Followme is perfect to handle FMFM and it is now also realtime, but it seems impossible to assign some value to a variable, from within the followme to store info for example about the tenant the followme is running under, like instead happen for exam..
, someone has successfully change different ringtone from dialpan with asterisk with this model Granstream?for example:exten => 0,1,Playback(pls-wait-connect-call)same=> n,SIPAddHeader(Alert-Info:;info=ring3)same=> n,Dial(SIP/310&SIP/318,30,t)can ..
This is driving me to distraction.I have a switch with multiple clients who are all working fine except for one and I cant figure out what makes them different.I have tried every NAT setting in the ATA(SPA112 ATA with 2 x FXS, 1 x LAN), stun server..
Im testing Asterisk at home, crummy connection.Skype works fine for me, but every SIP client, even without using Asterisk, fails to connect.Thats ok.Is swapping out SIP for Skype a big deal?Heh, well, I guess its dead:http://www.digium.com/en/products/software/skype-for-asteris..
Can anyone recommend a particular online WebRTC phone for testing with Asterisk?We tried:- JsSIP, but even with the enable video checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with Rejecting secure video stream with..