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120 Commits (a66fa4db24553d6ec6c8978c528081a94b1715a1)
Author | SHA1 | Message | Date |
---|---|---|---|
|
3470409dd6 |
res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint
A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1 |
7 years ago |
|
06966e91fe |
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 |
|
bea52b3706 |
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 |
7 years ago |
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ea14243971 |
res_pjsip: Fix initialization of extended stringfields.
It is possible for initialization of extended stringfields to fail. Add checks for this failure. Change-Id: I062e09852db3d37ceefaf6c2048958fa0118304f |
7 years ago |
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9ad9433257 |
res_pjsip: Endpoint destruction does not free DTLS configuration
ASTERISK-27679 #close Reported by: Mak Dee Change-Id: I89a2783a11be0763bf123d1619ed176b6225cf42 |
7 years ago |
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f35960d55b |
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 |
7 years ago |
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96040fa6bb |
res_pjsip.c: Fix ident_to_str() and refactor ident_handler().
* Extracted sip_endpoint_identifier_type2str() and sip_endpoint_identifier_str2type() to simplify the calling functions. * Fixed pjsip_configuration.c:ident_to_str() building the endpoint's identify_by value string. Change-Id: Ide876768a8d5d828b12052e2a75008b0563fc509 |
7 years ago |
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82b6ba976f |
Fix Common Typo's.
Fix instances of: * Retreive * Recieve * other then * different then * Repeated words ("the the", "an an", "and and", etc). * othterwise, teh ASTERISK-24198 #close Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31 |
8 years ago |
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c3bc44fa1b |
pjsip_configuration: Add correct file header
Change-Id: I25348c386a222bb704aff07f54375108a6402906 |
8 years ago |
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366cc259bc |
res_pjsip: Fix warning by deferring implicit type cast.
Mac doesn't like the comparison of -1 to an enum, so store the result of ast_sip_str_to_dtmf to an int so we can check for the negative return value. ast_sip_str_to_dtmf returns an int so this is only delaying the implicit type cast. Change-Id: I0c262c1719ee951aae1f437d733a301cf5f8ad29 |
8 years ago |
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e4fba95022 |
res_pjsip: Fix leak on error in ast_sip_auth_vector_init.
Change-Id: Ib0fc7a18f3135ca8990c3984c9e15f6d26e556e8 |
8 years ago |
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ffcb7e2a25 |
res_pjsip: Add to list of valid characters for from_user.
Fixes a regression where some characters were unable to be used in the from_user field of an endpoint. Additionally, the backtick was removed from the list of valid characters, since it is not valid, and it was replaced with a single quote, which is a valid character. ASTERISK-27387 Change-Id: Id80c10a644508365c87b3182e99ea49da11b0281 |
8 years ago |
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7385d1e017 |
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 |
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f1163c0f6f |
res_pjsip: Fix leak of persistent endpoint references.
Do not manually call sip_endpoint_apply_handler from load_all_endpoints. This is not necessary and causes memory leaks. Additionally reinitialize persistent->aors when we reuse a persistent object with a new endpoint. ASTERISK-27306 Change-Id: I59bbfc8da8a14d5f4af8c5bb1e71f8592ae823eb |
8 years ago |
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ed2a4ee81e |
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 |
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07d026b4cd |
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 |
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423d01cf16 |
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 |
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25e18bf514 |
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 |
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9fbc34d2bd |
res_pjsip: Add DTMF INFO Failback mode
The existing auto dtmf mode reverts to inband if 4733 fails to be negotiated. This patch adds a new mode auto_info which will switch to INFO instead of inband if 4733 is not available. ASTERISK-27066 #close Change-Id: Id185b11e84afd9191a2f269e8443019047765e91 |
8 years ago |
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a6e4899612 |
res_pjsip: New endpoint option "notify_early_inuse_ringing"
This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711 |
8 years ago |
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6af2dd34af |
res_pjsip: New endpoint option "refer_blind_progress"
This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2 |
8 years ago |
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398e5ec16c |
res_pjsip_session: Enable RFC3578 overlap dialing support.
Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6 |
8 years ago |
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7bc69753bc |
Add rtcp-mux support
This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5 |
8 years ago |
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75497c33ea |
Free endpoint ACLs when destroying PJSIP endpoints.
If endpoint ACLs were specified, they were not being freed when endpoints were destroyed. On systems with realtime endpoints, this could add up quickly since each DB lookup would allocate the ACL without freeing it. ASTERISK-26731 #close Reported by Ustinov Artem Change-Id: Ie1f8bf5b7a0de628c975beba01e69c56893331ad |
8 years ago |
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8160474d7d |
res_pjsip: alloca can never fail.
Change-Id: Ia2a6158e5fdf311bc2a1c0c43417978de504b1f1 |
8 years ago |
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e0bc17edff |
pjsip: Fix a few media bugs with reinvites and asymmetric payloads.
