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http://forums.asterisk.org/viewtopic.php?t=87606
Aug 10, 2013 · I have configured a single asterisk server (Asterisk 11) to take an incoming call from Google Voice, answer it and play back a soundfile. By watching the debugs, I can: 1) See the call arrive via the XMPP debugs 2) See the call get mapped to the correct context and the extension get matched 3) See the answer and playback functions invoked.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
Sep 22, 2016 · When building Asterisk 11, to get ICE support you'll need the UUID development library (uuid-dev for Debian, libuuid-devel for CentOS) library. If you don't have ICE support, then you'll likely run into audio issues in several scenarios, specifically when attempting to traverse NAT, as WebRTC uses ICE,STUN,TURN to do this.
https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
May 02, 2019 · You may already have some of the config from previous webrtc endpoints for certificates, keys, encryption, ice support etc and think you don't need to add the magical webrtc=yes but you do! The webrtc=yes flag does more than just shortcut already existing flags which are needed for proper SFU support.. There are two more Asterisk changes we need to make so no need to restart Asterisk just yet.
https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample
Apr 17, 2019 · All your code in one place. GitHub makes it easy to scale back on context switching. Read rendered documentation, see the history of any file, and collaborate with contributors on projects across GitHub.
https://reviewboard.asterisk.org/r/3086/diff/1-2/
pjsip.conf.sample update: improve documentation of pjsip endpoints behind NAT and update for snake case change Review Request #3086 - Created Dec. 19, 2013 and submitted Dec. 20, 2013, 1:19 p.m.
https://www.reddit.com/r/Asterisk/comments/3zm806/webrtc_video_on_asterisk_1360/
Jan 05, 2016 · The problem is that there are a log of old outdated articles discussing Asterisk 11, however in Asterisk 12, 13 the sipstack have been changed to pjsip. An updated guide can be found here: Asterisk WebRTC setup
https://reviewboard.asterisk.org/r/3992/diff/1-2/
Description: When enabling SRTP support in PJSIP it is either forced on or disabled. This means that if you specify SRTP but the client does not support it the session will fail.
https://asterisk-rd.blogspot.com/2015/05/webrtc-tutorial-using-sipml5.html
This tutorial written using Debian Squeeze 6.0.5, Asterisk 11.8.0-rc1 and Asterisk's chan_sip channel driver. We assume you are a little familiar with Asterisk, and have an Asterisk installation available via a public IP address, and control of the firewall in front of it. (you do have it fire-walled right?)
https://stackoverflow.com/questions/49682747/asterisk-gives-strict-rtp-learning-message-and-no-audio-for-chrome-webrtc-but
Entered [] for the "ICE servers" field (because I'm on a local LAN with no NAT involved, I don't need STUN or TURN, though I do have ICE enabled in my Asterisk config) Entered my "Websocket Server URL" value of wss://asterisk-ci.test:8089/ws; Clicked Save and then returned to the demo page.
https://community.freepbx.org/t/how-to-guide-for-google-voice-with-freepbx-14-asterisk-gvsip-ubuntu-18-04/50933
Next, simply installing fail2ban does not setup the jail for asterisk, only for sshd, so lets make a jail for asterisk that uses the default log configuration, this can be adjusted to point to different log files if you have made adjustments to your log file settings.
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