asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	2634e63f4d	Merge "res_rtp_asterisk: Add frame list cleanups to ast_rtp_read" into 17	6 years ago
Friendly Automation	2725c885e8	Merge "sip_to_pjsip.py: Fix trustrpid typo" into 17	6 years ago
Joshua C. Colp	3104c03476	Merge "configure: Add check for MySQL client bool and my_bool type usage." into 17	6 years ago
Joshua C. Colp	58a6d4ecf5	Merge "res_pjsip_session: Set stream state on created streams for incoming SDP." into 17	6 years ago
Pascal Cadotte Michaud	69c9b74836	sip_to_pjsip.py: Fix trustrpid typo ASTERISK-28664 #close Change-Id: I6c28b1002fd7075ae0ed36f026f8c1855c9418a6	6 years ago
Joshua C. Colp	5a80c36d62	configure: Add check for MySQL client bool and my_bool type usage. Instead of trying to use the defined MySQL client version from the header use a configure check to determine whether the bool or my_bool type should be used for defining a boolean. ASTERISK-28604 Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7	6 years ago
Friendly Automation	c7966269f5	Merge "res_pjsip_nat: Restore original contact for REGISTER responses" into 17	6 years ago
Friendly Automation	a7a5bf0129	Merge "PJSIP_CONTACT: add missing argument documentation" into 17	6 years ago
Joshua Colp	895a66f772	Merge "ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging." into 17	6 years ago
Joshua C. Colp	c9147a759e	res_pjsip_session: Set stream state on created streams for incoming SDP. A previous review, 13174, made a change whereby on an incoming offer SDP the pending topology was initialized to the configured. This caused a problem for bundle with WebRTC where bundle could reference a stream that did not actually exist if the configuration had both audio and video but the offer SDP only contained audio. This change undoes that review and instead fixes the original problem it sought to solve by setting the state of created streams based on the contents of the offer SDP. This way the stream state is not inactive until negotiation later completes. ASTERISK-28659 Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355	6 years ago
Asterisk Development Team	08a6e8c553	Update CHANGES and UPGRADE.txt for 17.1.0	6 years ago
Jaco Kroon	6041fc8b9c	ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging. Due to use in res_rtp_asterisk there is a need to be able to apply an ACL without logging any invalid/denies. It's probably sensible to at least validate the ACL once directly after load and report invalid ACLs. Change-Id: I256169229d945ca7c1bbf228fc492d91df345843 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Pascal Cadotte Michaud	39aff53cff	PJSIP_CONTACT: add missing argument documentation add missing argument "rtt" and "status" to the documentation The change to the dtd file allow an enumlist to contain one or many configOptionToEnum or enum. This is different from the previous patch I submitted when you could have a configOptionToEnum or (a configOptionToEnum followed by one or manu enums) or (one or many enums) ASTERISK-28626 Change-Id: Ia71743ee7ec813f40297b0ddefeee7909db63b6d	6 years ago
George Joseph	87d5fb1196	Merge "Revert "PJSIP_CONTACT: add missing argument documentation"" into 17	6 years ago
Joshua Colp	6a89784b78	Revert "PJSIP_CONTACT: add missing argument documentation" This reverts commit `450173a0ae`. Reason for revert: Regression in XML validation. validity error : Content model of enumlist is not determinist: (configOptionToEnum \| (configOptionToEnum , enum+) \| enum+) As we are preparing to do releases and this is not critical I am reverting this for now until resolved. Change-Id: I47a6714bb555b35a371861d434daf5b97c0cdcbd	6 years ago
George Joseph	86822722b6	res_rtp_asterisk: Add frame list cleanups to ast_rtp_read In Asterisk 16+, there are a few places in ast_rtp_read where we've allocated a frame list but return a null frame instead of the list. In these cases, any frames left in the list won't be freed. In the vast majority of the cases, the list is empty when we return so there's nothing to free but there have been leaks reported in the wild that can be traced back to frames left in the list before returning. The escape paths now all have logic to free frames left in the list. ASTERISK-28609 Reported by: Ted G Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a	6 years ago
Friendly Automation	5fb0d1d562	Merge "res_pjsip_registrar.c: Prevent potential double free if AOR is not found" into 17	6 years ago
Friendly Automation	ffbfc09fc1	Merge "app_queue: Fix old confusing comment about when the members are called" into 17	6 years ago