When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc |
9 years ago |
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b5e753227d |
pjsip_configuration.c: Ignore repeated identify by methods.
Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838 |
9 years ago |
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c16ef02318 |
res_pjsip: Default endpoints to the "offline" status.
A recent change attempted to optimize startup by not updating contact status. Instead, code responsible for qualifying contacts updates the status as it becomes known. The code even accounts for contacts/AORs that are not set to be qualified. The problem, though, is when there are no contacts associated with an endpoint. A common case is when an endpoint is set to register its contacts but has not done so yet. In this case, prior to registration, the endpoint's device state will appear to be "not in use" and hints associated with that device will appear to be "idle". In actuality, the device state and hint should both appear as "unavailable". The reason for the failure is that the optimization change made all persistent endpoint states set to "unknown". The fix here is to change the hard-coded "unknown" to be "offline" instead. The default state will be offline until the qualifying code determines that the contact is actually online. This way, if there are no contacts at all, then the state stays as offline, and device state and hints appear correctly. ASTERISK-26269 #close Reported by nappsoft Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a |
9 years ago |
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e0c807c33d |
Merge "compilation failed with -Werror=maybe-uninitialized" into 13
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9 years ago |
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b494b9f88c |
compilation failed with -Werror=maybe-uninitialized
The compilation failed for devmode --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES --enable DO_CRASH --enable TEST_FRAMEWORK res_pjsip/pjsip_configuration.c: In function dtls_handler: res_pjsip/pjsip_configuration.c:974:20: error: back may be used uninitialized in this function [-Werror=maybe-uninitialized] int size = strlen(front); ^ cc1: all warnings being treated as errors Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580 |
9 years ago |
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329507fe20 |
res_pjsip: Add contact_user to endpoint
contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4 |
9 years ago |
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1fc5c90014 |
res_pjsip res_pjsip_mwi: Misc fixes and cleanups.
* Eliminated RAII_VAR() usage in ast_sip_persistent_endpoint_update_state(). * Added a missing allocation failure check to persistent_endpoint_find_or_create(). * Made persistent_endpoint_find_or_create() create the new object without a lock as it isn't needed. * Cleaned up some ao2 container allocation idioms. * Reordered res_pjsip_mwi.c load_module() and unload_module() Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8 |
9 years ago |
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851b1c3a17 |
res_pjsip: Add fax_detect_timeout endpoint option.
The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d |
9 years ago |
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02877b4b4f |
pjsip_configuration.c: Misc cleanups.
* Fix some whitespace in various routines. * Rename i to iter in persistent_endpoint_update_state(). * Fix off-nominal copy/paste message wording in persistent_endpoint_contact_deleted_observer() Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8 |
9 years ago |
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8cea01ab1b |
Merge "res_pjsip: Fix statsd regression." into 13
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9 years ago |
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97b4c7a5b4 |
res_pjsip: Fix statsd regression.
The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f patch introduced several regressions when the newly created "Updated" state goes out for each endpoint registration refresh. 1) It restarted any OPTIONS RTT ping cycle. 2) It would interfere with a currently active ping and throw off that ping's resulting RTT calculation. 3) It cleared the RTT time each time the endpoint was refreshed. 4) The cleared RTT time was sent out as a statsd update each time. 5) It created two AMI events for each update. * Revert the original patch and reimplement it. Now the current contact status state is re-sent instead of the state being momentarily toggled every time the endpoint refreshes its registration. The statsd events are not created for the re-sent refresh because they are sent after every OPTIONS ping. ASTERISK-26160 #close Reported by: Matt Jordan Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1 |
9 years ago |
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2c16a81dd5 |
res_pjsip: Added "subscribe_context" to endpoint
If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no "subscribe_context" is specified, then the "context" setting is used. ASTERISK-25471 #close Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514 |
9 years ago |
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b3c787d1dd |
res_pjsip: improve realtime performance #2
The patch removes updating all Endpoints' status on startup. Instead, only non-qualified aors with static contact and non-qualified non-expired contacts are retrieved from the realtime to update the endpoint status to ONLINE. The endpoint name was added to the contact object to simply find the endpoint that created this contact. The status of endpoints with qualified aors will be updated by 'qualify' functions. ASTERISK-26061 #close Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df |
9 years ago |
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04c12561a7 |
res_pjsip: chatty verbose messages
There are a lot of verbose messages about Endpoint and Contact status changes if there are many dynamic endpoints. The patch sets verbose level 2 for Endpoint status changes and verbose level 3 for Contact status changes. ASTERISK-26055 #close Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7 |
9 years ago |
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524a302974 |
res_pjsip: Endpoint IP Access Controls
With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02 |
9 years ago |
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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 |
9 years ago |
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d8f0bc3572 |
res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited
