Hi,
I am completely new to this and don't understand a thing. I set up a server with Goautodial on it yesterday. I just want some help getting this working. I think I am having trouble with the dialplan settings.
I managed to get it to call out but only by doing a manual call, this was after mucking about with different settings but cannot remember what they were now.
I am now just getting "I am sorry, that is not a valid extension". I probably really need help in understanding (simply) what the dialplan information actually does. i.e. what each part of it is. I have searched online but none of it is is layman's terms.
I am hoping someone on this forum knows how to set this up for 2talk in New Zealand.
here is the configuration that was automatically populated:
register => 092812860:*mypassword*@peering.2talk.co.nz:5060/092812860
[2talk2talk]
disallow=all
allow=gsm
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
host=peering.2talk.co.nz
username=092812860
secret=*mypassword*
allow=alaw
exten => _4380812233.,1,AGI
exten => _4380812233.,2,Dial(SIP/${EXTEN:10}@2talk2talk,,tTo)
exten => _4380812233.,3,Hangup
If I put the following as the dialplan:
exten => _X.,1,AGI
exten => _X.,2,Dial(SIP/092812860${EXTEN:1}@2talk2talk,,tTo)
exten => _X.,3,Hangup
2talk disappears as an option for "Carrier to use for this Campaign:" But it does not give me the "that is not a valid extension" error.
I don't know if the following helps, but it is what the 2talk has provided regarding connecting to asterisk:
One of the most common IP-PBX platforms in use today is Asterisk (or one of it’s variants such as TrixBox). Below is an example ‘SIP peer’ definition for Asterisk allowing you to route calls to 2talk via your SIP trunk:
[2talk]
type=friend
context=from-trunk
host=trunk.2talk.co.nz
dtmfmode=rfc2833
insecure=very
nat=never
qualify=no
canreinvite=no
disallow=all
allow=gsm
allow=alaw
This is only an example. You will need to decide which codecs and context you wish to use in your own setup. You may then route calls out through your 2talk trunk by adding a directive such as the one below into your extensions configuration:
exten => _X.,1,Dial(SIP/2talk/${EXTEN},,T)
2talk New Zealand - Help with Dialplan