7459 Commits (9b92042c17ca31e46b82b2ebf08ff75a28e6cea0)

Author	SHA1	Message	Date
Richard Mudgett	7153c2c2c5	chan_sip: Replace sip_tls_read() and resolve the large SDP poll issue. ... Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. ASTERISK-18345 #close Reported by: Stephane Chazelas Patches: tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad Review: https://reviewboard.asterisk.org/r/3882/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@420434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	0789fefda6	chan_sip: sip_subscribe_mwi_destroy should not call sip_destroy ... sip_subscribe_mwi_destroy calls sip_destroy on the reference counted mwi->call. This results in the fields of mwi->call being freed, but mwi->call itself it leaked. If other code is still using mwi->call it can cause problems. This change uses dialog_unref instead, to balance the ref provided by sip_alloc(). ASTERISK-24087 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3834/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@419440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4542181b8a	chan_dahdi/sig_pri: Fix type mismatch in the idledial feature's channel creation. ... Square pegs in round holes don't work very well. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@418261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	863e7e528c	chan_dahdi: Add inband_on_setup_ack compatibility option. ... The new inband_on_setup_ack option causes Asterisk to assume inband audio may be present when a SETUP_ACKNOWLEDGE message is received. Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a dialtone is sent from the network side, progress indicator 8 "Inband info now available" MAY be sent to the CPE if no digits were received with the SETUP. It is thus implied that the ie is mandatory if digits came with the SETUP and dialtone is needed. This option should be enabled, when the network sends dialtone and you want to hear it, but the network doesn't send the progress indicator when needed. NOTE: For Q.SIG setups this option should be enabled when outgoing overlap dialing is also enabled because Q.SIG does not send the progress indicator with the SETUP ACK. The commit -r413714 (AST-1338) which causes this issue was dealing with a SIP-to-ISDN interoperability issue. This commit is a merge of the two patches indicated below. ASTERISK-23897 #close Reported by: Pavel Troller Patches: pri-4.diff (license #6302) patch uploaded by Pavel Troller jira_asterisk_23897_v11.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3633/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	67a11896e7	chan_sip: be more tolerant of whitespace between attributes in SDP fmtp line ... This patch is essentially a backport of a small portion of r397526 from ASTERISK-21981. In that patch, pass through support and format attribute negotiation was added for Opus. Part of that included being more tolerant to whitespace in the fmtp line of an SDP; that part of the patch is being applied here. As the author of the backport pointed out, in SDP, the fmtp line is allowed to include whitespace between attributes. RFC 3267 chapter 8.3 (from 2001) includes an example for this. This was not removed in the updated RFC 4867 in 2007. Note that this patch only applies to audio in Asterisk 1.8, which is a bit more limited in its support for format attributes. It does have limited support for some codecs, so this patch is still useful in this version. Review: https://reviewboard.asterisk.org/r/3658 ASTERISK-23916 Reported by: Alexander Traud patches: sdpFMTPspace_Asterisk11.patch uploaded by Alexander Traud (License 6520) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	c56504a148	chan_sip: Fix handling of "From" headers longer than 256 characters ... From headers were processed using a 256 character buffer on the stack. This change replaces that with a heap allocation by ast_strdup. ASTERISK-23790 #close Reported by: uniken1 Tested by: uniken1 Review: https://reviewboard.asterisk.org/r/3669/ Patches: chan_sip-large-from-header-1.8-r3.patch uploaded by wdoekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@417248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7752d56964	Fix build warnings with TEST_FRAMEWORK enabled ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416732 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1509ae5779	AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. ... ASTERISK-23673 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3617/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@416066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	883d2e6924	chan_sip: DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero ... Change debug level for messages in sdp_crypto.c from zero to one. This ensures the messages are not displayed when debugging is disabled. Change does not apply to 12+ as it was already fixed in those versions. ASTERISK-23246 #close Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/3605/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7550fe7efb	AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. ... Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	560380a018	chan_sip: Fix order of variables specified in SIPNotify action ... Prior to this patch, sequential variables would be ordered in reverse from the order specified in the manager action. Review: https://reviewboard.asterisk.org/r/3588/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@415359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	bf94ff5763	chan_sip: Start session timer at 200, not at INVITE. ... Asterisk started counting the session timer at INVITE while the other end correctly started at 200. This meant that for short session-expiries (90 seconds) combined with long ringing times (e.g. 30 seconds), asterisk would wrongly assume that the timer was hit before the other end thought it was time to send a session refresh. This resulted in prematurely ended calls. This changes the session timer to start counting first at 200 like RFC says it should. (Also removed a few excess NULL checks that would never hit, because if they did, asterisk would have crashed already.) ASTERISK-22551 #close Reported by: i2045 Review: https://reviewboard.asterisk.org/r/3562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	71df5a4a63	Backport Asterisk 11 r413876 to 1.8 ... ........ r413876 | jrose | 2014-05-13 12:40:00 -0500 (Tue, 13 May 2014) | 6 lines chan_sip: Add TLS and SRTP status to CLI command 'sip show channel' ASTERISK-23564 #close Reported by: Patrick Laimbock Review: https://reviewboard.asterisk.org/r/3474/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	46aff9f0f8	chan_local: Only block media frames when a generator is on both ends of a local channel. ... The fix for ASTERISK-12292 was a bit too aggressive. You could have generators pointed at each other on local channels but need to get other kinds of frames such as DTMF or CONNECTED_LINE frames accross. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ffe97a9c8f	chan_dahdi: Fix analog dialtone detection. ... * Check if waitingfordt (waitfordialtone) is enabled in dahdi_read() to allow the DSP to operate early enough to detect dialtone. * Made use the correct variable in my_check_waitingfordt(). ASTERISK-23709 #close Reported by: Steve Davies Patches: dialtone_detect_fix (license #5012) patch uploaded by Steve Davies Review: https://reviewboard.asterisk.org/r/3534/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@414067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8bf18ed4d3	chan_local+app_dial: Propagagate call answered elsewhere over local channels. ... AST_FLAG_ANSWERED_ELSEWHERE was not propagated back from local channels. It is now. That means that when a call is picked up from a callgroup of local channels, the other channels will now properly see it as "picked up". This occurs when you use a construct like Dial(Local/a@context&Local/b@context) where a@context and b@context dial two chan_sip devices respectively. If one device picks up, the other will not see "1 missed call" anymore. In this respect, it now behaves the same as when doing Dial(SIP/a&SIP/b). Review: https://reviewboard.asterisk.org/r/3540/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	adb0ee2dcd	chan_sip+CEL: Add missing ANSWER and PICKUP events to INVITE/w/replaces pickup. ... When doing a "BLF-style call pickup" -- an INVITE with Replaces: header -- the CEL log would lack the ANSWER and PICKUP events. This patch adds the two missing events to the handle_invite_replaces() function. ASTERISK-22977 #close Review: https://reviewboard.asterisk.org/r/3073/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	3dace58a49	chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing is enabled. ... When overlap dialing is enabled, the lack of inband audio available information in the SETUP_ACKNOWLEDGE events causes an interoperability problem with SIP. sig_pri doesn't know if there is dialtone present when a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183 Session Progress and blocks the desired 180 Ringing message when the ALERTING message comes in. * Made the configure script detect if the installed version of libpri supports the SETUP_ACKNOWLEDGE enhancements. * Using the new API, made generate an AST_CONTROL_PROGRESS frame on an incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio is present instead of assuming that dialtone is present. * Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio available indication only if dialtone is expected. The change also makes the fallback behaviour of sending the PROGRESS message better by sending it only if dialtone is expected. * Changed receiving a PROCEEDING message to not generate an AST_CONTROL_PROGRESS frame if the progress indication ie indicates non-end-to-end-ISDN. This helps interoperability with SIP. * Changed sending a PROCEEDING message in response to an AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It was silly to do so anyway because the channel driver doesn't know