Review Board 1.7.16


Review requests for asterisk-dev

Starred Summary (Ascending) Unsort Submitter (Ascending) Unsort Posted Last Updated Edit columns
Add Calling and Called subaddress support for Asterisk apps and funcs alecdavis October 12th, 2009, 10:59 p.m.
Add Congestion detail in CDR call logs when Congestion application is used. alecdavis January 6th, 2010, 1:52 a.m.
Add ISDN Calling and Called Subaddress support functions to LIBPRI alecdavis October 15th, 2009, 4:52 a.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.
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.
app_queue: initialise "available agent' hint on restart, and other senarios alecdavis September 20th, 2012, 9:38 p.m.
app_queue: support a 'logged in and available' hint on queue alecdavis September 19th, 2012, 12:45 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.
CDR: Add Calling and Called Subaddress fields to CDR record alecdavis January 14th, 2010, 3:01 a.m.
CDR: add Dialed Number Identifier field (DNID) field in record alecdavis January 7th, 2010, 2:05 a.m.
chan_sip: [general] maxforwards, not checked for a value greater than 255 alecdavis April 26th, 2012, 4:09 a.m.
cleanup dialog-info+xml Notify dialog alecdavis January 25th, 2012, 7:26 p.m.
Dahdi FXS line polarity reversal when remote party Answers and/or Hangups alecdavis July 22nd, 2010, 6:49 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.
dsp.c ast_dsp_call_progress use local short variable in loop, plus other cleanup alecdavis September 12th, 2012, 5:37 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 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: 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 optimize goerztzel sample loops, in dtmf_detect, mf_detect and tone_detect alecdavis September 1st, 2012, 6:03 a.m.
dsp.c refactor section of dtmf_detect alecdavis March 2nd, 2011, 5:33 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 User configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values alecdavis October 2nd, 2012, 2:28 a.m.
Fix *8 directed pickup locks system while sucessful pickupsound plays out. alecdavis May 26th, 2011, 6:59 a.m.
fix Deadlock with attended transfers of SIP calls alecdavis February 24th, 2011, 3:07 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 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.
Fix double DTMF digits when 'dtmfmode=inband' and client sends both 'inband' and 'SIP INFO' packets alecdavis October 17th, 2012, 1:53 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.
Generate VMWI neon pulses from FXS module to light NEON lamp on older 'non intellegent phones' alecdavis March 17th, 2011, 4:22 a.m.
IAX2 defer_full_frames fail to get sent alecdavis April 4th, 2013, 3:33 a.m.
IAX2: fix race condition when transferrring. alecdavis June 4th, 2013, 11:16 p.m.
IAX2, prevent network thread starting before all helper threads are ready alecdavis April 5th, 2013, 7:13 a.m.
IAX2: refactor nativebridge transfer. alecdavis June 9th, 2013, 12:13 a.m.
IAX2: Transfer Reject: Lock bridgecallno before touching it, refactor alecdavis June 11th, 2013, 8:01 a.m.
log Asterisk Version number, Build etc into log file alecdavis September 2nd, 2011, 11:46 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.
Prevent 'Bad Magic Number' caused when a channel is optimized out by masquerade alecdavis September 17th, 2010, 9:09 p.m.
Prevent DTMF incorrectly triggering during SAS+CAS (CallWaiting) signalling to an FXS port alecdavis October 19th, 2010, 5:49 a.m.
Prevent segfault when asterisk restarts. Happens if call arrives before fully booted. alecdavis September 2nd, 2011, 4:43 a.m.
RFC3261 Section 8.1.1.5. The sequence number value MUST be expressible as a 32-bit unsigned integer alecdavis January 27th, 2012, 12:35 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.
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.
Support a hint on a queue. alecdavis December 12th, 2011, 5:25 p.m.
sync chan_dahdi->p->outgoing with sig_XXX->p-outgoing alecdavis February 16th, 2012, 3:39 a.m.
use ie2str(full_ie) where possible in q931.c alecdavis December 19th, 2011, 3:34 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.
add initial support for AT+CUSD command Artem June 5th, 2009, 4:05 p.m.
Asterisk: Creating a named ARI bridge twice causes a crash asanders January 15th, 2015, 10:44 p.m.
Asterisk: For httpd server, need option to define server name for security purposes asanders January 26th, 2015, 7:57 p.m.
Asterisk: stasis: set a channel variable on websocket disconnect error asanders March 23rd, 2015, 4:04 a.m.
1 2 3 4 > » 63 pages

https://reviewboard.asterisk.org/ runs on a server provided by Digium, Inc. and uses bandwidth donated to the open source Asterisk community by API Digital Communications in Huntsville, AL USA.
Please report problems with this site to asteriskteam@digium.com.