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2016-08-10channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESHMatt Jordan
This patch adds a new PJSIP specific dialplan function, PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media session will be refreshed via either an UPDATE or re-INVITE request. When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, the formats in use on a PJSIP channel can be re-negotiated and changed dynamically after call setup. ASTERISK-26277 #close Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b (cherry picked from commit eec60dd77394f0519895fc6abce3a6f90f6470f1)
2016-07-22Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."zuul
2016-07-22chan_sip: Enable Session-Timers for SIP over TCP (and TLS).Alexander Traud
Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957
2016-07-22Merge "chan_sip: Prevent deadlock when issuing "sip show channels""zuul
2016-07-21Merge "res_srtp: Enable AES-256 and AES-GCM."zuul
2016-07-21Merge "chan_dahdi.c: Fix deadlock potential in fax redirection."zuul
2016-07-21Merge "chan_sip.c: Fix deadlock potential in fax redirection."zuul
2016-07-21Merge "chan_pjsip.c: Fix deadlock potential in fax redirection."zuul
2016-07-21Merge "chan_dahdi: Add faxdetect_timeout option."Joshua Colp
2016-07-21Merge "res_pjsip: Add fax_detect_timeout endpoint option."Joshua Colp
2016-07-21chan_sip: Prevent deadlock when issuing "sip show channels"George Joseph
sip_show_channels locks the dialogs container first then locks each sip_pvt so it can spit out the details. The rest of sip dialog processing locks the sip_pvt first then locks the dialogs container if it needs to. Both lock in the order they need but deadlocks can result. To fix, sip_show_channels and sip_show_channelstats have been converted to use an iterator rather than ao2_callback. This way the container is locked only while getting the next entry and is unlocked when the callback is called. ASTERISK-23013 #close Change-Id: Id9980419909e811f89484950ed46ef117b9eb990
2016-07-21res_srtp: Enable AES-256 and AES-GCM.Alexander Traud
ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b
2016-07-19chan_dahdi.c: Fix deadlock potential in fax redirection.Richard Mudgett
The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. ASTERISK-26216 #close Reported by: Richard Mudgett Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa
2016-07-19chan_sip.c: Fix deadlock potential in fax redirection.Richard Mudgett
The sip_read() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e
2016-07-19chan_pjsip.c: Fix deadlock potential in fax redirection.Richard Mudgett
The chan_pjsip_cng_tone_detected() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5
2016-07-19chan_dahdi: Add faxdetect_timeout option.Richard Mudgett
The new option allows the channel driver's faxdetect 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. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810
2016-07-19res_pjsip: Add fax_detect_timeout endpoint option.Richard Mudgett
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
2016-07-18Unit tests: Use AST_TEST_DEFINE in conditional code only.Corey Farrell
If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead code. This places all existing unit tests into a conditional block if they weren't already. ASTERISK-26211 #close Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686
2016-07-13chan_sip: Fix reference leak in mwi_event_cbCorey Farrell
Cleanup the peer reference when stasis_subscription_final_message is true. Also free peer_name even if peer exists, after reload a new peer_name will be allocated. ASTERISK-26193 #close Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69
2016-07-09chan_sip: Fix reference leaks in error paths.Corey Farrell
* get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error. * build_peer leaks peer on failure to allocate the endpoint. This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed with an unref in the appropriate place. ASTERISK-26184 #close Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12
2016-07-07chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled.Joshua Colp
Some T.38 implementations may send another re-invite after the initial one which adds additional negotiation details (such as the max bitrate). Currently this will fail when passthrough is being done in chan_sip as we do nothing if T.38 is already active. Other handlers of T.38 inside of Asterisk (such as res_fax) handle this scenario so this change adds support for it to chan_sip and res_pjsip_t38. If a request to negotiate is received while T.38 is already enabled a new re-INVITE is sent and negotiation is done again. ASTERISK-26179 #close Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c
