252 Commits (de5144e751ae9d4dd28515768cb15ade61063965)

Author	SHA1	Message	Date
Joshua Colp	de5144e751	pjsip: Clarify certificate configuration for Websocket. ... The Websocket transport uses the built-in HTTP server. As a result the TLS configuration is done in http.conf and not in pjsip.conf. This change adds a warning if this is configured in pjsip.conf and also clarifies in the sample configuration file. Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9	7 years ago
George Joseph	880fbff6b7	res_pjsip_session: Add ability to accept multiple sdp answers ... pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer	7 years ago
Joshua Colp	882e79b77e	pjsip: Rewrite OPTIONS support with new eyes. ... The OPTIONS support in PJSIP has organically grown, like many things in Asterisk. It has been tweaked, changed, and adapted based on situations run into. Unfortunately this has taken its toll. Configuration file based objects have poor performance and even dynamic ones aren't that great. This change scraps the existing code and starts fresh with new eyes. It leverages all of the APIs made available such as sorcery observers and serializers to provide a better implementation. 1. The state of contacts, AORs, and endpoints relevant to the qualify process is maintained. This state can be updated by external forces (such as a device registering/unregistering) and also the reload process. This state also includes the association between endpoints and AORs. 2. AORs are scheduled and not contacts. This reduces the amount of work spent juggling scheduled items. 3. Manipulation of which AORs are being qualified and the endpoint states all occur within a serializer to reduce the conflict that can occur with multiple threads attempting to modify things. 4. Operations regarding an AOR use a serializer specific to that AOR. 5. AORs and endpoint state act as state compositors. They take input from lower level objects (contacts feed AORs, AORs feed endpoint state) and determine if a sufficient enough change has occurred to be fed further up the chain. 6. Realtime is supported by using observers to know when a contact has been registered. If state does not exist for the associated AOR then it is retrieved and becomes active as appropriate. The end result of all of this is best shown with a configuration file of 3000 endpoints each with an AOR that has a static contact. In the old code it would take over a minute to load and use all 8 of my cores. This new code takes 2-3 seconds and barely touches the CPU even while dealing with all of the OPTIONS requests. ASTERISK-26806 Change-Id: I6a5ebbfca9001dfe933eaeac4d3babd8d2e6f082	8 years ago
Richard Mudgett	237d341bbd	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
Nathan Bruning	1cd704de36	res_pjsip_notify.c: enable in-dialog NOTIFY ... This patch adds support to send in-dialog SIP NOTIFY commands on chan_pjsip channels, similar to the functionality recently added for chan_sip (ASTERISK_27461). This extends res_pjsip_notify to allow for in-dialog messages. ASTERISK-27697 Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29	8 years ago
Richard Mudgett	71a67a98c4	res_pjsip: Update authenticate_qualify documentation. ... Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4	8 years ago
Corey Farrell	572a508ef2	loader: Convert reload_classes to built-in modules. ... * acl (named_acl.c) * cdr * cel * ccss * dnsmgr * dsp * enum * extconfig (config.c) * features * http * indications * logger * manager * plc * sounds * udptl These modules are now loaded at appropriate time by the module loader. Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so the module loader will abort startup on failure of these modules. Some of these modules are still initialized or shutdown from outside the module loader. logger.c is initialized very early and shutdown very late, manager.c is initialized by the module loader but is shutdown by the Asterisk core (too much uses it without holding references). Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f	8 years ago
Richard Mudgett	1a36a452bd	pjproject: Add cache_pools debugging option. ... The pool cache gets in the way of finding use after free errors of memory pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a pool is released because it gets put into the cache instead of being freed. * Added the "cache_pools" option to pjproject.conf. Disabling the option helps track down pool content mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the pool contents are used after free and who freed it. To disable the pool caching simply disable the cache_pools option in pjproject.conf and restart Asterisk. Sample pjproject.conf setting: [startup] cache_pools=no * Made current users of the caching pool factory initialization and destruction calls call common routines to create and destroy cached pools. ASTERISK-27704 Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828	8 years ago
