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${ noResults }
3100 Commits (b5801fe42c37f4a9c075b2e96b108ce2344d23e9)
Author | SHA1 | Message | Date |
---|---|---|---|
|
b5801fe42c |
Merge "res_pjsip_session: Don't invoke session supplements twice for BYE requests." into 13
|
10 years ago |
|
c2df44ad3e |
Merge "res_pjsip: Add common ast_sip_get_host_ip API." into 13
|
10 years ago |
|
c2c7319082 |
res_pjsip_session: Don't invoke session supplements twice for BYE requests.
When a BYE request is received the PJSIP invite session implementation creates and sends a 200 OK response before we are aware of it. This causes the INVITE session state callback to be called into and ultimately the session supplements run on the BYE request. Once this response has been sent the normal transaction state callback is invoked which invokes the session supplements on the BYE request again. This can be problematic in particular with res_pjsip_rfc3326 as it may attempt to update the hangup cause code on the channel while it is in the process of being hung up. This change makes it so the session supplements are only invoked once by the INVITE session state callback. ASTERISK-25318 #close Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a |
10 years ago |
|
6862c2a167 |
Chaos: handle failed allocation in get_media_encryption_type
If the ast_strndup() call fails to allocate a copy of the transport string for parsing, fail gracefully. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28 |
10 years ago |
|
2a4eee0cd9 |
res_pjsip: Add common ast_sip_get_host_ip API.
Modules commonly used the pj_gethostip function for retrieving the IP address of the host. This function does not cache the result and may result in a DNS lookup occurring, or additional work. If the DNS server is unreachable or network issues arise this can cause the pj_gethostip function to block for a period of time. This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string function which does the same thing but caches the host IP address at module load time. This results in no additional work being done each time the local host IP address is needed. ASTERISK-25342 #close Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e |
10 years ago |
|
d238cf33a9 |
Merge "res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced" into 13
|
10 years ago |
|
5f781cb799 |
Merge "res_pjsip/pjsip_configuration: Disregard empty auth values" into 13
|
10 years ago |
|
7c4d0c3506 |
res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced
When recreating a subscription it is possible for a freed sub_tree to be referenced when the initial NOTIFY fails to be created. Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788 |
10 years ago |
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bc6fe07f5c |
res_pjsip/pjsip_configuration: Disregard empty auth values
When an endpoint is backed by a non-static conf file backend (such as the AstDB or Realtime), the 'auth' object may be returned as being an empty string. Currently, res_pjsip will interpret that as being a valid auth object, and will attempt to authenticate inbound requests. This isn't desired; is an auth value is empty (which the name of an auth object cannot be), we should instead interpret that as being an invalid auth object and skip it. ASTERISK-25339 #close Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7 |
10 years ago |
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0582776f7f |
ari/ari_websockets.c: Fix ast_debug parameter type mismatch.
This is a type mismatch fix of the debugging commit
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10 years ago |
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94d93e4d40 |
Merge "res_ari.c: Add missing off nominal unlock and remove a RAII_VAR()." into 13
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10 years ago |
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77518d5434 |
res_http_websocket.c: Fix some off nominal path cleanup.
* Remove extraneous unlock on off-nominal path. * Add missing HTTP error reply. Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b |
10 years ago |
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c61547fee6 |
res_ari.c: Add missing off nominal unlock and remove a RAII_VAR().
Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf |
10 years ago |
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b719f56c72 |
res_pjsip_sdp_rtp: Restore removed NULL check.
When sending an RTP keepalive, we need to be sure we're not dealing with a NULL RTP instance. There had been a NULL check, but the commit that added the rtp_timeout and rtp_hold_timeout options removed the NULL check. Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64 |
10 years ago |
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e18c300550 |
res_http_websocket: When shutting down a session don't close closed socket
Due to the use of ast_websocket_close in session termination it is possible for the underlying socket to already be closed when the session is terminated. This occurs when the close frame is attempted to be written out but fails. Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b |
10 years ago |
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0bc6fe6f7d |
Merge "res_http_websocket: Forcefully terminate on write errors." into 13
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10 years ago |
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b4e9416138 |
res_http_websocket: Forcefully terminate on write errors.
