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Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006
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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
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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
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* 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)
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Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5
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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
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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
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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
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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
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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
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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
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* Add fax amplitude and frequency sweep tests.
* Add DTMF amplitude and twist unit tests.
Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7
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* Added doxygen to describe some struct members and what is going on in
the code.
Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d
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Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896
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Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae
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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
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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
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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
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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
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twice." into 13
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Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
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into 13
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* changes:
res_fax: Fix FAXOPT(faxdetect) timeout option.
chan_dahdi: Add faxdetect_timeout option.
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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
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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
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