Asterisk from external ip address


#1

I am trying to connect to my office asterisk box from home.
i can get the office phone to call my home phone and it works great voice both ways, all the phone features work, ie on hold transfer.
But i can not dial out, just keep getting engaged tone. any ideas.
my office rooter is on dmz and pointing to my aah box and at home dmz pointing to my phone. I am using Sip.

Sip.conf
[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=gsm
allow=ulaw
context = from-sip-external ; Send unknown SIP callers to this context
;callerid = Unknown
;externip= XXX.69.243.100
;localnet= 192.168.1.1/255.255.255.0

sip_additional.conf
[401]
username=401
type=friend
secret=1234
qualify=no
port=5060
pickupgroup=
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=
context=from-internal
canreinvite=no
callgroup=
callerid="Andrew <401>
allow=
permit=xx.144.231.45


#2

[quote]I am trying to connect to my office asterisk box from home.
i can get the office phone to call my home phone and it works great voice both ways, all the phone features work, ie on hold transfer.
But i can not dial out, just keep getting engaged tone. any ideas.
my office rooter is on dmz and pointing to my aah box and at home dmz pointing to my phone. I am using Sip. [/quote]

Can you rephrase your question? I would try and help you but it is hard to understand what you are asking.


#3

Thanks
All the Sip phones work in the office no problems, But when i connect from home to the AAH server over the internet, i can not dial from the ext of the home phone, it receives all calls, and can put the call on hold etc.

Are you in the uk, if so do you have any chat program so we can talk?

many thanks


#4

its a bit hard to understand what you are saying.

Could it be that your asterisk server is behind some kind of NAT or firewall ?
If so, you will need to do some portforwarding on the firewall, otherwise this will not work.

You at least need to portforward port 5060, rtp ports might not be needed if you use externip and localnet, if you dont use those you will need to portforward all RTP ports as configured in RTP.conf

I am working on a small tutorial for asterisk with NAT and SIP, but its not finished yet. I put a small preview online at

asteriskguru.com/natut.php

Dunno if it would be of any help to you.


#5

Basically I have a aah server running in the office with 4 SIP phones attached, all is working great no problems. The aah server is running on a sdsl line, The sdsl rooter has 4 ip address. I have setup DMZ on 1 of the ip address and it is pointing to the aah server.

At home I have a sip phone swissvoice ip10 connected to adsl, again dmz pointing to the phone. From the office I can phone the extension number of the phone and it rings, pick it up and you can talk no problem,

When It comes to dialing out it doesn’t work.

Can you help?


#6

What is Asterisk reporting on the console when you try and make the call? Is the phone in a context that knows how to get out?


#7

nothing happens on the console when you try and make the call?
how do i add the phone in a context so it knows how to get out?


#8

If you do not see anything in the console when you make the call then Asterisk is not getting the call. I would double check the sip configuration on your phone.