asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	f27c1c7dc2	Merge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus"	9 years ago
zuul	8982d0e007	Merge "res_xmpp: Include client name in connection related error messages"	9 years ago
Sean Bright	d08c69a9e2	res_pjsip_sdp_rtp: Set hangup cause for RTP timeouts chan_sip sets the hangup cause code to AST_CAUSE_REQUESTED_CHAN_UNAVAIL (44) when a channel is hung up due to an RTP timeout. So do the same when it happens with PJSIP for parity. Change-Id: I3546ebbde6460c22a27c9da1bf321711b5961ab8	9 years ago
Joshua Colp	6666deb907	Merge "res_xmpp: Don't crash when trying to send a message without a connection"	9 years ago
zuul	1a626ffb89	Merge "res_xmpp: Correctly check return value of SSL_connect"	9 years ago
zuul	d3ca0e0bc4	Merge "res_xmpp: Try to provide useful errors messages from OpenSSL"	9 years ago
Joshua Colp	59df60efb9	Merge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor."	9 years ago
Kevin Harwell	d2f2cdf476	AMI: Updated version Updated the AMI version for the following reason (see CHANGES for more details): The 'PJSIPShowEndpoint' command's response event of 'IdentifyDetail' now contains a new optional parameter, 'MatchHeader'. Change-Id: Ie206913ef1dcfa6a2ebe3282da2387e52d6f05b9	9 years ago
Kevin Harwell	12dde3b568	pjproject_bundled: raise timeout value used when downloading After configuring Asterisk with '--with-pjproject-bundled' the configure/build process attempts to download pjproject from its download site. Currently, a timeout of 10 seconds is used that will stop the download process if pjproject has not been fully downloaded in that time. For some systems this was not enough time and the process was timing out too early. This patch raises the download timeout value to '60'. Also, this patch fixes another bug where the DOWNLOAD_TIMEOUT variable was not being properly exported due to a naming error. DOWNLOAD_MAX_TIMEOUT is now properly renamed to DOWNLOAD_TIMEOUT. ASTERISK-26814 #close Change-Id: Ia56e4e8a3d39db76bc8a1852b2cf07ec10b39842	9 years ago
Sean Bright	98a88e9ffa	res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus The documentation for JABBER_STATUS (and the deprecated JabberStatus app) indicate that a return value of 7 indicates that the specified buddy was not in the roster. It also indicates that you can specify a "bare" JID (one without a resource). Unfortunately the actual behavior does not match the documented behavior. Assuming that our roster includes the buddy online and available "valid@example.org/Valid" and does not include the buddy "invalid@example.org", the JABBER_STATUS() function returns the following before this patch: +------------------------------+------------+--------------------------+ \| Buddy \| Status \| Result \| +------------------------------+------------+--------------------------+ \| valid@example.org \| Online \| 7 (Not in roster) \| \| valid@example.org/Valid \| Online \| 1 (Online) \| \| valid@example.org/Invalid \| N/A \| 7 (Not in roster) \| \| invalid@example.org \| N/A \| Error logged, no return \| \| invalid@example.org/Valid \| N/A \| Error logged, no return \| +------------------------------+------------+--------------------------+ And after this patch: +------------------------------+------------+--------------------------+ \| Buddy \| Status \| Result \| +------------------------------+------------+--------------------------+ \| valid@example.org \| Online \| 1 (Online) \| \| valid@example.org/Valid \| Online \| 1 (Online) \| \| valid@example.org/Invalid \| N/A \| 6 (Offline) \| \| invalid@example.org \| N/A \| 7 (Not in roster) \| \| invalid@example.org/Valid \| N/A \| 7 (Not in roster) \| +------------------------------+------------+--------------------------+ This brings the behavior in line with the documentation. ASTERISK-23510 #close Reported by: Anthony Critelli Change-Id: I9c3241035363ef4a6bdc21fabfd8ffcd9ec657bf	9 years ago
Sean Bright	be94105d6d	res_xmpp: Try to provide useful errors messages from OpenSSL If any errors occur during the TLS connection setup, we currently dump a fairly generic error message. So instead we try to pull in something useful from OpenSSL to report instead. ASTERISK-24712 Reported by: Matthias Urlichs Change-Id: I288500991a9681f447d92913b11fedaf426087f4	9 years ago
