Archives : March-2015
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..
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .the problems that i faced with this is the following and i hope i could get an adv..
list,i use chanspy with the code below[app-chanspy]exten => _007.,1,Macro(user-callerid,)exten => _007.,n,Answer exten => _007.,n,Authenticate(1111)exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)exten => _007.,n,Hangupi have a question related to chans..
Anyone know where it’s gone?.. Appears to have been down all day…
Please favorite the following bug and express desire to have wav49VoiceMails play natively in Google Desktop Chrome HTML5. https://code.google.com/p/chromium/issues/detail?idF5431.WAV already plays in Android Chrome and Chromebook and others, but ..
Hello.Asterisk 13.2, PJSIP.Problem: I do not get any AMI events when changing the status of the contact.When using chan_sip I got peerstatus event. When using res_pjsip and devices (endpoint configuration) I got peerstatus event. When using res_pjsip..