George Joseph	758409de56	AST-2018-005: res_pjsip_transport_management: Move to core ... Since res_pjsip_transport_management provides several attack mitigation features, its functionality moved to res_pjsip and this module has been removed. This way the features will always be available if res_pjsip is loaded. ASTERISK-27618 Reported By: Sandro Gauci Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d	8 years ago
George Joseph	de871515ba	AST-2018-005: Fix tdata leaks when calling pjsip_endpt_send_response(2) ... pjsip_distributor: authenticate() creates a tdata and uses it to send a challenge or failure response. When pjsip_endpt_send_response2() succeeds, it automatically decrements the tdata ref count but when it fails, it doesn't. Since we weren't checking for a return status, we weren't decrementing the count ourselves on error and were therefore leaking tdatas. res_pjsip_session: session_reinvite_on_rx_request wasn't decrementing the ref count if an error happened while sending a 491 response. pre_session_setup wasn't decrementing the ref count if while sending an error after a pjsip_inv_verify_request failure. res_pjsip: ast_sip_send_response wasn't decrementing the ref count on error. ASTERISK-27618 Reported By: Sandro Gauci Change-Id: Iab33a6c7b6fba96148ed465b690ba8534ac961bf	8 years ago
George Joseph	c53d8dcb68	AST-2018-005: Add a check for NULL tdata in ast_sip_failover_request ... It was discovered that there are some corner cases where a pjsip tsx might have no last_tx so calling ast_sip_failover_request with a NULL last_tx as its tdata would cause a crash. ASTERISK-27618 Reported By: Sandro Gauci Change-Id: Ic2b63f6d4ae617c4c19dcdec2a7a6156b54fd15b	8 years ago
Sean Bright	303e43f8a6	res_pjsip: Use pjsip_sip_uri.user_param instead of other_param ... There is a dedicated slot in the pjsip_sip_uri for the 'user' parameter, so use that instead of adding to the list of generic URI parameters. Change-Id: I0a0ce8a60ecee27489735bf56fd707719d8c2ed6	8 years ago
Richard Mudgett	0960de71ae	res_pjsip.c: Fix documentation typos. ... Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068	8 years ago
George Joseph	2b9aa6b5bb	res_pjsip_pubsub: Prune subs with reliable transports at startup ... In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	8 years ago
Jenkins2	a231e1d155	Merge "Remove redundant module checks and references."	8 years ago
Jenkins2	da5859246d	Merge "Add missing OPTIONAL_API and ARI dependences."	8 years ago
Corey Farrell	527cf5a570	Remove redundant module checks and references. ... This removes references that are no longer needed due to automatic references created by module dependencies. In addition this removes most calls to ast_module_check as they were checking modules which are listed as dependencies. Change-Id: I332a6e8383d4c72c8e89d988a184ab8320c4872e	8 years ago
Jenkins2	7ce34f4e6a	Merge "res_pjsip: Add AMI action 'PJSIPShowContacts'"	8 years ago
Sungtae Kim	5b8e71ab9f	res_pjsip: Add AMI action 'PJSIPShowContacts' ... Add an AMI action which provides information on all configured Contacts. ASTERISK-27581 Change-Id: I2eed42c74bbc725fad26b8b33b1a5b3161950c73	8 years ago
Corey Farrell	679fa5fb34	Add missing OPTIONAL_API and ARI dependences. ... I've audited all modules that include any header which includes asterisk/optional_api.h. All modules which use OPTIONAL_API now declare those dependencies in AST_MODULE_INFO using requires or optional_modules as appropriate. In addition ARI dependency declarations have been reworked. Instead of declaring additional required modules in res/ari/resource_*.c we now add them to an optional array "requiresModules" in api-docs for each module. This allows the AST_MODULE_INFO dependencies to include those missing modules. Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606	8 years ago
Sean Bright	5964061a21	res_pjsip: Document tlsv1_1 and tlsv1_2 methods ... Change-Id: I67ed9039bf3f132fb20ee7a750e0aef0f704d7d3	8 years ago
