I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue.My setup is as follows:Server:CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146asterisk-11.2..
Guys,I am able to divert a incoming phone call from asterisk to a sip softphone. Is it possible to redirect a call to a serial port? If so how would I do it? I dont mind a brief explanation. There is a ppp/dialup server listening on serial port…
In make menuselect =>application=>XXX app_meetme . I am doing confrence call using sip softphone.
I checked It Depends on: dahdi(E) .
How I can do app_meetme enable?
Hello. I would need some help trying to setup Asterisk 126.96.36.199-2+squeeze1 on a Debian 6.0 system. Id like to use the Debian packages, hence the strange version number… Since Im new to Asterisk, Im trying to follow The Asterisk Book at http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.h..
list, I have an Asterisk setup with the following details: 1. 3 x internal extensions / sip hardphones – Grandstream 2000 2. 2 x internal extensions / dahdi cordless phone 3. 1 x 2 FSX ports OpenVOX pci card 4. 1 x internal sip extension / sip softph..
Let me explain: When I dial into Asterisk ( I have a SIP trunk – which I need to make sure is not faulty), I only get one-way voice communication. The calling party, from the SIP trunk hears nothing – the extension rings on the Asterisk server (you ..