3430 Commits (9de5cd209e5918d27d03afaaddb41125d7a76701)

Author	SHA1	Message	Date
Sean Bright	9de5cd209e	res_ari: Correct Location headers returned by some ARI resources ... The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c	10 years ago
zuul	e6a946400f	Merge "res_hep: Provide an option to pick the UUID type" into 13	10 years ago
zuul	c735ce1a05	Merge "config_transport: Tell pjproject to allow all SSL/TLS protocols" into 13	10 years ago
Matt Jordan	89ae4466ea	res_hep: Provide an option to pick the UUID type ... At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c	10 years ago
zuul	1705c5d2ba	Merge "pjsip_distributor: Add missing newline to NOTICE" into 13	10 years ago
George Joseph	e2df15bae9	pjsip_distributor: Add missing newline to NOTICE ... There was a newline missing from the end of the "no matching endpoint" notice. Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181	10 years ago
Sebastian Damm	a94a12bbf7	res_pjsip_outbound_registration: generate correct Contact URI for TLS ... There are two types of SIP URIs indicating a secure transport: * sips:user@example.org * sip:user@example.org;transport=tls When using a sips URI, Asterisk checks incoming INVITEs and answers from the other side for sips URIs, and rejects the packet if there are only sip URIs. So Asterisk should only generate a sips Contact URI if the other side supports it. This patch makes Asterisk generate either a sip or sips Contact URI depending on the format of the server URI. If you want a sip URI, use: server_uri=sip:example.org\;transport=tls If you want a sips URI, use: server_uri=sips:example.org ASTERISK-25990 #close Reported-by: Sebastian Damm Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2	10 years ago
zuul	a01ce2b889	Merge "res_pjsip: improve realtime performance" into 13	10 years ago
zuul	81773ceb9c	Merge "res_fax/t38_gateway: Peer V.21 session is created on wrong channel" into 13	10 years ago
Kevin Harwell	4d063814ba	res_pjsip_authenticator_digest: Don't use source port in nonce verification ... From the issue reporter: "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of the timestamp, the source address, the source port, a server UUID that is calculated at startup, and the authentication realm. Rather than caching nonces that we create, we instead attempt to re-calculate the nonce when receiving an incoming request with authentication. We then compare the re-calculated nonce to the incoming nonce, and if they don't match, then authentication has failed early. The problem is that it is possible, especially when using TCP, to receive two requests from the same endpoint but have differing source ports for those requests. Asterisk itself commonly will use different source ports for outbound TCP requests." This patch removes the source port dependency when building the nonce. ASTERISK-25978 #close Change-Id: I871b5f4adce102df1c4988066283095ec509dffe	10 years ago
George Joseph	fb6227a372	config_transport: Tell pjproject to allow all SSL/TLS protocols ... The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2. SSL is not allowed. So, even if you specify "sslv3" for a transport method, it's silently ignored and one of the TLS protocols is used. This was a new behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that we never caught. Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default(). This tells pjproject to set the socket protocol to match the method. ASTERISK-26004 #close Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078	10 years ago
Alexei Gradinari	72eb7c8301	res_pjsip: module load priority ... The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_* and res_pjsip_registrar modules should load ASAP to avoid "No matching endpoint found" for legitimate endpoint. ASTERISK-25994 Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b	10 years ago
Alexei Gradinari	9c2032240e	res_pjsip: improve realtime performance ... This patch modified pjsip_options to retrieve only permament contacts for aor if the qualify_frequency is > 0 and persisted contacts if the qualify_frequency is > 0. This patch also fixed a bug in res_sorcery_astdb. res_sorcery_astdb doesn't save object data retrived from astdb. ASTERISK-25826 Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05	10 years ago
zuul	168a7b3dd8	Merge "res_fax: add FAXMODE variable" into 13	10 years ago
Alexei Gradinari	7a14e669f0	res_pjsip/AMI: add contact.updated event ... With the old SIP module AMI sends PeerStatus event on every successfully REGISTER requests, ie, on start registration, update registration and stop registration. With PJSIP AMI sends ContactStatus only when status is changed. Regarding registration: on start registration - Created on stop registration - Removed but on update registration nothing This patch added contact.updated event. ASTERISK-25904 Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f	10 years ago
