asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	56a07fbab9	menuselect: Various menuselect enhancements * Add 'external' as a support level. * Add ability for module directories to add entries to the menu by adding members to the <module_prefix>/<module_prefix>.xml file. * Expand the description field to 3 lines in the ncurses implementation. * Allow the description field to wrap in the newt implementation. * Add description field to the gtk implementation. Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808 (cherry picked from commit `90f445729d`)	9 years ago
zuul	948a9b615f	Merge "pbx.c: Fix handling of '-' in extension name and callerid" into 13	9 years ago
zuul	f2a93dc9f2	Merge "pjproject_bundled: Update for pjproject 2.5.5" into 13	9 years ago
zuul	c2aed1c6e0	Merge "pbx.c: Allow dangerous functions when adding a hint to dialplan." into 13	9 years ago
Joshua Colp	7f9369c1b6	astconfigparser: Handle case where line is simply a comment. Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5	9 years ago
Joshua Colp	cc5660406c	Merge "dsp.c: Add fax and DTMF detection unit tests." into 13	9 years ago
Joshua Colp	cc825e54b0	Merge "dsp.c: Added descriptive comments to Goertzel calculations." into 13	9 years ago
Joshua Colp	323042d99d	Merge "dsp.c: Fix incorrect format reference typo." into 13	9 years ago
zuul	d5e380b8fb	Merge "dsp.c: Correct DTMF twist dsp.conf documentation." into 13	9 years ago
Joshua Colp	78de6eabbc	Merge "astconfigparser.py: Update with realtime fixes." into 13	9 years ago
Corey Farrell	57e9c66819	pbx.c: Fix handling of '-' in extension name and callerid This adds a two strings to ast_exten. name to go with exten and cidmatch_display to go with cidmatch. The new fields contain input used to add the extension in the first place. The existing fields now contain stripped input that excludes insignificant spaces and dashes. These stripped fields should always be used for comparisons. The unstripped fields should normally be used for display, but displaying stripped values will not cause runtime errors. Note the actual string is only stored twice if it contains dashes. If no dashes are found then both 'char *' fields point to the same memory. So this change has a minimum effect on memory usage. The existing functions ast_get_extension_name and ast_get_extension_cidmatch return unstripped values as they did before this change. Other similar bugs likely still exist where unstripped extensions are saved outside pbx.c then passed back in. ASTERISK-26233 #close Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f	9 years ago
Richard Mudgett	873fc0fda5	pbx.c: Allow dangerous functions when adding a hint to dialplan. We can allow dangerous functions when adding a hint since altering dialplan is itself a privileged activity. Otherwise, we could never execute dangerous functions. ASTERISK-25996 #close Reported by: Andrew Nagy Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba	9 years ago
zuul	7883f128d5	Merge "rtp_engine: Failed assertion and wrong name given for codec" into 13	9 years ago
Alexei Gradinari	f00525a6f6	pjproject: fixed a few bugs This patch fixes the issue in pjsip_tx_data_dec_ref() when tx_data_destroy can be called more than once, and checks if invalid value (e.g. NULL) is passed to. This patch updates array limit checks and docs in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability(). Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40	9 years ago
George Joseph	972cee2e4c	pjproject_bundled: Update for pjproject 2.5.5 Add more --disable-* switches to Makefile.rules including --disable-opus which was causing bundled pjproject to fail with "undefined reference" errors in libasteriskpj. Changed PJ_ENABLE_EXTRA_CHECK to 1. Removed 2 obsolete patches and added a new one. The new one was merged by Teluu on 6/27/2016. ASTERISK-26148 #close Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063	9 years ago
David M. Lee	8902a51d59	Portably sscanf tv_usec In a timeval, tv_usec is defined as a suseconds_t, which could be different underlying types on different platforms. Instead of trying to scanf directly into the timeval, scanf into a long int, then copy that into the timeval. Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95	9 years ago
Kevin Harwell	852e763571	rtp_engine: Failed assertion and wrong name given for codec Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c	9 years ago
Richard Mudgett	e8c34680ca	dsp.c: Add fax and DTMF detection unit tests. * Add fax amplitude and frequency sweep tests. * Add DTMF amplitude and twist unit tests. Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7	9 years ago
Richard Mudgett	c1f240b818	dsp.c: Added descriptive comments to Goertzel calculations. * Added doxygen to describe some struct members and what is going on in the code. Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d	9 years ago
Richard Mudgett	003a52fd62	dsp.c: Fix incorrect format reference typo. Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896	9 years ago
Richard Mudgett	4c0a0cbe02	dsp.c: Correct DTMF twist dsp.conf documentation. Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae	9 years ago