Richard Mudgett	8494e78010	res_pjsip: Split type=identify to IP address and SIP header matching priorities ... The type=identify endpoint identification method can match by IP address and by SIP header. However, the SIP header matching has limited usefulness because you cannot specify the SIP header matching priority relative to the IP address matching. All the matching happens at the same priority and the order of evaluating the identify sections is indeterminate. e.g., If you had two type=identify sections where one matches by IP address for endpoint alice and the other matches by SIP header for endpoint bob then you couldn't predict which endpoint is matched when a request comes in that matches both. * Extract the SIP header matching criteria into its own "header" endpoint identification method so the user can specify the relative priority of the SIP header and the IP address matching criteria in the global endpoint_identifier_order option. The "ip" endpoint identification method now only matches by IP address. ASTERISK-27491 Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095	8 years ago
Corey Farrell	9cfdb81e91	loader: Add dependency fields to module structures. ... * Declare 'requires' and 'enhances' text fields on module info structure. * Rename 'nonoptreq' to 'optional_modules'. * Update doxygen comments. Still need to investigate dependencies among modules I cannot compile. Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf	8 years ago
Jenkins2	97a0e8591f	Merge "res_pjsip.c: Update the endpoint identification documentation."	8 years ago
Joshua Colp	3e8b951509	Merge "res_pjsip: Add AMI action 'PJSIPShowAuths'"	8 years ago
Richard Mudgett	8f3167c5f1	res_pjsip.c: Update the endpoint identification documentation. ... * Endpoint identify_by documentation. * IP/Header endpoint identifier documentation. Change-Id: Id92f00b495acca7be945daf749d2abd7f76a0b5a	8 years ago
Sungtae Kim	faeb9e1b26	res_pjsip: Add AMI action 'PJSIPShowAuths' ... Add an AMI action which provides information on all configured Auths. ASTERISK-27547 Change-Id: I1a88a75b38a2b1dd9d1de6c0307b20a3f584c817	8 years ago
Richard Mudgett	705e6c04b3	res_pjsip.c: Fix endpoint identifier registration name search. ... If an endpoint identifier name in the endpoint_identifier_order list is a prefix to the identifier we are registering, we could install it in the wrong position of the list. Assuming endpoint_identifier_order=username,ip,anonymous then registering the "ip_only" identifier would put the identifier in the wrong position of the priority list. * Fix incorrect strncmp() string prefix matching. Change-Id: Ib8819ec4b811da8a27419fd93528c54d34f01484	8 years ago
Sungtae Kim	ffbf5be116	res_pjsip: Add AMI action 'PJSIPShowAors' ... Add an AMI action which provides information on all configured AORs. ASTERISK-27537 Change-Id: If8b990a00909e5b6c0f04a3b8dccd9903dc445eb	8 years ago
Kevin Harwell	553306548c	AST-2017-014: res_pjsip - Missing contact header can cause crash ... Those SIP messages that create dialogs require a contact header to be present. If the contact header was missing from the message it could cause Asterisk to crash. This patch checks to make sure SIP messages that create a dialog contain the contact header. If the message does not and it is required Asterisk now returns a "400 Missing Contact header" response. Also added NULL checks when retrieving the contact header that were missing as a "just in case". ASTERISK-27480 #close Change-Id: I1810db87683fc637a9e3e1384a746037fec20afe	8 years ago
Kevin Harwell	b088cddc03	pjsip_options: wrongly applied "UNKNOWN" status ... A couple of places were setting the status to "UNKNOWN" when qualifies were being disabled. Instead this should be set to the "CREATED" status that represents when a contact is given (uri available), but the qualify frequency is set to zero so we don't know the status. This patch updates the relevant places with "CREATED". It also updates the "CREATED" status description (value shown in CLI/AMI/ARI output) to a value of "NonQualified"/"NonQual" as this description is hopefully less confusing. ASTERISK-27467 Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89	8 years ago
Joshua Colp	637b37fb98	Merge "dtls: Add support for ephemeral DTLS certificates."	8 years ago
Sean Bright	04d3785a79	dtls: Add support for ephemeral DTLS certificates. ... This mimics the behavior of Chrome and Firefox and creates an ephemeral X.509 certificate for each DTLS session. Currently, the only supported key type is ECDSA because of its faster generation time, but other key types can be added in the future as necessary. ASTERISK-27395 Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4	8 years ago
Corey Farrell	606ae3484a	Add missing menuselect dependencies. ... This adds menuselect dependencies for modules that use symbols of other modules. ASTERISK-27390 Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385	8 years ago