Alexei Gradinari	06d4ac0355	res_fax: add FAXMODE variable ... The app_fax set FAXMODE variable, but res_fax missing this feature. This patch add FAXMODE variable which is set to either "audio" or "T38". ASTERISK-25980 Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b	10 years ago
Alexei Gradinari	2d17fe06c5	res_fax/t38_gateway: Peer V.21 session is created on wrong channel ... The channel and peer V.21 sessions are created on the same channel now. The peer V.21 session should be created only on peer channel when one of channel can handle T.38. Also this patch enable debug for T.38 gateway session if global fax debug enabled. ASTERISK-25982 Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e	10 years ago
Alexei Gradinari	3cb8934de0	pjsip: Added "reg_server" to contacts. ... If the Asterisk system name is set in asterisk.conf, it will be stored into the "reg_server" field in the ps_contacts table to facilitate multi-server setups. ASTERISK-25931 Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8	10 years ago
Joshua Colp	d65023b5a5	Merge "res_pjsip: Start body generator users after suppliers." into 13	10 years ago
zuul	3e5666eadc	Merge "res_pjsip_pubsub.c: Fix body generator registration race." into 13	10 years ago
Joshua Colp	1ce30f1fb5	Merge "res_pjsip_outbound_publish.c: Remove redundant flag check." into 13	10 years ago
zuul	cc8a50631e	Merge "res_pjsip_pubsub.c: Add useful information to some messages." into 13	10 years ago
Richard Mudgett	7992923c70	res_pjsip: Start body generator users after suppliers. ... Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb	10 years ago
Richard Mudgett	5dc0e082b2	res_pjsip_pubsub.c: Add useful information to some messages. ... Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a	10 years ago
Richard Mudgett	f9e416f053	res_pjsip_pubsub.c: Fix body generator registration race. ... Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67	10 years ago
Richard Mudgett	b7f07fdff5	res_pjsip_outbound_publish.c: Remove redundant flag check. ... Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353	10 years ago
George Joseph	38bed4515d	res_pjsip: Add ability to identify by Authorization username ... A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d	10 years ago
Joshua Colp	e0e03cd2c8	Merge "res_pjsip: disable multi domain to improve realtime performace" into 13	10 years ago
Alexei Gradinari	df3639700a	res_pjsip: disable multi domain to improve realtime performace ... This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	10 years ago
Joshua Colp	e5c5fac8f1	Merge "res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c)" into 13	10 years ago
zuul	a8f8e3c340	Merge "res_agi: Prevent run_agi from eating frames it shouldn't" into 13	10 years ago
George Joseph	eb7c581806	res_agi: Prevent run_agi from eating frames it shouldn't ... The run_agi function is eating control frames when it shouldn't be. This is causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond transfer. Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie answers. Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE and is left thinking he's connected to Bob. In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on Charlie's channel. The fix was to accumulate deferrable frames in the "forever" loop instead of dropping them, and re-queue them just before running the actual agi command or exiting. ASTERISK-25951 #close Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645	10 years ago
zuul	77ae5cd7fd	Merge "res_stasis: Handle re-enter stasis bridge with swap channel." into 13	10 years ago
Joshua Colp	6112a94d03	Merge "res_pjsip_callerid: Clear out display name if id->name is not valid" into 13	10 years ago
Richard Mudgett	a036c35903	res_stasis: Handle re-enter stasis bridge with swap channel. ... We lose the fact that there is a swap channel if there is one. We currently wind up rejoining the stasis bridge as a normal join after the swap channel has already been kicked from the bridge. This patch preserves the swap channel so the AMI/ARI events can note that the channel joining the bridge is swapping with another channel. Another benefit to swaqpping in one operation is if there are any channels that get lonely (MOH, bridge playback, and bridge record channels). The lonely channels won't leave before the joining channel has a chance to come back in under stasis if the swap channel is the only reason the lonely channels are staying in the bridge. ASTERISK-25947 #close Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee	10 years ago