Sean Bright	ee81ee1f14	res_xmpp: Fix ref counting issue The only remaining reference to the endpoint is in the endpoints container, and because it is unlinked in ast_endpoint_shutdown, we don't have to explicitly cleanup the endpoint ourselves. Change-Id: I912a2692e52d3e2ed445b32d8ae3f9004bc2f2e8	9 years ago
Sean Bright	9493981419	res_xmpp: Correctly check return value of SSL_connect SSL_connect returns non-zero for both success and some error conditions so simply negating is inadequate. Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1	9 years ago
Sean Bright	7657c279b5	res_xmpp: Don't crash when trying to send a message without a connection If we never establish a connection to our Jabber server, iksemel never sets up its internal transport pointer, so attempting to send a message dereferences a NULL pointer and causes a crash. ASTERISK-21855 #close Reported by: Jeremy Kister Change-Id: I204a568894e4a53ab929783ecc594a000f04d79c	9 years ago
Sean Bright	0136ec12a3	res_xmpp: Include client name in connection related error messages ASTERISK-25622 #close Reported by: Sean Darcy Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9	9 years ago
Joshua Colp	c1ab8ca74c	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."	9 years ago
Joshua Colp	3a50311c17	Merge "CHANNEL(callid): Give dialplan access to the callid."	9 years ago
Kevin Harwell	9b103e7bea	rtp_engine: allocate RTP dynamic payloads per session Dynamic payload types were statically defined in Asterisk. This unfortunately limited the number of dynamic payloads that could be registered. With this patch dynamic payload type numbers are now assigned dynamically and per RTP instance. However, in order to limit any issues where some clients expect the old statically defined value this patch makes it so the value Asterisk used to pre- designate is used for the dynamic assignment if available. An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf) that turns the new dynamic behavior on or off. When off it reverts back to using statically defined payload values. This option defaults to "yes" in Asterisk 15. ASTERISK-26515 #close patches: ASTERISK-26515.diff submitted by jcolp (license 5000 Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc	9 years ago
zuul	06c9966608	Merge "res_pjsip_messaging: Check URI type before dereferencing"	9 years ago
zuul	58b42297fd	Merge "Revert "app_queue: Handle the caller being redirected out of a queue bridge""	9 years ago
zuul	30f011a460	Merge "app_queue: Member stuck as pending after forwarding previous call from queue"	9 years ago
Sebastian Gutierrez	bb2936f3e4	cdr: Allow setting of user field from 'h' extension The CDR code previously did not allow the user field to be set from the 'h' extension in the dialplan. This change removes that limitation and allows it to be set. ASTERISK-26818 Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6	9 years ago
zuul	f83a334434	Merge "pjsip: prevent memory corruption on creation of xml bodies"	9 years ago
Richard Begg	6b7697ed48	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	9 years ago
zuul	9f64980e60	Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references."	9 years ago
zuul	bd4a16da04	Merge "res_hep: Capture actual transport type in use"	9 years ago
Sean Bright	d4fcf196a2	res_hep: Capture actual transport type in use Rather than hard-coding UDP, allow consumers of the HEP API to specify which protocol is in use. Update the PJSIP provider to pass in the current protocol type. ASTERISK-26850 #close Change-Id: I54bbb0a001cfe4c6a87ad4b6f2014af233349978	9 years ago
Sean Bright	1bf839d44b	Revert "app_queue: Handle the caller being redirected out of a queue bridge" This reverts commit `163e9e53dc`. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b	9 years ago
Sean Bright	6b4b87787c	res_pjsip_messaging: Check URI type before dereferencing We aren't validating that the URI we just parsed is a SIP/SIPS one before trying to access the user, host, and port members of a possibly uninitialized structure. Also update the MessageSend documentation to indicate what 'from' formats are accepted. ASTERISK-26484 #close Reported by: Vinod Dharashive Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30	9 years ago
Joshua Elson	65ad554c98	pjsip: prevent memory corruption on creation of xml bodies ASTERISK-26776 #close Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2	9 years ago
