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[testcarrier]
disallow=all
allow=ulaw
type=friend
username=testcarrier
secret=test
host=dynamic
dtmfmode=rfc2833
context=trunkinbound
[sip.switch2voip.us]
type=peer
username= USERNAME
host=sip.switch2voip.us
fromuser= USERNAME
canreinvite=no
reinvite=no
secret=PASSWORD
[general]
bindport = 5060
bridge=no
reinvite=no
canreinvite=no
bindaddr = XXX.XXX.XXX.XXX ; (Your server IP)
context = main
disallow=all
allow=ulaw
allow=g729
disallow=all
dtmfmode=auto
tos=184
type=friend
defaultexpirey=3600
useragent=Asterisk PBX
[switch2voip]
disallow=all
allow=ulaw
type=friend
username=USERNAME
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
context=trunkinbound
exten => s,1,Ringing
exten => s,2,Wait(3)
exten => s,3,Answer
exten => s,4,DigitTimeout,5
exten => s,5,ResponseTimeout,2
exten => _X.,1,Dial(SIP/${EXTEN}@XX.XX.XXX.XXX)
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(newsip:test@10.10.10.15:5060/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup
aemae wrote:I need help configuring VICIdial to use the webphone, and help configuring the set up of the survey.
aemae wrote:Thats it, I dont see why I would pay for the help when it appears to be offered regularly here on the forum (thats the purpose of the support section to my knowledge)
[switch2voip]
disallow=all
allow=ulaw
type=friend
username=USERNAME
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
context=trunkinbound
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(USERNAME:PASSWORD@XX.XX.XX.XX:5060/${EXTEN:2},,tTor)
exten => _91NXXNXXXXXX,3,Hangup
Outgoing Settings: (To place calls)
username={ACCOUNT NUMBER}
type=peer
secret={PASSWORD}
progressinband=never
port=5060
nat=auto
insecure=very
ignoresdpversion=yes
host=sip.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=g729&g711&g723
username={ACCOUNT NUMBER}
user={ACCOUNT NUMBER}
type=user
port=5060
dtmfmode=rfc2833
pogressinband=never
context=from-pstn
allow=g729&g711&g723
register=>{ACCOUNT NUMBER}:{PASSWORD}@sip.switch2voip.us
[Switch2Voip]
type=peer
username={ACCOUNT NUMBER}
host=sip.switch2voip.us
fromuser={ACCOUNT NUMBER}
canreinvite=no
reinvite=no
secret={PASSWORD}
disallow=all
allow=g729
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@Switch2Voip,,o)
exten => _91XXXXXXXXXX,3,Hangup
Ok, so as of now, I can get the Vicidial to successfully call my softphone, and it says "you are the only one in the conference". Can I now assume that my phone and carrier settings are working normally?
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