28288 Commits (e85adbd947fac6b5385f1b74eba506e53c78f483)

 

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Author	SHA1	Message	Date
zuul	8878cedc0a	Merge "res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports."	10 years ago
zuul	01295fbdf4	Merge "menuselect: Add an opaque "member_data" string to the acceptable xml"	10 years ago
zuul	eba0ad457a	Merge "sorcery: Use more compatible regex for local expressions."	10 years ago
zuul	84f0007232	Merge "pjproject: fixed a few bugs"	10 years ago
Joshua Colp	73bce50ef8	sorcery: Use more compatible regex for local expressions. ... This changes the use of an empty regex for both res_sorcery_config and res_sorcery_memory to "." instead. This is a more compatible regular expression which also works on FreeBSD. ASTERISK-26206 #close Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388	10 years ago
Alexander Traud	3ff964c6b6	res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports. ... ASTERISK-26256 #close Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058	10 years ago
George Joseph	f6276441b1	menuselect: Add an opaque "member_data" string to the acceptable xml ... Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe	10 years ago
zuul	de7e9fe95b	Merge "astconfigparser: Handle case where line is simply a comment."	10 years ago
Joshua Colp	720a4ff663	Merge "Remove SILK payload mappings from Asterisk core."	10 years ago
Joshua Colp	b8fdd3ad79	Merge "pbx.c: Fix handling of '-' in extension name and callerid"	10 years ago
Mark Michelson	1cd79d6ee5	Remove SILK payload mappings from Asterisk core. ... SILK is a bit of a hog when it comes to using up our limited number of dynamic payload types in the RTP engine. By freeing up four slots, it allows for other codecs to potentially take the place. Now, codec_silk.so will dynamically use the payload slots in the RTP engine when it loads. A better fix would be make RTP dynamic payload types actually dynamic. However, at this stage of Asterisk 14 development, this is a risky move that would be imprudent. Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612	10 years ago
zuul	d462cb73a8	Merge "pjproject_bundled: Update for pjproject 2.5.5"	10 years ago
Joshua Colp	9cead7c233	Merge "pbx.c: Allow dangerous functions when adding a hint to dialplan."	10 years ago
Joshua Colp	a7ae48441f	astconfigparser: Handle case where line is simply a comment. ... Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5	10 years ago
Joshua Colp	6131d70aa8	Merge "astconfigparser.py: Update with realtime fixes."	10 years ago
Corey Farrell	89a0a1eb45	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	10 years ago
Joshua Colp	aa69877049	Merge "dsp.c: Add fax and DTMF detection unit tests."	10 years ago
Joshua Colp	28e61e43d7	Merge "dsp.c: Added descriptive comments to Goertzel calculations."	10 years ago
Joshua Colp	085da4eec0	Merge "dsp.c: Fix incorrect format reference typo."	10 years ago
Joshua Colp	639034d951	Merge "dsp.c: Correct DTMF twist dsp.conf documentation."	10 years ago
zuul	479f72c868	Merge "rtp_engine: Failed assertion and wrong name given for codec"	10 years ago
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	10 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	10 years ago
Joshua Colp	4cbb735c28	Merge "Portably sscanf tv_usec"	10 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	10 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	10 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	10 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	10 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	10 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	10 years ago
Richard Mudgett	6dfb34cf13	dsp.c: Fix incorrect format reference typo. ... Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896	10 years ago
Richard Mudgett	327136088e	dsp.c: Correct DTMF twist dsp.conf documentation. ... Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae	10 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	10 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	10 years ago
zuul	e2bfcb3e58	Merge "codecs: Add iLBC 20."	10 years ago
Joshua Colp	16971e0d3e	Merge "menuselect: Various menuselect enhancements"	10 years ago
zuul	a99d5a6997	Merge "asterisk.c: Add auto generation and persistence of UUID"	10 years ago
zuul	6c753ac225	Merge "pbx.c: Remove duplicate code."	10 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	10 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	10 years ago
zuul	616446fe24	Merge "Fix sqlalchemy error regarding identifier length."	10 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	10 years ago
zuul	7cfd9bf104	Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)."	10 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	10 years ago
zuul	2949e674a5	Merge "Create Asterisk-14: Update CHANGES and UPGRADE files"	10 years ago
zuul	8e79e382b4	Merge "res_pjsip: Whitespace and comment cleanup."	10 years ago
Joshua Colp	fd87c7a70c	Merge "res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice."	10 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	10 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	10 years ago
zuul	075f7c4aea	Merge "chan_sip: Prevent deadlock when issuing "sip show channels""	10 years ago