The res_http_websocket module will currently attempt to close the WebSocket connection if fatal cases occur, such as when attempting to write out data and being unable to. When the fatal cases occur the code attempts to write a WebSocket close frame out to have the remote side close the connection. If writing this fails then the connection is not terminated. This change forcefully terminates the connection if the WebSocket is to be closed but is unable to send the close frame. ASTERISK-25312 #close Change-Id: I10973086671cc192a76424060d9ec8e688602845 |
10 years ago |
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c126afe18f |
res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message.
If the saved SUBSCRIBE message is not parseable for whatever reason then Asterisk could crash when libpjsip tries to parse the message and adds an error message to the parse error list. * Made ast_sip_create_rdata() initialize the parse error rdata list. The list is checked after parsing to see that it remains empty for the function to return successful. ASTERISK-25306 Reported by Mark Michelson Change-Id: Ie0677f69f707503b1a37df18723bd59418085256 |
10 years ago |
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47d9ff1741 |
Merge "res/res_format_attr_silk: Expose format attributes to other modules" into 13
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10 years ago |
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c57b78d4c9 |
Merge "Replace htobe64 with htonll" into 13
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10 years ago |
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f733bc10b1 |
Merge "res_pjsip_pubsub: More accurately persist packet." into 13
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10 years ago |
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8e194047ac |
res/res_format_attr_silk: Expose format attributes to other modules
This patch adds the .get callback to the format attribute module, such that the Asterisk core or other third party modules can query for the negotiated format attributes. Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c |
10 years ago |
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26f0559a94 |
Replace htobe64 with htonll
We don't have a compatability function to fill in a missing htobe64; but we already have one for the identical htonll. Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac |
10 years ago |
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df9ce36366 |
ARI: Retrieve existing log channels
An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808 |
10 years ago |
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e9f1bc08cb |
ARI: Creating log channels
An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782 |
10 years ago |
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cf27200391 |
Merge "ARI: Deleting log channels" into 13
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10 years ago |
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4b1bd40d7e |
Merge "res_pjsip: Ensure sanitized XML is NULL terminated." into 13
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10 years ago |
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78364132ce |
ARI: Deleting log channels
An http request can be sent to delete a log channel in Asterisk. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/logging/mylog'" can be run in the terminal to access the newly implemented functionally for ARI. * Able to delete log channels using ARI ASTERISK-25252 Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6 |
10 years ago |
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e25569ef95 |
res_pjsip_pubsub: More accurately persist packet.
The pjsip_rx_data structure has a pkt_info.packet field on it that is the packet that was read from the transport. For datagram transports, the packet read from the transport will correspond to the SIP message that arrived. For streamed transports, however, it is possible to read multiple SIP messages in one packet. In a recent case, Asterisk crashed on a system where TCP was being used. This is because at some point, a read from the TCP socket resulted in a 200 OK response as well as an incoming SUBSCRIBE request being stored in rdata->pkt_info.packet. When the SUBSCRIBE was processed, the combination 200 OK and SUBSCRIBE was saved in persistent storage. Later, a restart of Asterisk resulted in the crash because the persistent subscription recreation code ended up building the 200 OK response instead of a SUBSCRIBE request, and we attempted to access request-specific data. The fix here is to use the pjsip_msg_print() function in order to persist SUBSCRIBE requests. This way, rather than using the raw socket data, we use the parsed SIP message that PJSIP has given us. If we receive multiple SIP messages from a single read, we will be sure only to save off the relevant SIP message. There also is a safeguard put in place to make sure that if we do end up reconstructing a SIP response, it will not cause a crash. ASTERISK-25306 #close Reported by Mark Michelson Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2 |
10 years ago |
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9182c9e4e6 |
Merge "res_rtp_asterisk.c: Fix off-nominal crash potential." into 13
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10 years ago |
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ba40b07ddc |
Merge topic 'misc_rtp_tweaks' into 13
* changes: rtp_engine.c: Must protect mime_types_len with mime_types_lock. res_pjsip_sdp_rtp.c: Fixup some whitespace. |
10 years ago |
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8521a86367 |
res_pjsip: Ensure sanitized XML is NULL terminated.