if inband audio is even available. This helps interoperability with SIP. This patch and a corresponding change in libpri work together to allow Asterisk to control the inband audio available progress indication ie on the SETUP_ACKNOWLEDGE message when dialtone is present. AST-1338 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3521/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	650b948d21	Fix 32bit build for chan_sip ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413591 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	3e9a54d857	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@413586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	5d924bc0fc	chan_sip: Add sendrpid trust options ... In r411189, some behavior was changed which made sendrpid behavior act in a more trusting manner by sending full user data for peers set with private caller presence in P-Asserted-Identity headers. Since this changed long time expected behaviors, we decided to pull that patch when that was pointed out by the community. Instead, this patch provides a trust_id_outbound setting which will expose the data per RFC-3325 if set to 'yes' and simply not send the PAI/RPID headers at all if set to 'no'. By default trust_id_outbound will be set to 'legacy' which will preserve the behavior prior to these patches. Extra special thanks to Walter Doekes for providing advice and feedback. (closes issue AST-1301) (closes issue ASTERISK-19465) Reported by: Krzysztof Chmielewski Review: https://reviewboard.asterisk.org/r/3447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	03ed8d30ba	channels/chan_oss: Fix compilation problem on SmartOS/Illumos/SunOS ... THis patch fixes an issue in chan_oss when building on certain platforms. It ensures that soundcard.h is found. Review: https://reviewboard.asterisk.org/r/3426 Note that this patch is a part of the patch on ASTERISK-23576; the Makefile portion only applies to Asterisk 11+. (issue ASTERISK-23576) Reported by: Sebastian Wiedenroth patches: fix-sunos.diff uploaded by Sebastian Wiedenroth (License 6597) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	445b1ced10	Reverting r411189 so that it can be put up for public review ... --- r411189 | jrose | 2014-03-26 10:50:48 -0500 (Wed, 26 Mar 2014) | 12 lines chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior to this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) --- git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	751dc2b7da	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output ... This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. Review: https://reviewboard.asterisk.org/r/3377/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@412114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a997ddc829	internal_timing: Remove the option and always make it enabled if a timing module is loaded. ... The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	588550c950	chan_sip: Add MESSAGE request to allowed methods ... The allowed methods advertised by chan_sip did not previously note the MESSAGE request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we should advertise that we support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504 #comment Reported by: Martin Kontsek ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587) Review: https://reviewboard.asterisk.org/r/3396/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	824c8d4b6b	Fix dialplan function NULL channel safety issues ... (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	e2708b03b6	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) ... Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7ae2541b15	chan_sip: Fix incorrect use of timers ... If update_provisional_keepalive() is called while send_provisional_keepalive_full() is waiting on the PVT lock, then pvt->provisional_keepalive_sched_id will be changed to a new sched_id value by update_provisional_keepalive(), but that new sched_id then may be overwritten with -1 by send_provisional_keepalive_full(), killing the pvt's reference to a schedule and "leaking" the reference. (closes issue ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies Patches: provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	9c7ed786e5	chan_sip: Always use fromdomain if set for domain, even if callerid is set to restricted. ... (closes issue ASTERISK-20841) Reported by: Kelly Goedert git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@411021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b0df15e1bf	AST-2014-002: chan_sip: Exit early on bad session timers request ... This change allows chan_sip to avoid creation of the channel and consumption of associated file descriptors altogether if the inbound request is going to be rejected anyway. (closes issue ASTERISK-23373) Reported by: Corey Farrell Patches: chan_sip-earlier-st-1.8.patch uploaded by Corey Farrell (license 5909) chan_sip-earlier-st-11.patch uploaded by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410308 