Newbie needs help setting up communication

Asterisk Asterisk Support
1

I have installed * 1.2.13 it’s running.

there is a demo with the install that I don’t know whether
needs to be commented out.

this is what I have setup under sip.conf
to register my server, local net and the registration.

[general]

externhost=xxx.xxx.xxx.xxx server IP
localnet=xxx.xxx.xxx.xxx/255.255.255.0 local net
nat=yes

;Voip.les.net registration bellow:
register => xxxxxxxxxx:password@did.voip.name.net/xxxxxxxxx

Lower in the same file I setup my peer between two sip proxis as follows.

[lesnet_peer]
type=friend
host=did.voip.name.net
dtmfmode=rfc2833
insecure=very
disallow=all
allow=ulaw
context=name-incoming ; incoming DID calls will arrive in the name-incoming context

That’s it under sip.

there is a demo setup in extensions.conf that I left as it is and

In extensions.conf I have set up the following under example main menu.

[custom-a2billing]
include => outbound-allroutes
exten => s,1,Answer
exten => s,2,Wait,2
exten => s,3,DeadAGI,a2billing.php
exten => s,4,Wait,2
exten => s,5,Hangup

That’s it, all I want to do is when the DID number is dialed to communicate with a2billing.

Thanks in advance for the help.

2

I had a similar issue at my prev. job, where we were using les.net for DIDs. I had to create a guest IAX user in order to make it work. I’m not sure about SIP (havent used les.net with SIP), but you can give it a try.

3

Thank you. yes I have used sip without a problem. I’m trying to get this to work and then I’ll try iax which is what Iwant to use.

How about the setup I listed. is there anything else I need to do or is there something you noticed I did wrong?

I was using trixbox before and decided to eliminate the midle man and learn how to use it directly.

Thank you.

4

Well if Les is sending the SIP calls directly to the IP address like the IAX calls I doubt you can use it behind nat. You can try forwarding the SIP port to the * machine though. And also you’re sending the incoming calls to “name-incoming” (les-incoming I guess) but you havent posted this context here. Try setting the sip debug to on in the console and see if the call reaches your server.