res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes on endpoints for unsolicited mwi and on aors for subscriptions required that the admin know in advance which the client wanted. If you specified mailboxes on the endpoint, subscriptions were rejected even if you also specified mailboxes on the aor. Voicemail extension: * Added a global default_voicemail_extension which defaults to "". * Added voicemail_extension to both endpoint and aor. * Added ast_sip_subscription_get_dialog for support. * Added ast_sip_subscription_get_sip_uri for support. When an unsolicited NOTIFY is constructed, the From header is parsed, the voicemail extension from the endpoint is substituted for the user, and the result placed in the Message-Account field in the body. When a subscribed NOTIFY is constructed, the subscription dialog local uri is parsed, the voicemail_extension from the aor (looked up from the subscription resource name) is substituted for the user, and the result placed in the Message-Account field in the body. If no voicemail extension was defined, the Message-Account field is not added to the NOTIFY body. mwi_subscribe_replaces_unsolicited: * Added mwi_subscribe_replaces_unsolicited to endpoint. The previous behavior was to reject a subscribe if a previous internal subscription for unsolicited MWI was found for the mailbox. That remains the default. However, if there are mailboxes also set on the aor and the client subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal subscription is removed and replaced with the external subscription. This allows an admin to configure mailboxes on both the endpoint and aor and allows the client to select which to use. ASTERISK-25865 #close Reported-by: Ross Beer Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea |
9 years ago |
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c7eb18d865 |
chan_pjsip: Add 'pjsip show channelstats'
Added the ability to show channel statistics to chan_pjsip (cli_functions.c) Moved the existing 'pjsip show channel(s)' functionality from pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's private header so it made sense to move the existing channel commands as well. Now using stasis_cache_dump to get the channel snapshots rather than retrieving all endpoints, then getting each one's channel snapshots. Much more efficient. Change-Id: I03b114522126d27434030b285bf6d531ddd79869 |
9 years ago |
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530cff5f5f |
res_pjsip: Strip spaces from items parsed from comma-separated lists
Configurations like "aors = a, b, c" were either ignoring everything after "a" or trying to look up " b". Same for mailboxes, ciphers, contacts and a few others. To fix, all the strsep(©, ",") calls have been wrapped in ast_strip. To facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were updated to handle null pointers. In some cases, an ast_strlen_zero() test was added to skip consecutive commas. There was also an attempt to ast_free an ast_strdupa'd string in ast_sip_for_each_aor which was causing a SEGV. I removed it. Although this issue was reported for realtime, the issue was in the res_pjsip modules so all config mechanisms were affected. ASTERISK-25829 #close Reported-by: Mateusz Kowalski Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2 |
9 years ago |
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b1b797e0e7 |
res_pjsip: Refactor load_module/unload_module
load_module was just too hairy with every step having to clean up all previous steps on failure. Some of the pjproject init calls have now been moved to a separate load_pjsip function and the unload_pjsip function was enhanced to clean up everything if an error happened at any stage of the load process. In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns and ast_threadpool_shutdowns were also corrected. Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302 |
9 years ago |
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236896f391 |
Merge "pjsip: Add option global/regcontext" into 13
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10 years ago |
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219c204a41 |
pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address
On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88 |
10 years ago |
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22801a06ee |
pjsip: Add option global/regcontext
Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62 |
10 years ago |
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21962dad93 |
res_pjsip: Add existence and readablity checks for tls related files
Both transport and endpoint now check for the existence and readability of tls certificate and key files before passing them on to pjproject. This will cause the object to not load rather than waiting for pjproject to discover that there's a problem when a session is attempted. NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located in build_peer which is gigantic and I didn't want to disturb it. Error messages will emit but it won't interrupt chan_sip loading. ASTERISK-25618 #close Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 Reported-by: George Joseph Tested-by: George Joseph |
10 years ago |
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450579e908 |
res_pjsip/contacts/statsd: Make contact lifecycle events more consistent
It will never be perfect or even pretty, mostly because of the differences between static and dynamic contacts. Created: Can't use the contact or contact_status alloc functions because the objects come and go regardless of the actual state. Can't use the contact_apply_handler, ast_sip_location_add_contact or a sorcery created handler because they only get called for dynamic contacts. Similarly, permanent_uri_handler only gets called for static contacts. So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is the only place it can go and not have duplicated code. Both permanent_uri_handler and contact_apply_handler call find_or_create. Removed: Can't use the destructors for the same reason as above. The only place to put this is in persistent_endpoint_contact_deleted_observer which I believe is the "correct" place but even that will handle only dynamic contacts. This doesn't called on shutdown however. There is no hook to use for static contacts that may be removed because of a config change while asterisk is in operation. I moved the cleanup of contact_status from ast_sip_location_delete_contact to the handler as well. Status Change and RTT: Although they worked fine where they were (in update_contact_status) I moved them to persistent_endpoint_contact_status_observer to make it more consistent with removed. There was logic there already to detect a state change. Finally, fixed a nit in permanent_uri_handler rmudgett reported eralier. ASTERISK-25608 #close Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d Reported-by: George Joseph Tested-by: George Joseph |
10 years ago |