I’m trying to set up a simple SIP inbound system that is managed with adhearsion.
I don’t have any VoIP/softphones/devices that I’m connecting to asterisk, i just want to use an application to interface the incoming call. I don’t particularly even want to dial out. I have looked up documentation for the past 2 days, and I don’t think I’m really understanding the configurations. I have an SIP account that registers fine, but I when I dial into asterisk from my cellphone, the call won’t complete and I get an error.
[color=#FF0000]Error: [Apr 7 19:44:01] NOTICE: chan_sip.c:19546 handle_request_invite: Call from ‘74275*****’ to extension ‘30355*****’ rejected because extension not found.[/color]
I’ve tried a bunch of different configurations but obviously i’m missing it.
Right now I would just like to call in, and have a soundfile play or something to verify that incoming calls work.
these are my current configurations.
dtmfmode = rfc2833
register => 30355*****:email@example.com/30355*****
; I briefly had this section in user.conf before i started getting this error
; app_macro.c:302 _macro_exec: No such context ‘macro-stdexten’ for macro ‘stdexten’
[general] static=yes writeprotect=no ; i added this after the fact, I realize it's probably completely useless, ; does asterisk reserve any context names? [default] exten => s,1,Verbose(Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [from-pstn] exten => s,1,Wait(1) exten => s,2,Answer exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => s,5,Background(demo-congrats)
Sorry for the nubcake but any help is appreciated.