2016-06-30res_pjsip: Add missing NULL checks when using pjsip_inv_end_session().Richard Mudgett
pjsip_inv_end_session() is documented as being able to return the passed in tdata parameter set to NULL on success. Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047
2016-06-29Merge "siren: Add format attribute modules for Siren7 and Siren14."zuul
2016-06-28BuildSystem: Fix a few issues hightlighted by gcc 6.xGeorge Joseph
gcc 6.1.1 caught a few more issues. Made sure the unit tests still pass for the func_env and stdtime issues. ASTERISK-26157 #close Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e
2016-06-23siren: Add format attribute modules for Siren7 and Siren14.Joshua Colp
This change removes hardcoded SDP parsing and generation for Siren7 and Siren14 from chan_sip and moves it to format attribute modules so it can also be used by chan_pjsip. With this the fmtp lines for both are added with the bitrate information. ASTERISK-26021 Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037
2016-06-22chan_unistim: Fix memcpy in get_to_addressGeorge Joseph
A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD) was using a pointer to a pointer as the destination of a memcpy and a '&' instead of '*' in the sizeof. ASTERISK-26138 #close Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708
2016-06-16Merge "chan_sip: bigger buffers for headers, better failure mode"zuul
2016-06-09Merge "chan_sip: Support auth username for callbackextension feature"zuul
2016-06-09Merge "chan_pjsip: Lock channel when checking for RTP changes."zuul
2016-06-09chan_pjsip: Lock channel when checking for RTP changes.Mark Michelson
bridge_native_rtp can call into an RTP-capable channel driver in order for the driver to update information about who the channel is communicating with. For SIP channel drivers, this means deactivating RTCP and sending a reinvite so that the endpoints can communicate directly. bridge_native_rtp does the right thing and has the channel locked when calling into the channel driver. chan_pjsip can't alter session properties in this thread, though. chan_pjsip queues a task on the session serializer in order to update properties there. The problem is that this queued task was not locking the channel. This meant that the queued task could attempt to deactivate RTCP at the same time that the channel thread was attempting to process an incoming RTCP packet. This could lead to a crash. This patch fixes the issue by locking the channel in the queued task when altering RTP properties. ASTERISK-26092 #close Reported by Niklas Larsson Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159
2016-06-09build: Fix ast_sockaddr initialization to be more portableGeorge Joseph
A change to glibc 2.22 changed the order of the sockadddr_storage members which caused the places where we do an initialization of ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those initializers (which we shouldn't have been using anyway) have been replaced with memsets. Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4
2016-06-09Merge "chan_sip: No rtpmap for static RTP payload IDs in SDP."Joshua Colp
2016-06-09Merge "Fixes to include signal.h"Joshua Colp
2016-06-08Fixes to include signal.hTimo Teräs
POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc
2016-06-08chan_sip: No rtpmap for static RTP payload IDs in SDP.Alexander Traud
This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over UDP, if many codecs are allowed in Asterisk. This new feature is enabled together with the optional feature compactheaders=yes via the file sip.conf. ASTERISK-25578 #close Change-Id: I16491b1937862de26f84fa0ffe679a6bab925044
2016-06-07chan_sip: bigger buffers for headers, better failure modeVasil Kolev
Currently chan_sip can give weird messages if the contacts don't fit in the From: or To: headers. This fix changes the from,to and invite variables to use ast_str, allocates and deallocates them and resizes them if needed. ASTERISK-26069 #close Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3
2016-06-06chan_rtp.c: Simplify options to UnicastRTP channel creation.Richard Mudgett
Change the awkward and not as flexible UnicastRTP options format From: Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]]) To: Dial(UnicastRTP/127.0.0.1[/[<options>]]) Where <options> can be standard Asterisk flag options: c(<codec>) - Specify which codec/format to use such as 'ulaw'. e(<engine>) - Specify which RTP engine to use such as 'asterisk'. More option flags can be easily added later such as the codec's RTP payload type to use when the codec does not have a static payload type defined. Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9
