723bc3b75b8d65e81e02da9ea9bc0c3da	9 years ago
Richard Mudgett	5655cded78	res_pjsip_registrar.c: Remove unnecessary CMP_STOP. Most uses of CMP_STOP are superfluous and are only respected when OBJ_MULTIPLE is used to search the container. Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8	9 years ago
Tzafrir Cohen	123c93a77c	Support GMIME 3.0 Support building the Asterisk httpd with version 3.0 of gmime as well as earlier versions of that library. ASTERISK-27173 Change-Id: I7e13dd05a3083ccb0df2dabf83110223f6a9fa8f	9 years ago
Kevin Harwell	521b6fed12	alembic/res_pjsip: Add "webrtc" configuration option When the "webrtc" option was added in res_pjsip it was not added to the alembic scripts. This patch adds the option for alembic. Also, changed the sorcery configuration type to an OPT_YESNO_T value instead of an OPT_BOOL_T so if this field is ever written to a database it will write out the correct value. ASTERISK-27119 #close Change-Id: I3e199f060aea25e193c439fc5cf96be4d3ed1c7b	9 years ago
Sean Bright	2be8d91c0f	res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation This change fixes PIDF content generation when the underlying device state is considered in use. Previously it was incorrectly marked as closed meaning they were offline/unavailable. The code now correctly marks them as open. Additionally: * Generate an XML element for our activity instead of a using a text node. * Consider every extension state other than "unavailable" to be 'open' status. * Update the XML namespaces and structure to reflect those documented in RFC 4480 * Use 'on-the-phone' (defined in RFC 4880) instead of 'busy' as the "in use" activity. This change results in eyeBeam using the appropriate icon for the watched user. This was tested on eyeBeam 1.5.20.2 build 59030 on Windows. ASTERISK-26659 #close Reported by: Abraham Liebsch patches: ASTERISK-26659.diff submitted by snuffy (license 5024) Change-Id: I6e5ad450f91106029fb30517b8c0ea0c2058c810	9 years ago
Joshua Colp	2a4283f3e7	res_pjsip: Add support for dnsmgr to external_media_address. The "external_media_address" option on transports is now resolved using dnsmgr. This allows it to be automatically refreshed regularly if refreshes are enabled in dnsmgr. If the system is using a dynamic IP address a dynamic DNS hostname can be provided to keep the IP address up to date. Change-Id: Ia54771720dff0105bde55d5bbb81a3ba437e05b2	9 years ago
Corey Farrell	58d032112b	Fix compiler warnings on Fedora 26 / GCC 7. GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88	9 years ago
Torrey Searle	65c560894d	chan_pjsip: add a new function PJSIP_DTMF_MODE This function is a replica of SIPDtmfMode, allowing the DTMF mode of a PJSIP call to be modified on a per-call basis ASTERISK-27085 #close Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612	9 years ago
Sean Bright	b3914df10b	res_rtp_asterisk: Fix mapping of pjsip's ICE roles to ours Change-Id: Ia578ede1a55b21014581793992a429441903278b	9 years ago
Joshua Colp	b610295b62	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues."	9 years ago
Joshua Colp	7ea6c66968	Merge "res_stasis_device_state: Unsubscribe should remove old subscriptions"	9 years ago
Joshua Colp	8412cc1e07	Merge "SDP: Rework SDP offer/answer model and update capabilities merges."	9 years ago
Sergej Kasumovic	4f4936fd72	res_stasis_device_state: Unsubscribe should remove old subscriptions Case scenario with Applications ARI: * Once you subscribe to deviceState with Applications REST API, it will be added into subscription pool. * When you unsubscribe it will remove from the device_state_subscription hash table but not from the subscription pool. * When you subscribe again, it will add it to pool again. * Now you will have two subscriptions and you will receive same event twice. This fix should now remove deviceState subscription from pool and it should fix unsubscribe on deviceState. ASTERISK-27130 #close Change-Id: I718b70d770a086e39b4ddba4f69a3c616d4476c4	9 years ago
Joshua Colp	f43fc91911	Merge "core: Add digit filtering to ast_waitfordigit_full"	9 years ago
Joshua Colp	680c491a62	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45	9 years ago
Jenkins2	647f539e15	Merge "res_pjsip: Add "webrtc" configuration option"	9 years ago
Jenkins2	29af7d5558	Merge "res_rtp_asterisk: Use RTP component for ICE if RTCP-MUX is in use."	9 years ago

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4180 Commits (60b44d1e38735d53dbfa4e346a9c3d8d8d925381)