Hello,
We have numerous outside numbers that need to be routed to specific extenstions. These are not DID numbers, and the internal extension may change a few times a year. Any help would be great, I’m very new to Asterisk and the task of setting it up/making changes it just fell in my lap.
thanks !
I assume that by “outside numbers” you mean a connection from a PSTN line to an Asterisk system. If that’s the case, you’ll need some sort of ATA like those sold by Digium or something like the Linksys SPA3102 (although that’s only for a single line, I think there are others that allow multiple lines but am not sure as I’ve never used them).
From there you should be able to find plenty of information on connecting a PSTN line to an Asterisk setup by Googling.
To clearify,
These numbers are already in use, they are just routed to the wrong internal phone extension.
I guess I just need to know where to find the file on Asterisk, and the file name to make changes to, so these calls are routed to a different extension. I hope that makes sense.
What type of connection are your outside numbers? You should find where that channel is defined in /etc/asterisk/sip.conf (or whatever). Then you need to figure out what context handles the incoming calls (look at the "context= " line) and change the extension being dialed in /etc/asterisk/extensions.conf. For example if your incoming calls were SIP and you had it defined in sip.conf, the context in the dialplan might look like this:
[outside-number-context]
exten => 1000,1,Dial(SIP/1000)
…you’d change “1000” to whatever extension you want.
here is a copy of my sip.conf…
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=ulaw
relaxdtmf=no
dtmfmode=rfc2833
canreinvite=no
qualify=no
;jbenable=yes
t38pt_udptl=no ; Default false
context=from-trunk
;context=closedfire1
tos_sip=cs3
tos_audio=ef
progressinband=never
subscribecontext=ext-local
;limitonpeers = yes
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
Looks like what you need would be in one of the included files; sip_nat.conf, etc.