Sample Asterisk config for StanaPhone

This works for me. All calls to my StanaPhone number ring on my extension 101, and after 20 seconds goes to my * voicemail. To use the StanaPhone network I dial 9 and the number.

sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
allow=ulaw

; This section is because i’m behind nat
externip = 206.45.88.212 ;Outside address
localnet = 192.168.2.10 ;Inside address
localmask = 255.255.255.0 ;Inside subnet

context = sip ; Default context for incoming calls
register => stanaphonenumber:[email protected]/101

[stanaphone]
type = friend
username = Stanaphonenumber
secret = Stanaphonepassword
host = sip.stanaphone.com
context = sip
nat = yes
canreinvite=no ; for NAT, but it will eat up TWICE the bandwidth because everything will go through *
insecure=very ; this will prevent * from sending a SIP 407 error back when a call comes in

[phone1] ;x-lite
type=friend
username=phone1
secret=12345678
host=dynamic
defaultip=192.168.2.199
dtmfmode=rfc2833
context=intern
callerid=”phone1″ <101>
mailbox=101

extensions.conf

[general]
static=yes
writeprotect=no

[globals]
STANAPHONEUSERID=stanaphonenumber

[intern]
include => stana-out
include => sip

[stana-out]
; STANAPHONE – Dial 9 & number to use stanaphone
exten => _9.,1,SetCallerID(${STANAPHONEUSERID})
exten => _9.,2,Dial(SIP/${EXTEN:1}@sip.stanaphone.com,60,tr)

[sip]
; phone1 – x-lite
exten => 101,1,Dial(SIP/phone1,20,tr)
exten => 101,2,Voicemail(u101)
exten => 101,3,Hangup
exten => 101,102,VoiceMail(b101)
exten => 101,103,Hangup

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For me this above did not work, I was getting “Failed to authenticate on INVITE to” or similar messages,

General SIP configuration

• sip.conf: Configure SIP channels