[asterisk-users] Asterisk 11.9 with webRTC demo integration
ohjelmistoarkkitehti at gmail.com
Wed May 14 11:59:35 CDT 2014
I'm far from being an expert, but as far as I know when you use https in
your website the browser will ask to use the audio devices only once and
then remembers your decision. When using http it will ask every time.
Sorry I can't be of more help but hope this helps.
2014-05-10 10:27 GMT+03:00 bhavik patel <bhavikpatel14388 at gmail.com>:
> Hi All,
> I am trying to configure webRTC phone example for SIPml5 and i found this
> info from
> 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
> For asterisk, I made some configuration like below. Please check :
> 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?
> Bhavik Patel
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