Sean Bright	fc794de756	bridge_softmix: Ignore non-voice frames from translator Some codecs - codec_speex specifically - take voice frames and return other types of frames, like CNG. If we subsequently treat those as voice frames, we'll run into trouble when destroying the frame because of the requirement that each voice frame have an associated format. ASTERISK-26880 #close Reported by: Kirsty Tyerman Change-Id: I43f8450c48fb276ad8b99db8512be82949c1ca7c	9 years ago
Joshua Colp	f5603cb1ec	Merge "res/res_pjsip_session: Only check localnet if it is defined"	9 years ago
Aaron An	25016a74f8	audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor. Fixed a bug in function "ast_audiohook_write_frame" that checked the variable other_factory_samples and only flushed the factories, so they would be in sync, when other_factory_samples > 0. When there is not any rtp incoming the variable other_factory_samples will be 0, and although the result of "our_factory_ms - other_factory_ms" may be very large, this led to the record file not syncing. ASTERISK-26875 #close Reported-by: Aaron An Tested-by: Aaron An Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22	9 years ago
zuul	24ada76408	Merge "thread safety: Don't use getprotobyname()"	9 years ago
Sean Bright	fc71c18a9b	thread safety: Don't use getprotobyname() POSIX does not require getprotobyname() to be thread safe and some implementations use static memory which causes issues when multiple threads are used. Further, our usage of it today is just to ultimately get IPPROTO_TCP for calls to setsockopt(). So instead we just use IPPROTO_TCP directly. Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48	9 years ago
Sean Bright	516e028b44	res_rtp_asterisk: Pass correct data length to ast_rtcp_interpret We are currently passing in the capacity of the read buffer instead of the number of bytes that we actually read off the wire. Change-Id: I60465049727d955c7f9a5e529e6f2aaff04cda36	9 years ago
Joshua Colp	b05d2fda0c	Merge "app_queue: Fix locking behavior in stasis message handlers"	9 years ago
Joshua Colp	3996c79328	Merge "chan_sip: Add rtcp-mux support"	9 years ago
Joshua Colp	77582634d7	Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is stopped."	9 years ago
Joshua Colp	0db211dc64	Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed."	9 years ago
Joshua Colp	317405ce69	Merge "app_confbridge: Fix ConfbridgeTalking AMI event description."	9 years ago
Joshua Colp	7f87cd7b4e	Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error"	9 years ago
Joshua Colp	15c72b3239	Merge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport"	9 years ago
Robert Mordec	79069f8ccb	app_queue: Member stuck as pending after forwarding previous call from queue Queue member will get stuck in pending_members if queue calls a device that is different from the one observed for state changes. This patch removes members from pending_members as a result of channel stasis events such as blind or attended transfers and hangup. ASTERISK-26862 #close Change-Id: I8bf6df487b9bb35726c08049ff25cdad5e357727	9 years ago
Richard Mudgett	8cb4f9cea1	CHANNEL(callid): Give dialplan access to the callid. * Added CHANNEL(callid) to retrieve the call identifier log tag associated with the channel. Dialplan now has access to the call log search key associated with the channel so it can be saved in case there is a problem with the call. ASTERISK-26878 Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f	9 years ago
Sean Bright	c13ea6080e	app_queue: Fix locking behavior in stasis message handlers The queue_stasis_data structure contains various mutable fields that require appropriate locking. Specifically, the 'dying,' 'member_uniqueid,' and 'caller_uniqueid' fields need to be locked when read from or written to. Change-Id: I246b7dbff8447acc957a1299f6ad0ebd0fd39088	9 years ago
Sean Bright	15aa3c0a23	chan_sip: Add rtcp-mux support ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	9 years ago
Richard Mudgett	57656e2b5b	app_confbridge: Fix ConfbridgeTalking AMI event description. Thanks to Chris Howard for pointing this out on the wiki. Change-Id: I18e56de09a70e736b5d04719d45ef29cf0636705	9 years ago
Richard Mudgett	82982a191c	res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed. struct ast_rtcp does not define the dtls member if SRTP is not enabled. ASTERISK-26732 Change-Id: Id15ea212e04490e012f2cf4a56818b4dd948875e	9 years ago
Joshua Colp	732367e806	Merge "res_pjsip: Symmetric transports"	9 years ago

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