The ast_sip_sanitize_xml function is used to sanitize a string for placement into XML. This is done by examining an input string and then appending values to an output buffer. The function used by its implementation, strncat, has specific behavior that was not taken into account. If the size of the input string exceeded the available output buffer size it was possible for the sanitization function to write past the output buffer itself causing a crash. The crash would either occur because it was writing into memory it shouldn't be or because the resulting string was not NULL terminated. This change keeps count of how much remaining space is available in the output buffer for text and only allows strncat to use that amount. Since this was exposed by the res_pjsip_pidf_digium_body_supplement module attempting to send a large message the maximum allowed message size has also been increased in it. A unit test has also been added which confirms that the ast_sip_sanitize_xml function is providing NULL terminated output even when the input length exceeds the output buffer size. ASTERISK-25304 #close Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302 |
10 years ago |
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c07fa843ec |
Merge "res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list." into 13
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10 years ago |
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56d11d4198 |
Merge "res_rtp_asterisk: Don't leak temporary key when enabling PFS." into 13
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10 years ago |
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9a12804e59 |
res_rtp_asterisk: Don't leak temporary key when enabling PFS.
A change recently went in which enabled perfect forward secrecy for DTLS in res_rtp_asterisk. This was accomplished two different ways depending on the availability of a feature in OpenSSL. The fallback method created a temporary instance of a key but did not free it. This change fixes that. ASTERISK-25265 Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396 |
10 years ago |
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27dc2094e9 |
res_http_websocket: Debug write lengths.
Commit |
10 years ago |
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1aa23a5d1b |
Merge "res_http_websocket: Avoid passing strlen() to ast_websocket_write()." into 13
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10 years ago |
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39cc28f6ea |
res_http_websocket: Avoid passing strlen() to ast_websocket_write().
We have seen a rash of test failures on a 32-bit build agent. Commit
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10 years ago |
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aed068844c |
res/res_rtp_asterisk: Add ECDH support
This will add ECDH support to Asterisk. It will detect auto ECDH support in OpenSSL (1.0.2b and above) during ./configure. If this is available, it will use it, otherwise it will fall back to prime256v1 (this behavior is consistent with other projects such as Apache and nginx). This fixes WebRTC being broken in Firefox 38+ due to Firefox now only supporting ciphers with perfect forward secrecy. ASTERISK-25265 #close Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b |
10 years ago |
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e28fbebc57 |
Merge "ARI: Rotate log channels." into 13
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10 years ago |
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1ae762634c |
ARI: Rotate log channels.
An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01 |
10 years ago |
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84262749d2 |
res_rtp_asterisk.c: Fix off-nominal crash potential.
ASTERISK-25296 Reported by: Richard Mudgett Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b |
10 years ago |
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a93b7a927c |
res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list.
Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2 |
10 years ago |
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741fa0d26d |
res_pjsip_sdp_rtp.c: Fixup some whitespace.
Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973 |
10 years ago |
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13eb491e35 |
res_pjsip_session.c: Fix crashes seen when call cancelled.
Two testsuite tests crashed in the same place as a result of an INVITE being CANCELed. tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp The session pointer is no longer in the inv->mod_data[session_module.id] location because the INVITE transaction has reached the terminated state. ASTERISK-25297 #close Reported by: Richard Mudgett Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427 |
10 years ago |
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48698a5e21 |
res_http_websocket: Properly encode 64 bit payload
A test agent was continuously failing all ARI tests when run against Asterisk 13. As it turns out, the reason for this is that on those test runs, for some reason we decided to use the super extended 64 bit payload length for websocket text frames instead of the extended 16 bit payload length. For 64-bit payloads, the expected byte order over the network is 7, 6, 5, 4, 3, 2, 1, 0 However, we were sending the payload as 3, 2, 1, 0, 7, 6, 5, 4 This meant that we were saying to expect an absolutely MASSIVE payload to arrive. Since we did not follow through on this expected payload size, the client would sit patiently waiting for the rest of the payload to arrive until the test would time out. With this change, we use the htobe64() function instead of htonl() so that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a |
10 years ago |
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2749721791 |
pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9 |
10 years ago |
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d9094ddd73 |
res_pjsip: Add rtp_keepalive endpoint option.
This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b |
10 years ago |
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a23adcca3d |
res/res_musiconhold: Add a warning when MOH does not exist
Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b |
10 years ago |
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03064daeb2 |
res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf
Misconfiguring sorcery.conf with a 'config' wizard with no extra data will currently crash Asterisk on startup, as the wizard requires a comma delineated list to parse. This patch updates res_sorcery_config to check for the presence of the data before it starts manipulating it. Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847 |
10 years ago |