Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support .
I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions. I have tried both browser Google Chrome and Firefox with their latest versions. For asterisk, I made some configuration like below. Please check : http://pastebin.com/7KCvtcNf For Outbound calls : when i am dialling 8002 -> 8001 every time Chrome Browser asking for allow microphone. Is there any way to disable asking permission and allowing it by default ? when i allow microphone then SIpml5 phone showing like "Not Allow". Here is the asterisk logs : http://pastebin.com/JZeDjyay For Incoming calls : When call come to browser,And allow microphone then Call rejected and asterisk showing like "Got SIP response 603 "Failed to get local SDP" in asterisk CLI. But After some google i found new link https://code.google.com/p/sipml5/wiki/Downloads for "SIPml-api.js" and after replacing that JS File Calls are comming in browser even i am able to answer that calls,Also in browser it says "In call" but in asterisk CLI it keep showing ringing and other end showing like "remote ringing" . Here is the asterisk logs : http://pastebin.com/e8Ap3bhq Can anyone please let me know what am i doing wrong? -- Thanks, Bhavik Patel
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