8297 Commits (082078de3ec52eea6c5246928994b5054e271683)

Author	SHA1	Message	Date
Michael Kuron	60c84bbad5	chan_sip: Only send video on outgoing channel if incoming channel supports it ... Previously, the settings videosupport=always and videosupport=yes behaved identically and unconditionally caused a video offer to be sent in the SDP on an outgoing call. This was a regression introduced with commit 5a1d90e1fb in Asterisk 1.6.1. This commit restores correct behavior: videosupport=always causes a video offer to be sent unconditionally, while videosupport=yes will only offer video on an outbound channel if the incoming channel it is bridged to also supports video. That way, the device receiving the outgoing call can display the correct user interface elements for audio or video and will not unnecessarily show a blank video window on an audio-only call. ASTERISK-17470 #close Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae	9 years ago
Alexander Traud	16d9d67329	chan_sip: Honor support of Symmetric Response (rport) for SIP requests. ... In the SIP channel driver chan_sip, the default is "auto_force_rport". When no NAT was detected, for example in case of IPv6, Asterisk uses the IP address from the headers within the SIP-REGISTER for subsequent SIP signaling. When the remote party specifies support for Symmetric Response (RFC 3581) via the parameter "rport", Asterisk should not extract the port from the SIP headers but reuse the port of the transport. This did not happen because of a typo. ASTERISK-26438 #close Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6	9 years ago
zuul	0b62db30ce	Merge "chan_sip: Resolve externhost not to IPv6; instead go for IPv4." into 14	9 years ago
zuul	f029ba6564	Merge "channels/chan_pjsip: fix HANGUPCAUSE function bug." into 14	9 years ago
Aaron An	7bcb1fc940	channels/chan_pjsip: fix HANGUPCAUSE function bug. ... HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. This patch change the call order of ast_queue_control_data and ast_queue_control in chan_pjsip_incoming_response. ASTERISK-26396 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Ide2d31723d8d425961e985de7de625694580be61	9 years ago
Alexander Traud	175975e994	chan_sip: Resolve externhost not to IPv6; instead go for IPv4. ... For the channel driver chan_sip, you specify externhost=example.com in sip.conf when your Asterisk is behind a NAT and your IP address is assigned dynamically. Or stated differently: You do not have a static IP address to use "externaddr" directly. This NAT support is quite handy but just about IPv4. Previously, Asterisk resolved "externhost" to any IP version. When the first DNS answer resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and connection (c=). This happened in outgoing SIP-REGISTER and while answering SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". ASTERISK-18232 #close Reported by: Jacek Kowalski Tested by: Alexander Traud patches: changes.patch submitted by Alessandro Crespi Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac	9 years ago
George Joseph	d64edafa63	chan_sip: Address runaway when realtime peers subscribe to mailboxes ... Users upgrading from asterisk 13.5 to a later version and who use realtime with peers that have mailboxes were experiencing runaway situations that manifested as a continuous stream of taskprocessor congestion errors, memory leaks and an unresponsive chan_sip. A related issue was that setting rtcachefriends=no NEVER worked in asterisk 13 (since the move to stasis). In 13.5 and earlier, when a peer tried to register, all of the stasis threads would block and chan_sip would again become unresponsive. After 13.5, the runaway would happen. There were a number of causes... * mwi_event_cb was (indirectly) calling build_peer even though calls to mwi_event_cb are often caused by build_peer. * In an effort to prevent chan_sip from being unloaded while messages were still in flight, destroy_mailboxes was calling stasis_unsubscribe_and_join but in some cases waited forever for the final message. * add_peer_mailboxes wasn't properly marking the existing mailboxes on a peer as "keep" so build_peer would always delete them all. * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions then just creating them again. All of this was causing a flood of subscribes and unsubscribes on multiple threads all for the same peer and mailbox. Fixes... * add_peer_mailboxes now marks mailboxes correctly and build_peer only deletes the ones that really are no longer needed by the peer. * add_peer_mwi_subs now only adds subscriptions marked as "new" instead of unsubscribing and resubscribing everything. It also adds the peer object's address to the mailbox instead of its name to the subscription userdata so mwi_event_cb doesn't have to call build_peer. With these changes, with rtcachefriends=yes (the most common setting), there are no leaks, locks, loops or crashes at shutdown. rtcachefriends=no still causes leaks but at least it doesn't lock, loop or crash. Since making rtcachefriends=no work wasnt in scope for this issue, further work will have to be deferred to a separate patch. Side fixes... * The ast_lock_track structure had a member named "thread" which gdb doesn't like since it conflicts with it's "thread" command. That member was renamed to "thread_id". ASTERISK-25468 #close Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0	9 years ago
Steve Davies	58f41ebf93	chan_sip: Fix session timeout on retransmit of non-UDP packets ... Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP connections, allowing the TCP layer to handle the retransmits. Unfortunately, this caused sessions to be terminated with a retransmit timeout becasue it stopped at the point of the first retrans call. This patch waits for the 64*T1 timer to expire instead. ASTERISK-19968 Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204 (cherry picked from commit 98e42cc662)	9 years ago
Alexander Traud	b3802e68b8	chan_sip: Enable Session-Timers for SIP over TCP (and TLS). ... 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 (cherry picked from commit 66c9dfb272)	9 years ago
Walter Doekes	52e6d03401	chan_sip: Allow target refresh (Contact update) on re-INVITE. ... Previously, the Contact was stored only on initial INVITE and on any 18X and 200. That meant that after re-INVITEs from *us* the Contact could get updated, but after re-INVITEs from the *peer*, it did not. This changeset fixes this inconsistency, properly allowing target refreshes through re-INVITES (RFC3261, 12.2). If your strictrtp setting allows it, this change allows you to switch the source IP of a connected/calling device mid-call with a simple re-INVITE from the new IP. ASTERISK-26358 #close Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435	9 years ago
Joshua Colp	d7d22a2be5	chan_sip: Don't allocate new RTP instances on top of old ones. ... In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73	9 years ago
Walter Doekes	9b59ebee2f	chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. ... Certain SNOM phones send so-called "optional crypto" in their SDP body. Regular SRTP setup looks like this: m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... SNOM-style "optional crypto" looks like this: m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... A crypto line is supplied, but the m-line does not have SAVP. When res_srtp.so is *not* loaded, then chan_sip.so treats the optional crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the incoming call with the following message: WARNING: process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio For platforms that want to start providing SRTP this presents a compatibility problem. This changeset lets chan_sip handle the SDP as if no crypto-line was supplied: i.e. accept the call as regular RTP, just like it did before res_srtp was loaded. Now you'll get this informative warning instead: WARNING: Ignoring crypto attribute in SDP because RTP transport is insecure ASTERISK-23989 #close Reported by: Olle Johansson Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2	9 years ago
Kevin Harwell	d2bee6b535	res_format_attr_g729: Add annexb=no format parameter to SDPs ... Historically, Asterisk has always specified annexb=no for the g729 format. However, when using res_pjsip no format attribute was specified. This patch makes it so the SDP now contains a format attribute line with annexb=no. Note, that this means only g729a is negotiated. Even for pass through support. According to rfc7261 the type of annex used (a or b) is dependent upon the answerer. However, Asterisk being a back to back user agent makes this tricky to support at this time, thus we only allow annex 'a' for now. ASTERISK-26228 #close patches: res_format_attr_g729.c submitted by Jason Parker (license 4993) Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0	9 years ago
Corey Farrell	ba91710bc4	Refactor usage pattern of xmldoc info tag. ... This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea	9 years ago
zuul	a50303cfc4	Merge "chan_sip: Fix lastrtprx always updated" into 14	9 years ago
cjack	19e323a6aa	chan_sip: Fix lastrtprx always updated ... Packets are read regulary, when there is no data in buffer fr->frametype is AST_FRAME_NULL. There was no check of frametype and lastrtprx always updated and, therefore, rtptimeout did not work at all. ASTERISK-25270 #close Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d	9 years ago
Matt Jordan	89dccb04bc	func_channel: Reorganize documentation ... * Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b	9 years ago
Matt Jordan	eec60dd773	channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH ... 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	9 years ago
Joshua Colp	804c3e6b9e	Merge "res_srtp: Enable AES-256 and AES-GCM." into 14	9 years ago
zuul	7d17164e6e	Merge "chan_dahdi.c: Fix deadlock potential in fax redirection." into 14	9 years ago
zuul	c1bb8a202c	Merge "chan_sip.c: Fix deadlock potential in fax redirection." into 14	9 years ago
zuul	50a02c4b83	Merge "chan_pjsip.c: Fix deadlock potential in fax redirection." into 14	9 years ago
Joshua Colp	bf5a18f39b	Merge "chan_sip: Prevent deadlock when issuing "sip show channels"" into 14	9 years ago
Richard Mudgett	5efb5b38e8	chan_dahdi.c: Fix deadlock potential in fax redirection. ... 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	9 years ago
Richard Mudgett	a1d36c89e0	chan_sip.c: Fix deadlock potential in fax redirection. ... 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	9 years ago
Richard Mudgett	4dfadcb025	chan_pjsip.c: Fix deadlock potential in fax redirection. ... 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	9 years ago
Richard Mudgett	c03e27c1c8	chan_dahdi: Add faxdetect_timeout option. ... 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	9 years ago
Richard Mudgett	d11731ac2f	res_pjsip: Add fax_detect_timeout endpoint option. ... 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	9 years ago
George Joseph	52cbdf2393	chan_sip: Prevent deadlock when issuing "sip show channels" ... 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	9 years ago
Alexander Traud	2103ad1fec	res_srtp: Enable AES-256 and AES-GCM. ... ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b	9 years ago
Corey Farrell	cf1188a1be	Unit tests: Use AST_TEST_DEFINE in conditional code only. ... 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	9 years ago
Corey Farrell	f73ddde7d4	chan_sip: Fix reference leak in mwi_event_cb ... 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	9 years ago
Corey Farrell	ad30d60c69	chan_sip: Fix reference leaks in error paths. ... * 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	9 years ago
Joshua Colp	302be4809a	chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. ... 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	9 years ago
Richard Mudgett	656ed73ac6	res_pjsip: Add missing NULL checks when using pjsip_inv_end_session(). ... pjsip_inv_end_session() is documented as being able to return the passed in tdata parameter set to NULL on success. Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047	9 years ago
zuul	ba872766fa	Merge "siren: Add format attribute modules for Siren7 and Siren14."	9 years ago
George Joseph	651290a809	BuildSystem: Fix a few issues hightlighted by gcc 6.x ... 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	9 years ago
Joshua Colp	75818b4084	siren: Add format attribute modules for Siren7 and Siren14. ... 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	9 years ago
George Joseph	c7309a5254	chan_unistim: Fix memcpy in get_to_address ... 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	9 years ago
zuul	947f76a971	Merge "chan_sip: bigger buffers for headers, better failure mode"	10 years ago
zuul	88dfcd21b2	Merge "chan_sip: Support auth username for callbackextension feature"	10 years ago
zuul	0388c40b8c	Merge "chan_pjsip: Lock channel when checking for RTP changes."	10 years ago
Mark Michelson	cdb7edbe7b	chan_pjsip: Lock channel when checking for RTP changes. ... 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	10 years ago
George Joseph	d21a77b325	build: Fix ast_sockaddr initialization to be more portable ... 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	10 years ago
Joshua Colp	2525563438	Merge "chan_sip: No rtpmap for static RTP payload IDs in SDP."	10 years ago
Joshua Colp	5c949d009e	Merge "Fixes to include signal.h"	10 years ago
Timo Teräs	39b69ab537	Fixes to include signal.h ... 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	10 years ago
Alexander Traud	784c18128b	chan_sip: No rtpmap for static RTP payload IDs in SDP. ... 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	10 years ago
Vasil Kolev	80ff7912a1	chan_sip: bigger buffers for headers, better failure mode ... 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	10 years ago
Richard Mudgett	dca052e531	chan_rtp.c: Simplify options to UnicastRTP channel creation. ... 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	10 years ago