George Joseph	fd14d7302f	res_pjsip_nat: Restore original contact for REGISTER responses RFC3261 Section 10 "Registrations", specifically paragraph "10.2.4: Refreshing Bindings", states that a user agent compares each contact address (in a 200 REGISTER response) to see if it created the contact. If the Asterisk endpoint has the rewrite_contact option set however, the contact host and port sent back in the 200 response will be the rewritten one and not the one sent by the user agent. This prevents the user agent from matching its own contact. Some user agents get very upset when this happens and will not consider the registration successful. While this is rare, it is acceptable behavior especially if more than 1 user agent is allowed to register to a single endpoint/aor. This commit updates res_pjsip_nat (where rewrite_contact is implemented) to store the original incoming Contact header in a new "x-ast-orig-host" URI parameter before rewriting it, and to restore the original host and port to the Contact headers in the outgoing response. This is only done if the request is a REGISTER and rewrite_contact is enabled. pjsip_message_filter was also updated to ensure that if a request comes in with any existing x-ast-* URI parameters, we remove them so they don't conflict. Asterisk will never send a request with those headers in it but someone might just decide to add them to a request they craft and send to Asterisk. NOTE: If a device changes its contact address and registers again, it's a NEW registration. If the device didn't unregister the original registration then all existing behavior based on aor/remove_existing and aor/max_contacts apply. ASTERISK-28502 Reported-by: Ross Beer Change-Id: Idc263ad2d2d7bd8faa047e5804d96a5fe1cd282e	6 years ago
George Joseph	2526c8e562	Merge "channel.c: Resolve issue with receiving SIP INFO packets for DTMF" into 17	6 years ago
Friendly Automation	af079f5085	Merge "res_pjsip_outbound_registration: add support for SRV failover" into 17	6 years ago
Friendly Automation	6685a1862c	Merge "res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases" into 17	6 years ago
Friendly Automation	e17bde758b	Merge "PJSIP_CONTACT: add missing argument documentation" into 17	6 years ago
Friendly Automation	aa2765d366	Merge "chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime." into 17	6 years ago
Friendly Automation	478a602e70	Merge "res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled" into 17	6 years ago
Friendly Automation	77b5ec3e7a	Merge "parking: Fall back to parker channel name even if it matches parkee." into 17	6 years ago
Sean Bright	4e057eb9d2	res_pjsip_registrar.c: Prevent potential double free if AOR is not found The simple fix here is simply to NULL out username and password after we call ast_free on them. Unfortunately, I noticed that we weren't checking for allocation failures for username and password, and adding those checks made things noisy and cumbersome. So instead we partially rollback the recent LGTM patch, and move the alloca calls into find_aor_name(). ASTERISK-28641 #close Reported by: Ross Beer Change-Id: Ic9d01624e717a020be0b0aee31f0814e7f1ffbe2	6 years ago
Sean Bright	f26e5bacc0	res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases We're appropriately sizing the id_domain_alias buffer, but then copying the data into the id_domain one. We were then using the uninitialized id_domain_alias buffer we just allocated. This is ASTERISK~28641 adjacent, but significant enough to warrant its own patch. Change-Id: I81c38724d18deab8c6573153e2b99dbb6e2f33d9	6 years ago
Walter Doekes	e1eb5e8dc2	app_queue: Fix old confusing comment about when the members are called ASTERISK-28644 Change-Id: I2771a931d00a8fc2b9f9a4d1a33ea8f1ad24e06b	6 years ago
Frederic LE FOLL	3e73893e53	chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime. During capabilities selection (joint capabilities of us and peer, configured capability for this peer, or general configured capabilities), if sip_new() does not keep framing information, then directmedia activation will fail for any framing different from default framing. ASTERISK-28637 Change-Id: I99257502788653c2816fc991cac7946453082466	6 years ago
Sean Bright	4d56adf8fb	res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled We need to copy the endpoint name before we call ao2_cleanup() on it, otherwise we might try to access memory that has been reclaimed. ASTERISK-28445 #close Reported by: Bernhard Schmidt Change-Id: I404b952608aa606e0babd3c4108346721fb726b3	6 years ago
George Joseph	c0ab241cc0	Merge "media_cache.c: Various CLI improvements" into 17	6 years ago
George Joseph	88150323a2	channel.c: Resolve issue with receiving SIP INFO packets for DTMF The problem is essentially the same as in ASTERISK~28245. Besides the direct media scenario we have an additional scenario where a special client is involved. This device mutes audio by default in transmit direction (no rtp frames) and activates audio only by a foot switch. In this situation dtmf input (pin for conferences, transfer features codes , etc) using SIP INFO mode is not understood properly especially when SIP INFO messages are sent quickly. This patch ensures that SIP INFO frames are properly queued and processed in the above scenario. The patch also corrects situations where successive dtmf events are received quicker than the signalled event duration (plus minimum gap/pause) allows, i.e. DTMF events have to be buffered in the ast channel read queue and emulation has to be processed asynchronously at slower speed. Reported by: Thomas Arimont patches: trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525) Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2	6 years ago