Joshua Colp	87433c2566	astconfigparser.py: Update with realtime fixes. When configuring SIP URIs in the pjsip.conf file it is necessary to escape the semicolon so the parser does not treat it as a comment. This change allows this to work in the astconfigparser implementation. A secondary bug where some data was lost if a configuration option included a "=" in its value was also fixed. A bug where sections would be considered equal despite being different has also been fixed. Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8	9 years ago
Richard Mudgett	159e437e5a	dsp.c: Fix erroneous fax tone detection. The Goertzel calculations get less accurate the lower the signal level being worked with becomes because there is less resolution remaining. If it is too low we can erroneously detect a tone where none really exists. The searched for fax frequencies not only need to be so much stronger than the background noise they must also be a minimum strength. * Add needed minimum threshold test to tone_detect(). * Set TONE_THRESHOLD to allow low volume frequency spread detection. ASTERISK-26237 #close Reported by: Richard Mudgett Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc	9 years ago
zuul	7ec9819403	Merge "Fix sqlalchemy error regarding identifier length." into 13	9 years ago
zuul	17e0e058ca	Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)." into 13	9 years ago
Mark Michelson	eda95236d1	Fix sqlalchemy error regarding identifier length. sqlalchemy was complaining: sqlalchemy.exc.IdentifierError: Identifier 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 characters This fixes the problem by changing the index name to be "ps_contacts_qualifyfreq_exp" instead. ASTERISK-26227 #close Reported by Mark Michelson Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9	9 years ago
zuul	8d6a7b89bd	Merge "res_pjsip: Whitespace and comment cleanup." into 13	9 years ago
Alexander Traud	66c9dfb272	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	9 years ago
Joshua Colp	0de05c2938	Merge "chan_sip: Prevent deadlock when issuing "sip show channels"" into 13	9 years ago
zuul	e3fbb4e099	Merge "res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice." into 13	9 years ago
Richard Mudgett	33716106e0	res_pjsip: Whitespace and comment cleanup. Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38	9 years ago
zuul	80a9899100	Merge "chan_dahdi.c: Fix deadlock potential in fax redirection." into 13	9 years ago
zuul	2b001fd6aa	Merge "chan_sip.c: Fix deadlock potential in fax redirection." into 13	9 years ago
zuul	00ed6b74ea	Merge "chan_pjsip.c: Fix deadlock potential in fax redirection." into 13	9 years ago
zuul	ffbaefa48f	Merge "res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook." into 13	9 years ago
Joshua Colp	0b8448a74b	Merge changes from topic 'ASTERISK-26214' into 13 * changes: res_fax: Fix FAXOPT(faxdetect) timeout option. chan_dahdi: Add faxdetect_timeout option.	9 years ago
George Joseph	52ab0bf258	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
Joshua Colp	efebb1b9e0	Merge "res_pjsip: Add fax_detect_timeout endpoint option." into 13	9 years ago
zuul	dec1e31f45	Merge "Add conditional support for noreturn functions." into 13	9 years ago
Alexei Gradinari	5997ec7c9e	res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. This patch removed call of pjsip_tx_data_dec_ref in send_notify if send_request failed. The pjsip_dlg_send_request deletes the message on error by itself. It seems this patch fixes next issues: ASTERISK-26199 ASTERISK-26166 ASTERISK-26174 Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a	9 years ago
zuul	d4242d6250	Merge "Makefile: Retain XML Declaration and DTD in docs." into 13	9 years ago
zuul	7dacb14c03	Merge "Unit tests: Use AST_TEST_DEFINE in conditional code only." into 13	9 years ago
zuul	290269bb23	Merge "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets." into 13	9 years ago
zuul	7ce180a754	Merge "res_pjsip_mwi: remove unneeded check on endpoint's contacts." into 13	9 years ago
Corey Farrell	7fdf7c3d4c	Add conditional support for noreturn functions. This adds support for tagging functions with the noreturn attribute. If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE and DO_CRASH are enabled then failed assertions never return. This can resolve a large number of false positives with static analyzers. ASTERISK-26220 #close Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753	9 years ago
Richard Mudgett	dcb8aa8c1c	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	fa91cf3eec	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	2e1bdc3775	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	628e8c91d5	res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. The fax_detect_framehook() 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. * Made only detach the framehook if we detected a fax and not on other possible frames. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d	9 years ago
Richard Mudgett	676aeede36	res_fax: Fix FAXOPT(faxdetect) timeout option. The fax detection timeout option did not work because basically the wrong variable was checked in fax_detect_framehook(). As a result, the timer would timeout immediately and disable fax detection. * Fixed ignoring negative timeout values. We'd complain and then go right on using the negative value. * Fixed destroy_faxdetect() in the off-nominal case of an incomplete object creation. * Added more range checking to FAXOPT(gateway) timeout parameter. ASTERISK-26214 #close Reported by: Richard Mudgett Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976	9 years ago

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