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	dbfbcbb1c7	chan_sip: Fix deadlock of monlock between unload_module and do_monitor ... Release monlock before calling pthread_join. This ensures do_monitor cannot freeze by locking monlock during module unload. (closes issue ASTERISK-21406) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3284/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	78089aa1cf	chan_sip: Allow static realtime members to be qualified during module load. ... When a static realtime peer with qualify=yes is loaded, Asterisk will fail to send an OPTIONS request due to the lastms being equal to 0. This results in the peer being unable to receive calls from Asterisk because the status is permanently UNKNOWN. This patch allows an OPTIONS request to be sent during module load by ignoring the lastms value on startup only. Review: https://reviewboard.asterisk.org/r/3294/ (closes issue ASTERISK-17523) Reported by: Maciej Krajewski Tested by: wushumasters patches: realtime_fix_11.7.0.txt uploaded by Trevor Peirce (license 6112) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@410105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	28b726e79b	Add update_peer function to unistim_rtp_glue, improve other unistim_rtp_glue functions conforming to other channel drivers. Do not forget auto-detected and user-selected phone settings on 'unistim reload' ...	11 years ago
Sean Bright	2942d6fdac	Minor whitespace change to 'sip show peers' output. ... (closes issue ASTERISK-23406) Reported by: ibercom Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5e47d100d1	chan_sip: Add precautionary p->owner checks. ... * Add precautionary p->owner checks in sip_hangup(), get_refer_info(), get_also_info(), and interpret_t38_parameters(). * Simplify some tangled logic in get_refer_info(), get_also_info(), and add_rpid(). * Removed some dead code in handle_request_invite(). (closes issue ASTERISK-23323) Reported by: Walter Doekes Patches: issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5f27cde1ea	chan_sip: Fix crash in ast_channel_hangupcause_set(). ... * Fix crash in ast_channel_hangupcause_set() because p->owner not checked before calling. Regression introduced by the fix for ASTERISK-22621. (closes issue ASTERISK-23135) Reported by: OK (issue ASTERISK-23323) Reported by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@409156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	2197b1d114	channel.c: MOH is not working for transferee after attended transfer ... Updated the code to check to see if MOH is playing on the transferor and if so then start it on the channel that replaces it during a masquerade. Example scenario of the problem: Alice calls Bob and then Bob begins the attended transfer process into a queue. Upon going on hold Alice hears music and so does Bob once he is in the queue. Bob then transfers Alice into the queue and then music for Alice stops even though she should be hearing it since has now replaced Bob in the queue. The problem that was occurring is that once the channel was masqueraded the app (queues, confbridge, etc...) had no way of knowing that the channel had just been swapped out thus it did not start music for the present channel. Credit to Olle Johansson for pointing me in the right direction on this issue. (closes issue ASTERISK-19499) Reported by: Timo Teräs Review: https://reviewboard.asterisk.org/r/3226/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@408642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	8ee947ed5e	chan_dahdi: handle DAHDI_EVENT_REMOVED on a pri D-Channel ... When a DAHDI device is removed at run-time it sends the event DAHDI_EVENT_REMOVED on each channel. This is intended to signal the userspace program to close the respective file handle, as the driver of the device will need all of them closed to properly clean-up. This event has long since been handled in chan_dahdi (chan_zap at the time). However the event that is sent on a D-Channel of a "PRI" (ISDN) span simply gets ignored. This commit adds handling for closing the file descriptor (and shutting down the span, while we're at it). It also adds a CLI command 'pri destroy span <N>' to destroy the span and its DAHDI channels. Backported from trunk/12. Review: https://reviewboard.asterisk.org/r/726/ ........ Merged revisions 394552 394567 from http://svn.asterisk.org/svn/asterisk/trunk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407817 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	414d336f45	chan_iax2: Add some more iaxs[] NULL checks to a routine already full of them. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6e6a7f2af0	chan_iax2: Block unnecessary control frames to/from the wire. ... Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later) results in an unexpected call disconnect. The problem happens because newer values in the enum ast_control_frame_type are not consistent between the branch versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) using IAX2 2) v1.8 answers and sends a connected line update control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 receives the control frame as an end-of-q (on v1.4 AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the receive queue becomes empty. Several things are done by this patch to fix the problem and attempt to prevent it from happening again in the future: * Added a warning at the definition of enum ast_control_frame_type about how to add new control frame values. * Made block sending and receiving control frames that have no reason to go over the wire. * Extended the connectedline iax.conf parameter to also include the redirecting information updates. * Updated the connectedline iax.conf parameter documentation to include a notice that the parameter must be "no" when the peer is an Asterisk v1.4 instance. (closes issue AST-1302) Review: https://reviewboard.asterisk.org/r/3174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@407678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	1dd94a2e19	chan_sip: Decline image streams on unsupported transports ... This change allows chan_sip to decline individual image streams over unsupported transports in the SDP of the 200 response. Previously, an image stream offer with RTP/AVP as the transport would cause chan_sip to respond with a 488. (closes issue ASTERISK-22988) Reported by: adomjan Original patch by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406170 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4d70fa4750	chan_mgcp: Enforce locking for oseq ... This restricts direct usage of global oseq so that all accesses are locked and threads are not racing to get oseq values that they did not claim. This also fixes a build error in res_pktccops under dev mode. (closes issue ASTERISK-23100) Reported by: adomjan Patch by: adomjan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@406037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f7fe7d533f	chan_dahdi/PRI: Suppress CONNECTED_LINE updates when nothing in the udpate is valid. ... * Also simplified some subddress handling code. (closes issue ASTERISK-23008) Reported by: Michael Cargile git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	66cce0927e	Documentation: doc fixes across various parts of the code for ASTERISK issues 23061,23028,23046,23027 ... Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	097acbdd2c	chan_sip: No BYE message sent after INVITE with Replaces ... Setting channel state DOWN is an unnecessary step that was only being done in handle_invite_replaces(). This changes that by removing the call and reducing locking. (closes issue ASTERISK-23010) Reported by: Ryan Tilton Review: https://reviewboard.asterisk.org/r/3116/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	e2cf92fd96	chan_sip: fix Local From tag on outbound register regression ... In ASTERISK-12117, an improvement to insure consistant local from tags on outbound registrations resulted in an undesirable behavior - caused by leftover unexpired sip_pvt dialogs (with the previous cseq number), resulting in many uncessary REGISTER requests. Instead of significant rework of transmit_register(), this change deletes the dialogs after a 200 OK response indiciating a successful registration, keeping the old dialogs from interfering with normal operation. (closes issue ASTERISK-22946) Reported by: Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/3109/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ad821983f8	chan_sip: Hangup transferer/transferee when transfer to Parking fails ... When performing a SIP transfer to a Park extension, if the Park fails, chan_sip will currently not hang up either the transferer or the transfer target. This results in the channels being orphaned with no thread to service frames, resulting in stuck channels. This patch immediately hangs up the two channels if a Park fails. (closes issue ASTERISK-22834) Reported by: rsw686 (closes issue ASTERISK-23047) Reported by: Tommy Thompson Review: https://reviewboard.asterisk.org/r/3107 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	745c1a623e	Add the missing part of r400140 ... When the patch to add retry-on-forbidden-response was committed, part of the patch for chan_sip was not committed which caused the feature to be entirely nonfunctional. This corrects the code in question. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@405033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	b3094e976a	chan_dahdi: dahdi show channels slices PRI channel dnid on output ... dahdi show channels output slices the callerid (which is dnid copied over on PRI channels). If the channel naming structures look like: 'DAHDI/i1/1408409XXXX-6' then the output slices 1408409XXXX down to 1408409XXX. This patch just opens it up to 15 chars so you can see the whole thing. (closes issue ASTERISK-22918) Reported by: outtolunc Patches: svn_chan_dahdi.c.format12_15.diff.txt uploaded by outtolunc (license 5198) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@404784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago