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2017-03-24Merge "rtp_engine: allocate RTP dynamic payloads per session"zuul
2017-03-24res_xmpp: Use incremental backoff when a read error occursSean Bright
If a read error occurs, we immediately attempt a reconnect without any delay. Instead, let's sleep and backoff up to 60 seconds before we try again. ASTERISK-24712 #close Reported by: Matthias Urlichs Change-Id: I6fe10ef4734837727437beab715e336777f13f48
2017-03-24Merge "pjproject_bundled: raise timeout value used when downloading"zuul
2017-03-24Merge "res_xmpp: Correct implementation of JABBER_STATUS & JabberStatus"zuul
2017-03-24Merge "res_xmpp: Include client name in connection related error messages"zuul
2017-03-24res_pjsip_sdp_rtp: Set hangup cause for RTP timeoutsSean Bright
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
2017-03-24Merge "res_xmpp: Don't crash when trying to send a message without a connection"Joshua Colp
2017-03-24Merge "res_xmpp: Correctly check return value of SSL_connect"zuul
2017-03-24Merge "res_xmpp: Try to provide useful errors messages from OpenSSL"zuul
2017-03-24Merge "audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor."Joshua Colp
2017-03-23AMI: Updated versionKevin Harwell
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
2017-03-23pjproject_bundled: raise timeout value used when downloadingKevin Harwell
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
2017-03-23res_xmpp: Correct implementation of JABBER_STATUS & JabberStatusSean Bright
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
2017-03-23res_xmpp: Try to provide useful errors messages from OpenSSLSean Bright
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
2017-03-23res_xmpp: Fix ref counting issueSean Bright
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
2017-03-23res_xmpp: Correctly check return value of SSL_connectSean Bright
SSL_connect returns non-zero for both success and some error conditions so simply negating is inadequate. Change-Id: Ifbf882896e598703b6c615407fa456d3199f95b1
2017-03-23res_xmpp: Don't crash when trying to send a message without a connectionSean Bright
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
2017-03-23res_xmpp: Include client name in connection related error messagesSean Bright
ASTERISK-25622 #close Reported by: Sean Darcy Change-Id: I8472cb7bfb58d411a3cfbd482da98cae2d94d1e9
2017-03-22Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."Joshua Colp
2017-03-22Merge "CHANNEL(callid): Give dialplan access to the callid."Joshua Colp
2017-03-22rtp_engine: allocate RTP dynamic payloads per sessionKevin Harwell
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
2017-03-22Merge "res_pjsip_messaging: Check URI type before dereferencing"zuul
2017-03-22Merge "Revert "app_queue: Handle the caller being redirected out of a queue ↵zuul
bridge""
2017-03-22Merge "app_queue: Member stuck as pending after forwarding previous call ↵zuul
from queue"
2017-03-22cdr: Allow setting of user field from 'h' extensionSebastian Gutierrez
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
2017-03-22Merge "pjsip: prevent memory corruption on creation of xml bodies"zuul
2017-03-22res_pjsip_session: Enable RFC3578 overlap dialing support.Richard Begg
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
2017-03-21Merge "autochan/mixmonitor/chanspy: Fix unsafe channel locking and references."zuul
2017-03-21Merge "res_hep: Capture actual transport type in use"zuul
2017-03-21res_hep: Capture actual transport type in useSean Bright
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
2017-03-21Revert "app_queue: Handle the caller being redirected out of a queue bridge"Sean Bright
This reverts commit 163e9e53dc7d84dd42721e733b7706c8147bdd27. Change-Id: Ief28479c77a298879dfe2c56be7ee92dc465da4b
2017-03-21res_pjsip_messaging: Check URI type before dereferencingSean Bright
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
2017-03-21pjsip: prevent memory corruption on creation of xml bodiesJoshua Elson
ASTERISK-26776 #close Change-Id: I884b6f4e8233a355d0be687ec78d41bc0e4d3fd2
2017-03-20bridge_softmix: Ignore non-voice frames from translatorSean Bright
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
2017-03-20Merge "res/res_pjsip_session: Only check localnet if it is defined"Joshua Colp
2017-03-20audiohook.c: Lost RTP packets lead to out-of-sync MixMonitor.Aaron An
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
2017-03-20Merge "thread safety: Don't use getprotobyname()"zuul
2017-03-20thread safety: Don't use getprotobyname()Sean Bright
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
2017-03-19res_rtp_asterisk: Pass correct data length to ast_rtcp_interpretSean Bright
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
2017-03-18Merge "app_queue: Fix locking behavior in stasis message handlers"Joshua Colp
2017-03-18Merge "chan_sip: Add rtcp-mux support"Joshua Colp
2017-03-18Merge "res_rtp_asterisk: Fix crash when RTCP is not present when DTLS is ↵Joshua Colp
stopped."
2017-03-18Merge "res_pjsip_asterisk.c: Fix compile error if libsrtp is not installed."Joshua Colp
2017-03-17Merge "app_confbridge: Fix ConfbridgeTalking AMI event description."Joshua Colp
2017-03-17Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error"Joshua Colp
2017-03-17Merge "res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit ↵Joshua Colp
transport"
2017-03-17app_queue: Member stuck as pending after forwarding previous call from queueRobert Mordec
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
2017-03-17CHANNEL(callid): Give dialplan access to the callid.Richard Mudgett
* 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
2017-03-17app_queue: Fix locking behavior in stasis message handlersSean Bright
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
2017-03-17chan_sip: Add rtcp-mux supportSean Bright
ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639
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