Review Board 1.7.16

Review requests for asterisk-dev

Starred Summary Submitter (Ascending) Unsort Posted Last Updated Edit columns
IAX2: Transfer Reject: Lock bridgecallno before touching it, refactor alecdavis June 11th, 2013, 8:01 a.m.
IAX2: refactor nativebridge transfer. alecdavis June 9th, 2013, 12:13 a.m.
IAX2: fix race condition when transferrring. alecdavis June 4th, 2013, 11:16 p.m.
NOTIFYs for BLF start queuing up and fail to be sent out after retries fail alecdavis April 25th, 2013, 7:35 a.m.
Asterisk doesn't honor Session-Expires in 200OK response when it's a RE-INVITE when asterisk is the refresher. alecdavis April 22nd, 2013, 10:26 p.m.
SIP Session-Expires: Set timer to correctly expire at (~2/3) of the expiry interval when not the refresher. alecdavis May 1st, 2013, 6:58 p.m.
Fix Distributed Device State broken at sites using res_xmpp or res_jabber where Secuity Advisory AST-2012-015 is inplace. alecdavis April 15th, 2013, 12:29 a.m.
IAX2 defer_full_frames fail to get sent alecdavis April 4th, 2013, 3:33 a.m.
IAX2, prevent network thread starting before all helper threads are ready alecdavis April 5th, 2013, 7:13 a.m.
DIALOG_INFO_XML stale (and timeout) notifiy subscriptions stop BLF's working on Grandstream GXP20xx phones. alecdavis March 12th, 2012, 7:44 p.m.
Fix double DTMF digits when 'dtmfmode=inband' and client sends both 'inband' and 'SIP INFO' packets alecdavis October 17th, 2012, 1:53 a.m.
dsp.c User Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END alecdavis October 4th, 2012, 2:02 a.m.
dsp.c fix incorrect DTMF Digit_Duration, it's always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if hitstobegin=2 alecdavis October 4th, 2012, 5:56 a.m.
dsp.c User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values alecdavis October 2nd, 2012, 2:28 a.m.
app_queue: initialise "available agent' hint on restart, and other senarios alecdavis September 20th, 2012, 9:38 p.m.
dsp.c ast_dsp_call_progress use local short variable in loop, plus other cleanup alecdavis September 12th, 2012, 5:37 a.m.
app_queue: support a 'logged in and available' hint on queue alecdavis September 19th, 2012, 12:45 a.m.
Support a hint on a queue. alecdavis December 12th, 2011, 5:25 p.m.
dsp.c: in ast_mf_detect_init incorrectly sets goertzel samples to 160, should be 120, or better MF_GSIZE alecdavis September 5th, 2012, 3:52 a.m.
dsp.c: dtmf_detect, Fix multiple issues when no-interdigit delay is present. alecdavis August 26th, 2012, 5:35 a.m.
dsp.c optimize goerztzel sample loops, in dtmf_detect, mf_detect and tone_detect alecdavis September 1st, 2012, 6:03 a.m.
chan_sip: [general] maxforwards, not checked for a value greater than 255 alecdavis April 26th, 2012, 4:09 a.m.
Generate VMWI neon pulses from FXS module to light NEON lamp on older 'non intellegent phones' alecdavis March 17th, 2011, 4:22 a.m.
RFC3261 Section The sequence number value MUST be expressible as a 32-bit unsigned integer alecdavis January 27th, 2012, 12:35 a.m.
sync chan_dahdi->p->outgoing with sig_XXX->p-outgoing alecdavis February 16th, 2012, 3:39 a.m.
rfc4235 - Section 4.1: Versions MUST be representable using a non-negative 32 bit integer. alecdavis January 26th, 2012, 3:11 a.m.
cleanup dialog-info+xml Notify dialog alecdavis January 25th, 2012, 7:26 p.m.
dsp.c refactor section of dtmf_detect alecdavis March 2nd, 2011, 5:33 a.m.
use ie2str(full_ie) where possible in q931.c alecdavis December 19th, 2011, 3:34 a.m.
log Asterisk Version number, Build etc into log file alecdavis September 2nd, 2011, 11:46 p.m.
Prevent segfault when asterisk restarts. Happens if call arrives before fully booted. alecdavis September 2nd, 2011, 4:43 a.m.
Add Congestion detail in CDR call logs when Congestion application is used. alecdavis January 6th, 2010, 1:52 a.m.
Fix *8 directed pickup locks system while sucessful pickupsound plays out. alecdavis May 26th, 2011, 6:59 a.m.
Fix directed group pickup feature code *8 with pickupsounds enabled , deadlock and segfault, affects 1.8.0 and trunk alecdavis April 17th, 2011, 5:52 a.m.
Fix SEGFAULT in remote_bridge_loop after a SIP to SIP attended transfer with external IAX2 or DAHDI call alecdavis March 1st, 2011, 1:31 a.m.
fix Deadlock with attended transfers of SIP calls alecdavis February 24th, 2011, 3:07 a.m.
Prevent DTMF incorrectly triggering during SAS+CAS (CallWaiting) signalling to an FXS port alecdavis October 19th, 2010, 5:49 a.m.
Prevent 'Bad Magic Number' caused when a channel is optimized out by masquerade alecdavis September 17th, 2010, 9:09 p.m.
Dahdi FXS line polarity reversal when remote party Answers and/or Hangups alecdavis July 22nd, 2010, 6:49 a.m.
VoicemailMain and VMauthenticate: Like VoiceMail allow escape to the 'a' extension when a single '*' is entered in Mailbox or Password alecdavis February 5th, 2010, 3:10 a.m.
app_directory, support option 'p(n)' as documented with CLI online help, branches 1.6.1 to trunk. alecdavis February 2nd, 2010, 3:05 a.m.
CDR: add Dialed Number Identifier field (DNID) field in record alecdavis January 7th, 2010, 2:05 a.m.
CDR: Add Calling and Called Subaddress fields to CDR record alecdavis January 14th, 2010, 3:01 a.m.
Add Calling and Called subaddress support for Asterisk apps and funcs alecdavis October 12th, 2009, 10:59 p.m.
Allow app_dial to play 'indication tone while ringing' like 'option m' which will provide music on hold while ringing alecdavis December 16th, 2009, 4:28 a.m.
Add ISDN Calling and Called Subaddress support functions to LIBPRI alecdavis October 15th, 2009, 4:52 a.m.
add initial support for AT+CUSD command Artem June 5th, 2009, 4:05 p.m.
Asterisk: stasis: set a channel variable on websocket disconnect error asanders March 23rd, 2015, 4:04 a.m.
chan_sip: Asterisk fails to re-activate an inactive media session when an offer does not contain a=sendrecv asanders February 23rd, 2015, 9:38 p.m.
Asterisk: For httpd server, need option to define server name for security purposes asanders January 26th, 2015, 7:57 p.m.
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