[asterisk-users] Asterisk 11.9 with webRTC demo integration

Rusty Newton rnewton at digium.com
Wed May 14 11:28:36 CDT 2014

On Sat, May 10, 2014 at 2:27 AM, bhavik patel
<bhavikpatel14388 at gmail.com> wrote:
> Hi All,
> 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".

That is a question about Chrome, not about Asterisk.  A quick Google
search pulls up this information:


" If you select Allow on a "http" URL your preference will not be
remembered in future visits. If you select Allow on a "https" URL,
your preference will be remembered in future visits. "

> 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" .

Not sure what is going on here. You can try following my tutorial for
testing with the SIPML5 demo here:
, It also uses Asterisk 11 and chan_sip which matches what you are

Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

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