Hey all, It’s been a bit over half a year since AstriDevCon last year. Unfortunately I wasn’t able to attend, but I wanted to review the notes and see how everybody in the community is doing with any items they wanted or committed to, as well as report progress that has been made from Digium’s perspective on areas it has engaged upon.
In conjunction with that, the cutting of the 14 branch is fast approaching, coming with all restrictions associated with merging code into release branches. Consider yourself warned :-) So, referencing https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2015, I’m just going to list the actionable items of interest that I was able to pull from the notes from AstriDevCon. If I missed something it wasn’t intentional. I also added notes next to the things that I have seen completed or in progress in my short time back in Asterisk land. As mentioned, feel free to reply with things I missed or statuses that are incorrect. From Loway/Lorenzo Emilitri’s presentation: app_queue skills based routing patch slide - patch at https://github.com/pascomnet/asterisk_sbr - Has this been submitted to be merged? Music on hold slide - no patches currently posted so not sure what kind of progress was made on it. Comments? Transfers slide - Couldn’t find much as far as specifically actionable items WebRTC integration - Want easier way than browser based SIP clients - Just using Asterisk/chan_sip not sure if things are better yet Remote file linking (HTTP): - Implemented by Digium/Matt Jordan Remote audio streams - status unknown Deploy a production box via ARI - Can push PJSIP configuration and other things via ARI now - https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration From Ben Klang’s presentation: MRCP via ARI - status unknown DTMF Matcher (with SRGS) - status unknown SSML Parser - status unknown Media Playback Optimizer - status unknown Arbitrary Tone Detectors - status unknown Asymmetric Audio Bridges - status unknown Remote fetch/store audio - Matt Jordan/Digium added support for this in master/14 From Torrey Searle’s presentation: New Function added to chan_sip: SIP_SDP_OFFER – returns a csv of codec/bitrate/type - status unknown New Channel Variable SIP_CODEC_OUTBOUND_ORDER – allows a csv of codecs to be specified, controls the order of SIP Codecs on the Outbound Leg, codecs not allowed by the peer are removed - status unknown If a call is bridged copy the codec order from the B leg to the A leg in the answer - status unknown If a call is bridged and there are common codecs between the B leg and the A leg, remove all not supported by B in A's response - status unknown I had a harder time sorting out Sean McCord’s presentation due to a lack of slides, but here are a few key points that I found: Improved ARI documentation of media URIs - status unknown Improved ARI documentation of when to use body versus path ids - status unknown Improved documentation of how to add new ARI functionality - status unknown When websocket connections are lost, issue in that channels hang round and new channel can enter without notifying anybody. Behavior improvement desired here. - status unknown Arbitrary tone detection (again) - status unknown Add docker file to Asterisk source tree - Leif Madsen submitted one and it has been merged. From other discussion section: Streaming MOH / audio - status unknown Update ARI spec to conform to a newer version of swagger - some discussion on the -dev list, but no code submitted. Here are a few additional major initiatives that Digium has worked on: Publishing extension/device presence to external SIP presence servers such as Kamailio - Done - https://wiki.asterisk.org/wiki/display/AST/Publishing+Extension+State ARI early bridging support: - Mostly completed Thanks again for all of your help and contributions. Asterisk would not be great without all of you who put in the time and effort necessary to move it forward. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