George Joseph	516c626a7d	res_pjsip_callerid: Clear out display name if id->name is not valid ... When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning the From header, then it overwrites the display name and uri from the channel's connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was leaving the display name from the From header in the new RPID or PAI header. On an attended transfer where the originator had a caller id number set but not a display name, the re-INVITE to the final transferee had the number of the originator but the display name of the transferer. Added a check to clear out the display name in the new header if connected.id.name was invalid. ASTERISK-25942 #close Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b	10 years ago
Mark Michelson	efae187217	PJSIP: Remove PJSIP parsing functions from uri length validation. ... The PJSIP parsing functions provide a nice concise way to check the length of a hostname in a SIP URI. The problem is that in order to use those parsing functions, it's required to use them from a thread that has registered with PJLib. On startup, when parsing AOR configuration, the permanent URI handler may not be run from a PJLib-registered thread. Specifically, this could happen when Asterisk was started in daemon mode rather than console-mode. If PJProject were compiled with assertions enabled, then this would cause Asterisk to crash on startup. The solution presented here is to do our own parsing of the contact URI in order to ensure that the hostname in the URI is not too long. The parsing does not attempt to perform a full SIP URI parse/validation, since the hostname in the URI is what is important. ASTERISK-25928 #close Reported by Joshua Colp Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60	10 years ago
Joshua Colp	47adff8977	Merge "stasis_bridge.c: Update stasis bridge push diagnostic messages." into 13	10 years ago
Joshua Colp	a048a0ffbd	Merge "res_pjsip_transport_management: Allow unload to occur." into 13	10 years ago
Mark Michelson	f436b9ab11	res_pjsip_registrar: Fix bad memory-ness with user_agent. ... Recent changes to the PJSIP registrar resulted in tests failing due to missing AOR_CONTACT_ADDED test events. The reason for this was that the user_agent string had junk values in it, resulting in being unable to generate the event. I'm going to be honest here, I have no idea why this was happening. Here are the steps needed for the user_agent variable to get messed up: * REGISTER is received * First contact in the REGISTER results in a contact being removed * Second contact in the REGISTER results in a contact being added * The contact, AOR, expiration, and user agent all have to be passed as format parameters to the creation of a string. Any subset of those parameters would not be enough to cause the problem. Looking into what was happening, the thing that struck me as odd was that the user_agent variable was meant to be set to the value of the User-Agent SIP header in the incoming REGISTER. However, when removing a contact, the user_agent variable would be set (via ast_strdupa inside a loop) to the stored contact's user_agent. This means that the user_agent's value would be incorrect when attempting to process further contacts in the incoming REGISTER. The fix here is to use a different variable for the stored user agent when removing a contact. Correcting the behavior to be correct also means the memory usage is less weird, and the issue no longer occurs. ASTERISK-25929 #close Reported by Joshua Colp Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08	10 years ago
Joshua Colp	49bfdc9ac0	res_pjsip_transport_management: Allow unload to occur. ... At shutdown it is possible for modules to be unloaded that wouldn't normally be unloaded. This allows the environment to be cleaned up. The res_pjsip_transport_management module did not have the unload logic in it to clean itself up causing the res_pjsip module to not get unloaded. As a result the res_pjsip monitor thread kept going processing traffic and timers when it shouldn't. Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a	10 years ago
Richard Mudgett	724acb6ce7	stasis_bridge.c: Update stasis bridge push diagnostic messages. ... Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a	10 years ago
Mark Michelson	5f78801859	transport management: Register thread with PJProject. ... The scheduler thread that kills idle TCP connections was not registering with PJProject properly and causing assertions if PJProject was built in debug mode. This change registers the thread with PJProject the first time that the scheduler callback executes. AST-2016-005 Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283	10 years ago
