Review Board 1.7.16


Olle E Johansson
Last logged in December 19th, 2014
Joined October 22nd, 2008


oej's review requests

Starred Summary Submitter Posted Last Updated Edit columns
chan_sip: Add support for a few more 4xx error responses oej April 11th, 2014, 7:29 a.m.
Support multiple Require: and Supported: headers in the same request oej April 29th, 2013, 10:05 a.m.
Handle 183 without SDP - don't always convert to ringing oej June 2nd, 2014, 2:01 p.m.
DTMF emulation bad calculation that hurts RTP oej May 16th, 2014, 1:50 p.m.
Implement SIP TImer C in Asterisk oej April 11th, 2014, 8:41 a.m.
chan_sip: Support a=rtcp attribute in SDP oej April 11th, 2014, 1:46 p.m.
Implement Externaddr on a sip device basis oej September 29th, 2011, 7:07 a.m.
Pre-review of work to handle SRTP lifetime and MKI in a less bad way, but not the best way oej April 11th, 2014, 2:53 p.m.
AMI :: Debug manager actions in the CLI oej September 7th, 2011, 9:04 a.m.
Manage translation table between SIP and ISDN hangup causes oej December 3rd, 2012, 9:53 a.m.
Play periodic prompts for first call in a call queue oej January 7th, 2013, 7:36 a.m.
Add Path header support to chan_sip oej December 7th, 2012, 4:34 a.m.
Expand MEETME_INFO() with new options oej July 28th, 2010, 5 a.m.
Enable support for early media in AMI originate action oej September 29th, 2011, 9:14 a.m.
Dialplan function for manager account checks - AMI_CLIENT() oej November 3rd, 2009, 1:12 p.m.
AMI - Set channel variables with setvar= when manager user originates calls oej September 5th, 2011, 7:18 a.m.
Pinequeue: Play queue prompts in the background - making call available to agent faster oej April 26th, 2012, 3:35 a.m.
Make it possible to change DTMF min duration in asterisk.conf oej April 24th, 2012, 4:31 a.m.
Preserve DTMF length in main/features.c oej September 27th, 2011, 8:50 a.m.
Manager originate: Don't fall back to "s" when given extension does not exist oej September 19th, 2011, 11:02 a.m.
Add CLI-command "cdr pgsql status" based on "cdr mysql status" oej September 17th, 2010, 6:22 a.m.
AGI Streamfile does not restart MOH. oej September 29th, 2011, 4:40 a.m.
Make pbx_start NOT default to s@default if extension not found oej September 19th, 2011, 12:09 p.m.
Change default strictrtp to "yes" - enabled - in rtp.conf.sample and res_rtp_asterisk.c oej September 20th, 2011, 10:08 a.m.
Meetme option: Drop conference when only one participant remains at exit oej August 23rd, 2011, 7:41 a.m.
Set default tonezone for SIP devices oej September 12th, 2011, 7:30 a.m.
Lock the peer->mwipvt to avoid crashed in sip history management when overflowing history entries oej August 18th, 2011, 6:41 a.m.
Add manager event for local "semi-bridge" oej March 30th, 2011, 10:01 a.m.
Chan_sip: Voice frame dropped for every early media audio call oej April 18th, 2011, 4:12 a.m.
Chan_local crashes in local_fixup oej April 1st, 2011, 9:02 a.m.
Don't delay DTMF in core bridge while listening for DTMF features oej February 3rd, 2011, 2:41 a.m.
Swedish saynumber (again :-) ) - correctly say thousands and millions oej November 23rd, 2010, 9:03 a.m.
saynumber(1,n) doesn't work with language = SE oej November 22nd, 2010, 1:50 p.m.
Fix memory leak in manager.c action originate when using channel variables oej August 20th, 2010, 5:36 a.m.
Make sure that we handle possible format error in ast_openstream_full() - channel.c oej September 7th, 2010, 2:27 p.m.
Add ability to set Max-Forwards header from dialplan, general and device configuration oej July 13th, 2010, 11:26 a.m.
LDAP realtime driver corrections oej July 22nd, 2010, 7:32 a.m.
Add dialplan function to check if a queue exists or not oej July 13th, 2010, 9:54 a.m.
CDR_CSV: Add ability to disable cdr files per account code oej May 31st, 2010, 10:41 a.m.
Add a new option "require" to modules.conf to make Asterisk fail if a module does not load oej November 12th, 2009, 7:40 a.m.
SIP Contact ACL's return bad error code oej October 26th, 2009, 3:34 p.m.
Support RTCP reports without any report blocks correctly oej February 17th, 2010, 11:08 a.m.
AMI Setvar: Return error when function does not exist or generate error oej November 30th, 2009, 2:42 p.m.
Add mutestream manager action and MUTESTREAM() dialplan function oej August 31st, 2009, 12:37 p.m. 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.
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