George Joseph	5b15a1c639	CI: Turn off shallow cloning altogether Change-Id: I73ed4aef33a92f20080128aafc34e19fd4457196	6 years ago
Joshua Colp	41d58a4ce2	parking: Fall back to parker channel name even if it matches parkee. ASTERISK-28631 Change-Id: Ia74d084799fbb9bee3403e30d2391aacd46243cc	6 years ago
Sean Bright	cc59e21409	media_cache.c: Various CLI improvements * Use ast_cli_completion_add() to improve performance when large number of cached items are present. * Only complete one URI for commands that only accept a single URI. * Change command documentation to wrap at 80 characters to improve readability. Change-Id: Iedb0a2c3541e49561bc231dca2dcc0ebd8612902	6 years ago
Salah Ahmed	4ac0299bfb	res_pjsip_t38: T.38 error correction mode selection at 200 ok received if asterisk offer T38 SDP with none error correction scheme and the endpoint respond with redundancy EC scheme, asterisk switch to that mode. Since we configure the endpoint as none EC mode we should not switch to any other mode except none. following logic implemented in code. 1. If asterisk offer none, and anything except none in answer will be ignored. 2. If asterisk offer fec, answer with fec, redundancy and none will be accepted. 3. If asterisk offer redundancy, answer with redundancy and none will be accepted. ASTERISK-28621 Change-Id: I343c62253ea4c8b7ee17abbfb377a4d484a14b19	6 years ago
Friendly Automation	e9b9141d09	Merge "manager.c: Prevent the Originate action from running the Originate app" into 17	6 years ago
Ben Ford	665a94cb76	chan_sip.c: Prevent address change on unauthenticated SIP request. If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option. ASTERISK-28589 #close Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df	6 years ago
George Joseph	6b1ba58967	manager.c: Prevent the Originate action from running the Originate app If an AMI user without the "system" authorization calls the Originate AMI command with the Originate application, the second Originate could run the "System" command. Action: Originate Channel: Local/1111 Application: Originate Data: Local/2222,app,System,touch /tmp/owned If the "system" authorization isn't set, we now block the Originate app as well as the System, Exec, etc. apps. ASTERISK-28580 Reported by: Eliel Sardañons Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa	6 years ago
George Joseph	6aa9ee1661	Merge "chan_dahdi: PRI span status may stay "Down, Active" after a short alarm" into 17	6 years ago
Friendly Automation	715e4d0e72	Merge "app_senddtmf: Add receive mode to AMI Action PlayDTMF" into 17	6 years ago
Pascal Cadotte Michaud	450173a0ae	PJSIP_CONTACT: add missing argument documentation add missing argument "rtt" and "status" to the documentation ASTERISK-28626 Change-Id: I8419e4c8203e411b87d93dc395acdbcf7526dedf	6 years ago
Kevin Harwell	b1c9176b3f	Merge "app_amd: Fixed timeout issue" into 17	6 years ago
Kevin Harwell	8c99930375	res_pjsip_outbound_registration: add support for SRV failover ASTERISK-28624 Change-Id: I8da7c300dd985ab7b10dbd5194aff2f737808561	6 years ago
Friendly Automation	ecdc5405f9	Merge "func_curl.c: Support custom http headers" into 17	6 years ago
George Joseph	8a02454ec6	Merge "res_pjsip_registrar: Fix uninitlized variable warning" into 17	6 years ago
George Joseph	2a92e6b576	CI: Fix missing script block in jenkinsfiles Change-Id: I9f44a3d5085ea7880fad1a3883a4820907e29ea3 (cherry picked from commit `95213b01d2`)	6 years ago
George Joseph	f0d1ce50af	CI: Fix missing script block in jenkinsfiles Change-Id: Ib4b6e4887695f230ea7a5b0c879b29fc5a13be4f (cherry picked from commit `d60f23ecbd`) (cherry picked from commit `ce8a23fdf9`)	6 years ago
George Joseph	46dceab33f	CI: Increase clone depth and do better cleanup The original clone depth of 10 was causing the need to rebase changes whose parent was older than the 10 commits. The clone depth has been increased to 100. Workspace cleanup was only happening for successful builds which wasn't enough to keep the 8G workspace in-memory drives on the docker slaves from filling up. Now the workspaces are cleaned up after every build regardless of success/failure. If you need to preserve builds temporarily, you can log into Jenkins/Manage Jenkins/Configure System and change the CLEANUP_WS_* environment variable for the job type you're troubleshooting to "FALSE". Change-Id: I0d7366e87cea714e5dbc9488caf718802fce75ca	6 years ago

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