Joshua Colp	9e1fbab382	res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint. ... When the identify_by option on an endpoint is set to ip it will only be identified using the res_pjsip_endpoint_identifier_ip module. This ensures that it is not mistakenly matched using the username of the From header. To ensure behavior has not changed the default has been changed to "username,ip" for the identify_by option. ASTERISK-27206 Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd	8 years ago
Daniel Tryba	a56316423f	res_pjsip: Prevent "user=phone" being added multiple times to header ... ast_sip_add_usereqphone adds "user=phone" to the header every time is is called without checking whether the param already exists. Preventing this by searching to string representation of header for "user=phone". ASTERISK-26988 #close Change-Id: Ib84383b07254de357dc6a98d91fc1d2c2c3719e6	8 years ago
Jenkins2	1505c1bb09	Merge "res_pjsip_registrar.c: Update remove_existing AOR contact handling."	8 years ago
Richard Mudgett	fb19799b62	res_pjsip_registrar.c: Update remove_existing AOR contact handling. ... When "rewrite_contact" is enabled, the "max_contacts" count option can block re-registrations because the source port from the endpoint can be random. When the re-registration is blocked, the endpoint may give up re-registering and require manual intervention. * The "remove_existing" option now allows a registration to succeed by displacing any existing contacts that now exceed the "max_contacts" count. Any removed contacts are the next to expire. The behaviour change is beneficial when "rewrite_contact" is enabled and "max_contacts" is greater than one. The removed contact is likely the old contact created by "rewrite_contact" that the device is refreshing. ASTERISK-27192 Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b	8 years ago
Corey Farrell	65399a5eda	res_pjsip: Add REF_DEBUG info to module references. ... This provides better information to REF_DEBUG log for troubleshooting when the system is unable to unload res_pjsip.so during shutdown due to module references. ASTERISK-27306 Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612	8 years ago
Corey Farrell	7d04544986	res_pjsip: Fix issues that prevented shutdown of modules. ... res_pjsip and res_pjsip_session had circular references, preventing both modules from shutting down. * Move session supplement registration to res_pjsip. * Use create internal functions for use by pjsip_message_filter.c. ASTERISK-27306 Change-Id: Ifbd5c19ec848010111afeab2436f9699da06ba6b	8 years ago
George Joseph	d178f497d2	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	8 years ago
George Joseph	446d48fd49	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	8 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	8 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	8 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	8 years ago
Kevin Harwell	7da6ddda30	res_pjsip: Add "webrtc" configuration option ... This patch creates a new configuration option called "webrtc". When enabled it defaults and enables the following options that are needed in order for webrtc to work in Asterisk: rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled media_encryption=dtls dtls_verify=fingerprint dtls_setup=actpass When "webrtc" is enabled, this patch also parses the "msid" media level attribute from an SDP. It will also appropriately add it onto the outgoing session when applicable. Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. ASTERISK-27119 #close Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd	8 years ago
Jenkins2	0f45c979a3	Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."	8 years ago
Joshua Colp	065c3005ad	res_rtp_asterisk / res_pjsip: Add support for BUNDLE. ... BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11	8 years ago
Benjamin Keith Ford	8f72128e66	res_pjsip: Fix crash with from_user containing invalid characters. ... If the from_user field contains certain characters (like @, {, ^, etc.), PJSIP will return a null value for the URI when attempting to parse it. This causes a crash when trying to dial out through a trunk that contains these invalid characters in its from_user field. This change checks the configuration and ensures that an endpoint will not be created if the from_user contains an invalid character. It also adds a null check to the PJSIP URI parsing as a backup. ASTERISK-27036 #close Reported by: Maxim Vasilev Change-Id: I0396fdb5080604e0bdf1277464d5c8a85db913d0	8 years ago
Jenkins2	d2b32cd009	Merge "chan_pjsip: Fix ability to send UPDATE on COLP"	8 years ago