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2016-08-03Add missing checks during startup.Corey Farrell
This ensures startup is canceled due to allocation failures from the following initializations. * channel.c: ast_channels_init * config_options.c: aco_init ASTERISK-26265 #close Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611
2016-08-03astconfigparser: Really handle case where line is simply a comment.Joshua Colp
The regular expression would match causing the code that handled the line if it was merely a comment to never get executed. Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819
2016-08-02Merge "asterisk.c: Add auto generation and persistence of UUID" into 13zuul
2016-08-02Merge "Remove SILK payload mappings from Asterisk core." into 13zuul
2016-08-02Merge "res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 ↵Joshua Colp
transports." into 13
2016-08-02asterisk.c: Add auto generation and persistence of UUIDGeorge Joseph
Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d
2016-08-02Merge "menuselect: Add an opaque "member_data" string to the acceptable ↵Joshua Colp
xml" into 13
2016-08-02Merge "rest-api: Code out of sync with the model" into 13Joshua Colp
2016-08-02rest-api: Code out of sync with the modelKevin Harwell
Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006
2016-08-02Merge "sorcery: Use more compatible regex for local expressions." into 13Joshua Colp
2016-08-02Remove SILK payload mappings from Asterisk core.Mark Michelson
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 (cherry picked from commit d50895c7b04036aeaad58990089399e46db4c817)
2016-08-02Merge "pjproject: fixed a few bugs" into 13Joshua Colp
2016-08-02sorcery: Use more compatible regex for local expressions.Joshua Colp
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
2016-08-02res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports.Alexander Traud
ASTERISK-26256 #close Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058
2016-08-01menuselect: Add an opaque "member_data" string to the acceptable xmlGeorge Joseph
Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe
2016-08-01Merge "Replace strdupa with more portable ast_strdupa" into 13zuul
2016-08-01Merge "menuselect: Various menuselect enhancements" into 13zuul
2016-08-01Merge "astconfigparser: Handle case where line is simply a comment." into 13zuul
2016-08-01Replace strdupa with more portable ast_strdupaDavid M. Lee
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
2016-08-01menuselect: Various menuselect enhancementsGeorge Joseph
* 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 90f445729d5d86050d9d379485ff0a99f4a006c1)
2016-08-01Merge "pbx.c: Fix handling of '-' in extension name and callerid" into 13zuul
2016-07-29Merge "pjproject_bundled: Update for pjproject 2.5.5" into 13zuul
2016-07-29Merge "pbx.c: Allow dangerous functions when adding a hint to dialplan." into 13zuul
2016-07-29astconfigparser: Handle case where line is simply a comment.Joshua Colp
Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5
2016-07-28Merge "dsp.c: Add fax and DTMF detection unit tests." into 13Joshua Colp
2016-07-28Merge "dsp.c: Added descriptive comments to Goertzel calculations." into 13Joshua Colp
2016-07-28Merge "dsp.c: Fix incorrect format reference typo." into 13Joshua Colp
2016-07-28Merge "dsp.c: Correct DTMF twist dsp.conf documentation." into 13zuul
2016-07-28Merge "astconfigparser.py: Update with realtime fixes." into 13Joshua Colp
2016-07-28pbx.c: Fix handling of '-' in extension name and calleridCorey Farrell
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
2016-07-28pbx.c: Allow dangerous functions when adding a hint to dialplan.Richard Mudgett
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
2016-07-28Merge "rtp_engine: Failed assertion and wrong name given for codec" into 13zuul
2016-07-28pjproject: fixed a few bugsAlexei Gradinari
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
2016-07-28pjproject_bundled: Update for pjproject 2.5.5George Joseph
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
2016-07-27Portably sscanf tv_usecDavid M. Lee
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
2016-07-27rtp_engine: Failed assertion and wrong name given for codecKevin Harwell
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
2016-07-26dsp.c: Add fax and DTMF detection unit tests.Richard Mudgett
* Add fax amplitude and frequency sweep tests. * Add DTMF amplitude and twist unit tests. Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7
2016-07-26dsp.c: Added descriptive comments to Goertzel calculations.Richard Mudgett
* Added doxygen to describe some struct members and what is going on in the code. Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d
2016-07-26dsp.c: Fix incorrect format reference typo.Richard Mudgett
Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896
2016-07-26dsp.c: Correct DTMF twist dsp.conf documentation.Richard Mudgett
Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae
2016-07-26astconfigparser.py: Update with realtime fixes.Joshua Colp
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
2016-07-25dsp.c: Fix erroneous fax tone detection.Richard Mudgett
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
2016-07-23Merge "Fix sqlalchemy error regarding identifier length." into 13zuul
2016-07-22Merge "chan_sip: Enable Session-Timers for SIP over TCP (and TLS)." into 13zuul
2016-07-22Fix sqlalchemy error regarding identifier length.Mark Michelson
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
2016-07-22Merge "res_pjsip: Whitespace and comment cleanup." into 13zuul
2016-07-22chan_sip: Enable Session-Timers for SIP over TCP (and TLS).Alexander Traud
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
2016-07-22Merge "chan_sip: Prevent deadlock when issuing "sip show channels"" into 13Joshua Colp
2016-07-22Merge "res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called ↵zuul
twice." into 13
2016-07-21res_pjsip: Whitespace and comment cleanup.Richard Mudgett
Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
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