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InnovateAsterisk / Browser-Phone

A fully featured browser based WebRTC SIP phone for Asterisk
https://www.innovateasterisk.com
GNU Affero General Public License v3.0
471 stars 240 forks source link

Webrtc to SIP calls Issue #470

Open karthik0112 opened 9 months ago

karthik0112 commented 9 months ago

I have cloned your code and it's working fine in my local. But In my dev server, there is an issue while making calls. Sip to Sip calls work fine whereas (Sip to webrtc calls or webrtc to sip calls) is causing an issue. I'm able to make a call from both sides while picking up the calls it's disconnected. could you give me a solution to sort it out?

InnovateAsterisk commented 9 months ago

You will have to check the Developer Console to see what the error is. Please post it here.