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${ noResults }
4598 Commits (de5144e751ae9d4dd28515768cb15ade61063965)
Author | SHA1 | Message | Date |
---|---|---|---|
|
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 |
|
3f7a75b481 |
Merge "res_pjsip_messaging: Allow application/* for in-dialog MESSAGEs"
|
7 years ago |
|
e3585353f6 |
res_pjsip_messaging: Allow application/* for in-dialog MESSAGEs
In addition to text/* content types, incoming_in_dialog_request now accepts application/* content types. Also fixed a length issue when copying the body text. It was one character short. ASTERISK-27942 Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818 |
7 years ago |
|
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 |
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017b7849bc |
Merge "VECTOR: Passing parameters with side effects to macros is dangerous."
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7 years ago |
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cad50d6dbf |
VECTOR: Passing parameters with side effects to macros is dangerous.
* Fix several instances where we were bumping a ref in the parameter and then unrefing the object if it failed. The way the AST_VECTOR_APPEND() and AST_VECTOR_REPLACE() macros are implemented means if it fails the new value was never evaluated. Change-Id: I2847872a455b11ea7e5b7ce697c0a455a1d0ac9a |
7 years ago |
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bbea9cfc3b |
res_http_post: Enable GMime in Solaris 11.
Change-Id: Ie434541f18f894c751d2e44bcb3efb3cac626019 |
7 years ago |
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da54605b8a |
ARI POST DTMF: Make not compete with channel's media thread.
There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e |
7 years ago |
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986cffa18e |
Merge "app_confbridge: Enable sending events to participants"
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7 years ago |
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e7a7506f9c |
app_confbridge: Enable sending events to participants
ConfBridge can now send events to participants via in-dialog MESSAGEs. All current Confbridge events are supported, such as ConfbridgeJoin, ConfbridgeLeave, etc. In addition to those events, a new event ConfbridgeWelcome has been added that will send a list of all current participants to a new participant. For all but the ConfbridgeWelcome event, the JSON message contains information about the bridge, such as its id and name, and information about the channel that triggered the event such as channel name, callerid info, mute status, and the MSID labels for their audio and video tracks. You can use the labels to correlate callerid and mute status to specific video elements in a webrtc client. To control this behavior, the following options have been added to confbridge.conf: bridge_profile/enable_events: This must be enabled on any bridge where events are desired. user_profile/send_events: This must be set for a user profile to send events. Different user profiles connected to the same bridge can have different settings. This allows admins to get events but not normal users for instance. user_profile/echo_events: In some cases, you might not want the user triggering the event to get the event sent back to them. To prevent it, set this to false. A change was also made to res_pjsip_sdp_rtp to save the generated msid to the stream so it can be re-used. This allows participant A's video stream to appear as the same label to all other participants. Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e |
7 years ago |
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b01fc2ef3d |
res_rtp_asterisk: Instead of ./configure use OPENSSL_NO_SRTP.
Previously, Asterisk used its script ./configure, to test whether OpenSSL was built with no-srtp (or was simply too old). However, the header file <openssl/opensslconf.h> is the preferred way to detect the local configuration of OpenSSL. As a positive side-effect the script ./configure does not interleave the detection of the Open Settlement Protocol Toolkit (OSPTK) with the detection of individual features of OpenSSL anymore. Change-Id: I3c77c7b00b2ffa2e935632097fa057b9fdf480c0 |
7 years ago |
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8b4c72caea |
Merge "res_rtp_asterisk: Allow OpenSSL configured with no-deprecated."
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7 years ago |
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782064852a |
Merge "res_crypto: Allow OpenSSL configured with no-deprecated."
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7 years ago |
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a46e658bc0 |
Merge "res_srtp: Repair ./configure --with-ssl=PATH."
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7 years ago |
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81ac32a85f |
AST-2018-008: Fix enumeration of endpoints from ACL rejected addresses.
When endpoint specific ACL rules block a SIP request they respond with a 403 forbidden. However, if an endpoint is not identified then a 401 unauthorized response is sent. This vulnerability just discloses which requests hit a defined endpoint. The ACL rules cannot be bypassed to gain access to the disclosed endpoints. * Made endpoint specific ACL rules now respond with a 401 unauthorized which is the same as if an endpoint were not identified. The fix is accomplished by replacing the found endpoint with the artificial endpoint which always fails authentication. ASTERISK-27818 Change-Id: Icb275a54ff8e2df6c671a6d9bda37b5d732b3b32 |
7 years ago |
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0743ad6422 |
res_rtp_asterisk: Allow OpenSSL configured with no-deprecated.
Furthermore, allow OpenSSL configured with no-dh. Additionally, this change allows auto-negotiation of the elliptic curve/group for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer. This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a side-effect. ASTERISK-27910 Change-Id: I5b0dd47c5194ee17f830f869d629d7ef212cf537 |
7 years ago |
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ca682f0030 |
res_crypto: Allow OpenSSL configured with no-deprecated.
