28718 Commits (97a75e3829e2e263da1828a06dad97b40c7ff32c)

 

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Author	SHA1	Message	Date
Richard Mudgett	68ebf86e2f	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
Alexei Gradinari	b5bc2fdda8	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
Joshua Colp	4cbb735c28	Merge "Portably sscanf tv_usec"	9 years ago
George Joseph	b4f1c6380e	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	feb1a43412	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	1d364ac54f	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
David M. Lee	8802e55c26	Replace strdupa with more portable ast_strdupa ... The strdupa function is a GNU extension, and not widely portable. We have an ast_strdupa function used within Asterisk which is preferred. I pulled the definition up from menuselect.c into the menuselect.h header file so it can be shared across menuselect. Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e	9 years ago
Richard Mudgett	737471f131	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	a8cd5d255a	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	6dfb34cf13	dsp.c: Fix incorrect format reference typo. ... Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896	9 years ago
Richard Mudgett	327136088e	dsp.c: Correct DTMF twist dsp.conf documentation. ... Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae	9 years ago
Joshua Colp	1e7168aee0	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	49461f37b7	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	e2bfcb3e58	Merge "codecs: Add iLBC 20."	9 years ago
Joshua Colp	16971e0d3e	Merge "menuselect: Various menuselect enhancements"	9 years ago
zuul	a99d5a6997	Merge "asterisk.c: Add auto generation and persistence of UUID"	9 years ago
zuul	6c753ac225	Merge "pbx.c: Remove duplicate code."	9 years ago
George Joseph	b4c5dcad01	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	9 years ago
Joshua Colp	9db420c69d	ari: Update version. ... New functionality has been added so the version has been bumped to one over the 13 version. Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69	9 years ago
zuul	616446fe24	Merge "Fix sqlalchemy error regarding identifier length."	9 years ago
George Joseph	8852a4c3db	asterisk.c: Add auto generation and persistence of UUID ... Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d	9 years ago
zuul	7cfd9bf104	Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."	9 years ago
Mark Michelson	76781a0964	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	2949e674a5	Merge "Create Asterisk-14: Update CHANGES and UPGRADE files"	9 years ago
zuul	8e79e382b4	Merge "res_pjsip: Whitespace and comment cleanup."	9 years ago
Joshua Colp	fd87c7a70c	Merge "res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice."	9 years ago
Alexander Traud	9be69c1636	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
Alexander Traud	8fb807009f	codecs: Add iLBC 20. ... Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests this. ASTERISK-26218 #close ASTERISK-26221 #close Reported by: Aaron Meriwether Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa	9 years ago
zuul	075f7c4aea	Merge "chan_sip: Prevent deadlock when issuing "sip show channels""	9 years ago
Richard Mudgett	4286a369a1	res_pjsip: Whitespace and comment cleanup. ... Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38	9 years ago
Corey Farrell	68de3a9e51	pbx.c: Remove duplicate code. ... Merge code found in both branches of a conditional in ast_add_extension2_lockopt. The updated code initializes peer_table and peer_label_table of the extension before linking it to the context. Change-Id: Ic759e27cdc9906c6877df41d28ee9c5be8f41c20	9 years ago
zuul	9473818659	Merge "res_srtp: Enable AES-256 and AES-GCM."	9 years ago
zuul	9372fe1b95	Merge "chan_dahdi.c: Fix deadlock potential in fax redirection."	9 years ago
zuul	a58f15ee4b	Merge "chan_sip.c: Fix deadlock potential in fax redirection."	9 years ago
zuul	ba2da66bd5	Merge "chan_pjsip.c: Fix deadlock potential in fax redirection."	9 years ago
zuul	3abf482393	Merge "res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook."	9 years ago
Joshua Colp	7f36b79f87	Merge "res_fax: Fix FAXOPT(faxdetect) timeout option."	9 years ago
Joshua Colp	4ffffa8bc4	Merge "chan_dahdi: Add faxdetect_timeout option."	9 years ago
Joshua Colp	0933f0cf96	Merge "res_pjsip: Add fax_detect_timeout endpoint option."	9 years ago
George Joseph	15bf6a87dc	Create Asterisk-14: Update CHANGES and UPGRADE files ... Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc	9 years ago
George Joseph	1b4922466b	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
zuul	194d0f606b	Merge "pbx: Create pbx_sw.c for management of 'struct ast_sw'."	9 years ago
zuul	fbdeb01edf	Merge "Add conditional support for noreturn functions."	9 years ago
Corey Farrell	a36a174c4b	pbx: Create pbx_sw.c for management of 'struct ast_sw'. ... This changes context switches from a linked list to a vector, makes 'struct ast_sw' opaque to pbx.c. Although ast_walk_context_switches is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_switches_count (AST_VECTOR_SIZE) * ast_context_switches_get (AST_VECTOR_GET) As with ast_walk_context_switches callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the switches, they have been converted to use the new functions. Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998	9 years ago
Alexei Gradinari	81ea024d93	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
Alexander Traud	1d2173c7ae	res_srtp: Enable AES-256 and AES-GCM. ... ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b	9 years ago
zuul	3ca6407dab	Merge "Makefile: Retain XML Declaration and DTD in docs."	9 years ago
zuul	7ff9bed7b0	Merge "Unit tests: Use AST_TEST_DEFINE in conditional code only."	9 years ago
zuul	b1c45dc815	Merge "pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'."	9 years ago
zuul	e51b40bd87	Merge "res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets."	9 years ago