I copied a setup from an older 1.8.5 installation to an 11.15 installation, and Im having problems getting call files to work.Here is the extension setup Im using:[outbound-swift]exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure);ex..
Im reading the OReilly Asterisk the definitive guide, 4th ed, with a starfish on it.In some ways, astonishing that its not really that definitive, its more general — and it only clocks in at one ream of paper!In any event, Im having some port probl..
I am working with 188.8.131.52 which I have patched with t38-gateway and PRACK. t38 is tested and working fine with Zoiper client but I cant get the t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found FAXCOM announces that it suppo..
Guys We have a client running on a polycom vvx400 IP phone on our asterisk1.8.18 system The issue we have is the switchboard lady uses ## to transfer calls but sometimes it just does not work and just plays the DTMF tone to the calling party. Is th..
list. We have a problem with loss peers after sip reload, our configuration:Asterisk 11.6-cert1, SIP realtime peers, sip.conf: – rtcachefriends=yes- rtsavesysname=yes- rtupdate=yes- rtautoclear=yes When we do sip reload , peers are removing from availab..