The header <openssl/rsa.h> had to be included explicitly. ASTERISK-27906 Change-Id: I41743801eed998c039d73db7a0762d104a4f75b2 |
7 years ago |
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234bf4b7ff |
res_srtp: Repair ./configure --with-ssl=PATH.
ASTERISK-27905 Change-Id: Ibb7dc148a0048f4f9c3b12937ba4240dff0d15e2 |
7 years ago |
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7b5fc5d20f |
Merge "pjsip_options: handle modification of qualify options in realtime"
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7 years ago |
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7af5e86821 |
pjsip_options: show/reload AOR qualify options using CLI
Currentrly pjsip_options code does not handle the situation when the AOR qualify options were changed. Also there is no way to find out what qualify options are using. This patch add CLI commands to show and synchronize Aor qualify options: pjsip show qualify endpoint <id> Show the current qualify options for all Aors on the PJSIP endpoint. pjsip show qualify aor <id> Show the PJSIP Aor current qualify options. pjsip reload qualify endpoint <id> Synchronize the qualify options for all Aors on the PJSIP endpoint. pjsip reload qualify aor <id> Synchronize the PJSIP Aor qualify options. ASTERISK-27872 Change-Id: I1746d10ef2b7954f2293f2e606cdd7428068c38c |
7 years ago |
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e46b442e38 |
pjsip_options: handle modification of qualify options in realtime
Currentrly pjsip_options code does not handle the situation when the qualify options were changed in realtime database. Only 'module reload res_pjsip' helps. This patch add a check on contact add/update observers if the contact qualify options are different than local aor qualify options. If the qualify options were modified then synchronize the pjsip_options AOR local state. ASTERISK-27872 Change-Id: Id55210a18e62ed5d35a88e408d5fe84a3c513c62 |
7 years ago |
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4ea98e49f1 |
Merge "rtp: Add support for RTP extension negotiation and abs-send-time."
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8 years ago |
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c5d2bf05f4 |
res/res_rtp_asterisk: ensure marker bit is correctly set on ssrc change
Certain race conditions between changing bridge types and DTMF can cause the current FLAG_NEED_MARKER_BIT to send the marker bit before the actual first packet of native bridging. This logic keeps track of the ssrc the bridge is currently sending and will correctly ensure the marker bit is set if SSRC as changed from the previous sent packet. ASTERISK-27845 Change-Id: I01858bd0235f1e5e629e20de71b422b16f55759b |
8 years ago |
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25764691b0 |
Merge "netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API"
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8 years ago |
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a507c73a78 |
rtp: Add support for RTP extension negotiation and abs-send-time.
When RTP was originally created it had the ability to place a single extension in an RTP packet. In practice people wanted to potentially put multiple extensions in one and so RFC 5285 (obsoleted by RFC 8285) came into existence. This allows RTP extensions to be negotiated with a unique identifier to be used in the RTP packet, allowing multiple extensions to be present in the packet. This change extends the RTP engine API to add support for this. A user of it can enable extensions and the API provides the ability to retrieve the information (to construct SDP for example) and to provide negotiated information (from SDP). The end result is that the RTP engine can then query to see if the extension has been negotiated and what unique identifier is to be used. It is then up to the RTP engine implementation to construct the packet appropriately. The first extension to use this support is abs-send-time which is defined in the REMB draft[1] and is a second timestamp placed in an RTP packet which is for when the packet has left the sending system. It is used to more accurately determine the available bandwidth. ASTERISK-27831 [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 Change-Id: I508deac557867b1e27fc7339be890c8018171588 |
8 years ago |
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9f9dce05b2 |
netsock2: Add ast_sockaddr_resolve_first_af to netsock2 public API
This function originally was used in chan_sip to enable some simplifying assumptions and eventually was copy and pasted into res_pjsip_logger and res_hep. Since it's replicated in three places, it's probably best to move it into the public netsock2 API for these modules to use. Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04 |
8 years ago |
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f10fc135d4 |
res_pjsip_endpoint_identifier_ip: Unregister the module for headers.
Asterisk uses Reference Counting to track whether a module can be unloaded. Every consumer who requires a module, increases the reference count. When the consumer goes, is unloaded itself, it has to decrease the reference count on all its used/required modules. That way core stop gracefully works on the command-line interface (CLI): One module after the other is unloaded. A recent change broke this for the module res_pjsip. ASTERISK-27861 Change-Id: I261abcb411d026bbb0691cc78f28300bfd3103a3 |
8 years ago |
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195af35026 |
Merge "Fix GCC 8 build issues."
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8 years ago |
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a103221de2 |
Merge "pjsip: Rewrite OPTIONS support with new eyes."
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8 years ago |
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b5914d90ac |
Fix GCC 8 build issues.
