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``` What steps will reproduce the problem? 1.Register extension and call to a demo extension in asterisk,which plays an audio file 2.In firefox everything is fine and audio is heard...
``` What steps will reproduce the problem? 1. ./configure --with-ssl --with-srtp --with-vpx --with-yuv --with-amr --with-speex --with-speexdsp --with-opus --with-gsm --with-ilbc --with-g729 --with-ffmpeg --enable-debug What is the expected output? What do you...
``` What steps will reproduce the problem? 1. Environment setup using Asterisk, WebRTC2SIP and Browser (Chrome and/or FF) 2. Login by SIP client on (http://sipml5.org/call.htm?svn=222#) 2. Initiate call from legacy...
``` Hi, I am getting an Assertion failure while running a voice call through the newly created Boghe build in plugin: pluginWinMF in file: mf_utils.cxx while calling function: assert(pSession &&...
``` Hi, I need to make calls between sipml5 web clients and legacy sip clients. Everything is working when I am connected with internet. However whenever I try the same...
``` I have got the following error while installing webrtc2sip. I have followed http://linux.autostatic.com/installing-webrtc2sip-on-ubuntu-1204. I am getting the following message while installing. Please help me on this. make -j `getconf...
``` What steps will reproduce the problem? 1.Asterisk 11.6-cert9 installed through below reference: http://www.voip-info.org/wiki/view/Asterisk+11+Installation+on+CentOS+6 2. Webrtc2sip and related items installed through below guide: http://webrtc2sip.org/technical-guide-1.0.pdf installed only audio codec and mandatory...
``` What steps will reproduce the problem? 1. Asterisk and webrtc2sip is installed on centOS 6.6 2. Created certificates for config.xml 3. Try to make call from sipml5 client What...
``` Hi, the call seems established, asterisk is sending rtp packages, no errors on the javascript console on sipml5 but in webrtc2sip i get following error: warning: The VAD has...
``` What steps will reproduce the problem? 1. Enable RTCWeb Breaker 2. Place a call 3. Have the SIP proxy challenge the INVITE with a 401 What is the expected...