hello all,
first of all let me apologize for any “silly questions” i might have though im a newbie concerning asterisk so bear with me
ive setup asterisk a few days ago…
i have 4 Vonage lines who i have analog access on…
and 1 PSTN line…
i’ve successfully tested the SIP where i could call from 1 extension to another as soon as i add those extensions in both Sip.conf and Extensions.conf
though the thing is i cant call out! and thats obviously the main idea i got into this from the first place…
ill provide a list of configurations ive made so far… as well as the error am getting in CLI when i try to dial out…
[quote]
Extensions.conf:
[general]
static = yes
writeprotect = no
autofallthrough = no
clearglobalvars = no
