webrtc2sip

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Integration with freepbx or fusionpbx

1

I am trying to understand what it is. I have just installed freepbx and I was configuring outbound routes and I came to know that I would need a SIP...

ugintl

How to compile webrtc2sip correctly?

1

when I run 'make' of "Building webrtc2sip export PREFIX=/opt/webrtc2sip cd webrtc2sip && ./autogen.sh && ./configure --prefix=$PREFIX make clean && make && make install cp -f ./config.xml $PREFIX/sbin/config.xml" there are some...

XDhughie

Hi!! I was using webrtc2sip in a DigitalOcean's droplet without any problema but now i just made a new a new droplet taking a snapshot of the first to create...

llKoull

Log feature: * All log will append to file * Separate Application Log for tracking SIP message Stat feature: * count incoming and outgoing message * count number of connections...

pepsi7959

Hello, I have working webrtc2sip server with my PBX. But I wonder if there is possibility to use 302 Moved Temporarily message to redirect call from js app to sip...

wieczorekmichal1

Via header not stripped by webrtc2sip in 200ok response

3

``` What steps will reproduce the problem? 1. REGISTER with a sipml5 endpoint through webrtc 2. 200ok response contains an extra Via header (not stripped by webrtc2sip) Via: SIP/2.0/UDP 92.11.169.63:10060;rport=10060;received=92.11.169.63;branch=z9hG4bK0AP8Pm9NAxnT3...

GoogleCodeExporter

Problem in BYE send from Webrtc2sip (Version 2.6.0)

1

``` What steps will reproduce the problem? 1. Make a Call from Sip Client (softphone registered on a PBX) 2. Call is Answer in SIPML5, next Hangup the Call in...

GoogleCodeExporter

Got a problem with calls

1

Hello. When I try to call I see this errors and nothing happens, Please hellp me solve this problem... __**[DOUBANGO ERROR]: function: "tnet_sockfd_sendto()" file: "src/tnet_utils.c" line: "1814" MSG: sendto(fd=3) failed...

praeitor

Correct digest authentication call flow for authentication of INVITEs at the network

10

``` Authentication fails, if the SIP domain configured in SIPML5 does not match the authentication realm of the network. Furthermore, WebRTC2SIP expects the ha1 used for digest authentication in a...

GoogleCodeExporter

Dear all, I've setup webrtc2sip and SIPml5 using Asterisk as SIP server. All works well but I'm confused about the CPU utilization, also when I make a video-call between two...

epasqualotto