SIpml5 demo not working with asterisk 11.9.0

1
votes

I am trying to configure an 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?

1
asterisksipml
sipml/webrtc are experemental. It not always working even when used by expert. – arheops

1 Answers

1
votes

If you are going to test webRTC, i successfully tested the Flashphoner Web Call Server with Asterisk 1.8.x versions using different call scenarios. Regarding sipML5, i would suggest you to please try using this tutorial by Sanjay Willy : http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html I hope it will be helpful for you.

Regards,