- Jul 27, 2016
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David M. Lee authored
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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- Jul 26, 2016
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Richard Mudgett authored
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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zuul authored
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Joshua Colp authored
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zuul authored
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zuul authored
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- Jul 25, 2016
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George Joseph authored
* 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
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- Jul 24, 2016
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Joshua Colp authored
New functionality has been added so the version has been bumped to one over the 13 version. Change-Id: I5d30077f62640c0ac83599b4e9a9b657bf184f69
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- Jul 23, 2016
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zuul authored
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George Joseph authored
Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d
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- Jul 22, 2016
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zuul authored
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Mark Michelson authored
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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zuul authored
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zuul authored
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Joshua Colp authored
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Alexander Traud authored
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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Alexander Traud authored
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
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zuul authored
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Richard Mudgett authored
Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38
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Corey Farrell authored
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
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zuul authored
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zuul authored
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zuul authored
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zuul authored
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zuul authored
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- Jul 21, 2016
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Joshua Colp authored
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Joshua Colp authored
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Joshua Colp authored
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George Joseph authored
Change-Id: I35b5f6657670cfa8985796fa1e1fe86ad299efdc
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George Joseph authored
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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zuul authored
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zuul authored
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Corey Farrell authored
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
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Alexei Gradinari authored
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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Alexander Traud authored
ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b
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- Jul 20, 2016