George Joseph	9740277713	res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) ... There are several places that do scheduled tasks or periodic housecleaning, each with its own implementation: * res_pjsip_keepalive has a thread that sends keepalives. * pjsip_distributor has a thread that cleans up expired unidentified requests. * res_pjsip_registrar_expire has a thread that cleans up expired contacts. * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. There are also places where we should be doing scheduled work but aren't. A good example are the places we have sorcery observers to start registration or qualify. These don't work when changes are made to a backend database without a pjsip reload. We need to check periodically. As a first step to solving these issues, a new ast_sip_sched facility has been created. ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. When a task is ready to run, ast_sip_task_pusk is called for it. This ensures that the task is executed in a PJLIB registered thread and doesn't hold up the ast_sched thread so it can immediately continue processing the queue. The serializer used by ast_sip_sched is one of your choosing or a random one from the res_pjsip pool if you don't choose one. Another feature is the ability to automatically clean up the task_data when the task expires (if ever). If it's an ao2 object, it will be dereferenced, if it's a malloc'd object it will be freed. This is selectable when the task is scheduled. Even if you choose to not auto dereference an ao2 task data object, the scheduler itself maintains a reference to it while the task is under it's control. This prevents the data from disappearing out from under the task. There are two scheduling models. AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the specific interval. That is, every "interval" milliseconds, regardless of how long the task takes. If the task takes longer than the interval, it will be scheduled at the next available multiple of interval. For exmaple: If the task has an interval of 60 secs and the task takes 70 secs (it better not), the next invocation will happen at 120 seconds. AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start "interval" milliseconds after the current invocation has finished. Also, the same ast_sched facility for fixed or variable intervals exists. The task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. One res_pjsip.h housekeeping change was made. The pjsip header files were added to the top. There have been a few cases lately where I've needed res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because I didn't add the pjsip header files to my source even though I never referenced any pjsip calls. Finally, a few new convenience APIs were added to astobj2 to make things a little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros were also copied from res_phoneprov because I got tired of having to duplicate the same hash, sort and compare functions over and over again. The AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for aor_container_alloc into your source. This facility can be used immediately for the situations where we already have a thread that wakes up periodically or do some scheduled work. For the registration and qualify issues, additional sorcery and schema changes would need to be made so that we can easily detect changed objects on a periodic basis without having to pull the entire database back to check. I'm thinking of a last-updated timestamp on the rows but more on this later. Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c	10 years ago
Joshua Colp	13cb5ea73f	Merge "res_pjsip_transport_management: Kill idle TCP connections." into 13	10 years ago
Joshua Colp	120493d5c0	Merge "Rename res_pjsip_keepalive res_pjsip_transport_management" into 13	10 years ago
Mark Michelson	7fb3724a77	res_pjsip_transport_management: Kill idle TCP connections. ... "Idle" here means that someone connects to us and does not send a SIP request. PJProject will not automatically time out such connections, so it's up to Asterisk to do it instead. When we receive an incoming TCP connection, we will start a timer (equivalent to transaction timer D) waiting to receive an incoming request. If we do not receive a request in that timeframe, then we will shut down the TCP connection. ASTERISK-25796 #close Reported by George Joseph AST-2016-005 Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6	10 years ago
Mark Michelson	707fd4dcd0	Rename res_pjsip_keepalive res_pjsip_transport_management ... ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433	10 years ago
Mark Michelson	0b4bb19e0b	AST-2016-004: Fix crash on REGISTER with long URI. ... Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55	10 years ago
Joshua Colp	4ab4fc9141	Merge "res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event" into 13	10 years ago