I have a question, I have installed a tdm400, to use a voip service it is needed to use an ata? i have an analog phone, if it is connected just to FXS port is enough? or I really need the ATA to use it in the VOIP? please help me…
If you plug your phone in your FXS port you do not need a ATA box !
ok, help me alexis I have voip service (junction network), and I configured as its web page says, but it doesnt work. some example to follow it and be able to make calls in voip?, please help!!..
1- Tell us what does’nt work
2- Your Voip provider is SIP or IAX
3- Paste your conf file here
it is the extension.conf configuration, it is as the junction network web page says
[outgoing]
exten => _1NXXNXXXXXX,1,Dial(IAX2/jnctn_out/${EXTEN})
exten => _1NXXNXXXXXX,2,Congestion()
exten => _1NXXNXXXXXX,102,Busy()
[incoming]
exten => _15105501403,1,Playback(beep)
exten => _15105501403,2,SayDigits(${EXTEN})
exten => _15105501403,3,Goto(testdtmf|s|1)
[testdtmf]
exten => s,1,Background(beep)
exten => s,2,ResponseTimeout(5)
exten => s,3,WaitExten(10)
exten => _x,1,SayDigits(${EXTEN})
exten => _x,2,Goto(testdtmf|s|1)
exten => i,1,Goto(testdtmf|s|1)
exten => t,1,Hangup
It is iax.conf configuration,
[jnctn]
type=user
auth=rsa
inkeys=jnctn
context=incoming
[jnctn_out]
type=peer
host=iax.jnctn.net
username=xxxxxx
secret=xxxxxx
I dont know how to do that it works, I am stick on this…HELPPPP!!!
Like i Said … Tell us what is your problem, Asterisk is big and you can have many problems …
Ex: - I don’t receive any Inbound/Outbound call
- The extension is ringing but i have no audio
- My server explose each time i receive a call
- I see that extension is ringinig in console but the phone don’t ring
Help us help you!