[asterisk-users] Pass variable to voicemail script

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? --

[asterisk-users] Ast under CentOS 7 - slice messages

I'm building a CentOS 7 Asterisk and find my system log full of messages like this: Mar 5 17:07:01 pbx2 systemd: Started Session 823 of user asterisk. Mar 5 17:07:01 pbx2 systemd: Starting Session 823 of user asterisk. Mar 5 17:07:11 pbx2 systemd: Removed slice user-1001.slice. Mar 5

[asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip. In my snom 760 the setup for these two accounts is identical. When I call echo test from the account using chan_sip audio comes through fine. When I call echo test from the account using pjsip there is no audio. With