2016-06-03chan_sip: Support auth username for callbackextension featureTimo Teräs
ASTERISK-20527 #close Change-Id: I659cf7f00836a09d09d146ad226a40477d731239
2016-05-27multicast RTP: Add dialing optionsMark Michelson
This adds a new parameter to the end of a multicast RTP dialing string. This parameter defines the following options: * i: Set the interface from which multicast RTP is sent * l: Set whether multicast packets are looped back to the sender * t: Set the TTL for multicast packets * c: Set the codec to use for RTP ASTERISK-26068 #close Reported by Mark Michelson Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
2016-05-20chan_rtp.c: Cleanup ast_request() parameter parsing.Richard Mudgett
* Fixed NULL crash potential if parameters are missing. * Reordered some operations so further diagnostic messages can be more helpful. Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70
2016-05-17chan_sip: Prevent extra Session-Expires headers from being addedGeorge Joseph
When chan_sip does a re-INVITE to refresh a session and authentication is required, the INVITE with the Authorization header containes a second Session-Expires header without the ";refersher=" parameter. This is causing some proxies to return a 400. Also, when Asterisk is the uas and the refresher, it is including the Session-Expires and Min-SE headers in OPTIONS messages which is not allowed per RFC4028. This patch (based on the reporter's) Checks to see if a Session-Expires header is already in the message before adding another one. It also checks that the method is INVITE or UPDATE. ASTERISK-26030 #close Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9
2016-04-29Merge "chan_sip: Make autocreated peers send PeerStatus events"Joshua Colp
2016-04-26chan_sip: Give more time for TCP/TLS threads to stop.Joshua Colp
The unload process currently tells each TCP/TLS to terminate but does not wait for them to do so. This introduces a race condition where the container holding the threads may be destroyed before the threads are able to remove themselves from it. When they finally do the container is invalid and can't be used causing a crash. A previous change existed which waited a bit to wait for any stranglers to finish. This change extends this and waits longer. ASTERISK-25961 #close Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6
2016-04-22chan_sip: Make autocreated peers send PeerStatus eventskkm
Since Stasis has been introduced, an attempt to send AMI messages by an autocreated peer caused a crash, and all events from autocreated peers were semi-inadvertently disabled altogether in 0b83761. This change restores the disabled functionality. ASTERISK-25950 Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974
2016-04-18chan_sip: Don't verify table if rtupdate=noJaco Kroon
If rtupdate=no do not verify sipregs/peers table has updatable fields. ASTERISK-25934 #close Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d
2016-04-07pbx.h: Make ast_state_cb_type take more const.Richard Mudgett
This eliminates some casts that I made a note saying v10 and above would no longer need them. Better late than never :) Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572
2016-03-31Merge "chan_sip: Do not send all codecs on INVITE. Do not break on ↵zuul
Session-Timers."
2016-03-29chan_pjsip: Add 'pjsip show channelstats'George Joseph
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
2016-03-24chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers.Alexander Traud
Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those codecs, which the caller did not request/support. That fix was not complete because on the second Session Timer all codecs were sent again. Some VoIP/SIP clients interpreted that complete codec-list as a change in the SIP session. Because of that, Asterisk did not send the RTP audio via NAT anymore which created a non-audio scenario after the second Session Timer fired. ASTERISK-24543 #close Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66
2016-03-22chan_sip.c: Space after port causes unnecessary resolution attemptFrancesco Castellano
check_via() already skips leading blanks where the sent-by address (with the optional port) should be placed. Since RFC 3261 allows for blanks between the port ant the Via parameters: > https://tools.ietf.org/html/rfc3261#section-20.42 (actually it allows a lot of blanks more ;-)). I just switched from ast_skip_blanks() to ast_strip() on the local copy of the string. ASTERISK-21301 #close Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06
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