This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84 |
8 years ago |
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8f55f7c333 |
res_hep: Adds hostname resolution support for capture_address
Previously, only an IP address would be accepted for the capture_address config setting in hep.conf. This change allows capture_address to be a resolvable hostname or an IP address. ASTERISK-27796 #close Reported-By: Sebastian Gutierrez Change-Id: I33e1a37a8b86e20505dadeda760b861a9ef51f6f |
8 years ago |
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57ad488451 |
Merge "res_ari: Remove requirement that body exists when debug is on."
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8 years ago |
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8e228fc138 |
Merge "res_pjsip/pjsip_distributor.c: Add missing off-nominal request response."
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8 years ago |
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0565a6c909 |
Merge "Add the ability to read the media file type from HTTP header for playback"
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8 years ago |
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e538fc8e86 |
Merge "res_rtp_asterisk: Always update SRTP on local SSRC change."
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8 years ago |
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706b899358 |
Merge "pjsip: Increase maximum number of usable ciphers & other cleanups"
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8 years ago |
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de3ca9bada |
res_ari: Remove requirement that body exists when debug is on.
The "ari set debug" code for incoming requests incorrectly assumed that all requests would contain a body. If one did not exist the request would be incorrectly rejected. The response that was sent was also incomplete as an incorrect function was used to construct the response. The code has now been changed to no longer require a request to have a body and the response updated to use the correct function. ASTERISK-27801 Change-Id: I4eef036ad54550a4368118cc348765ecac25e0f8 |
8 years ago |
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239074c759 |
pjsip: Increase maximum number of usable ciphers & other cleanups
* Increase maximum number of ciphers from 100 to 256 (or whatever PJ_SSL_SOCK_MAX_CIPHERS is #define'd to) * Simplify logic in cipher_name_to_id() * Make signed/unsigned comparison consistent Re: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=897412 Reported by: Ondřej Holas Change-Id: Iea620f03915a1b873e79743154255c3148a514e7 |
8 years ago |
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11b7de82c5 |
res_pjsip/pjsip_distributor.c: Add missing off-nominal request response.
Change-Id: I389579b39c523d1d1e8ce020ef549a8bb5781c9b |
8 years ago |
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6cab3c836a |
res_pjsip/pjsip_distributor.c: Pull some assignments out of if tests.
Change-Id: I3d30d638b53a4bbe9bf9aad853c649d583894112 |
8 years ago |
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afdca5c68c |
res_rtp_asterisk: Always update SRTP on local SSRC change.
When the local SSRC changes we need to update the SRTP information so that the proper key is used. This is commonly done as a result of bridging two channels together. Previously we only updated the SRTP information if media had already flowed, but in practice the channel driver may have already performed SRTP negotiation and set up the previous SSRC. We now always do it on a local SSRC change. ASTERISK-27795 ASTERISK-27800 Change-Id: Ia7c8e74c28841388b5244ac0b8fd6c1dc6ee4c10 |
8 years ago |
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0827d5cc53 |
Add the ability to read the media file type from HTTP header for playback
How it works today: media_cache tries to parse out the extension of the media file to be played from the URI provided to Asterisk while caching the file. What's expected: Better will be to have Asterisk get extension from other ways too. One of the common ways is to get the type of content from the CONTENT-TYPE header in the HTTP response for fetching the media file using the URI provided. Steps to Reproduce: Provide a URL of the form: http://host/media/1234 to Asterisk for media playback. It fails to play and logs show the following error line: [Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c: File http://host/media/1234 does not exist in any format Scenario this issue is blocking: In the case where the media files are stored in some cloud object store, following can block the media being played via Asterisk: Cloud storage generally needs authenticated access to the storage. The way to do that is by using signed URIs. With the signed URIs there's no way to preserve the name of the file. In most cases Cloud storage returns a key to access the object and preserving file name is also not a thing there ASTERISK-27286 Reporter: Gaurav Khurana Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89 |
8 years ago |
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3bad41257b |
Merge "BuildSystem: Add DragonFly BSD."
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8 years ago |
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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 |
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9c430569d4 |
Merge "bridge_softmix: Forward TEXT frames"
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8 years ago |
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661fec4b59 |
core: Remove unused/incomplete SDP modules.
Change-Id: Icc28fbdc46f58e54a21554e6fe8b078f841b1f86 |
8 years ago |
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56a9338fc1 |
Merge "Build System: Add missing ASTMM_LIBC to flex output."
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8 years ago |
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efe40ff671 |
BuildSystem: Add DragonFly BSD.
ASTERISK-27820 Change-Id: I310896143e94d65da1c2be3bb448204a8b86d557 |
8 years ago |
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6ccf08c543 |
Merge "stringfields: Collect extended stringfields into the stringfield section."
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8 years ago |
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7b59cfc777 |
Merge "res_pjsip: Fix deadlock on reliable transport shutdown."
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8 years ago |