Asterisk Setup for Community VoIP

VoIP User Asterisk Setup

The following configuration should allow you to use the outbound service and 3p/min 0844 inbound number from the Asterisk PBX. Obviously you need to replace USERID with your voipuser.org SIP user id, PASSWORD with your voipuser.org SIP password (which is the same as your password for the VoIP User website) and 448441234567 with your 0844 number or whatever you want your incoming extension to be called.

In sip.conf:

register => USERID:PASSWORD@voipuser.org/448441234567

[voipuser]
type=friend
context=incoming_voipuser
username=USERID
secret=PASSWORD
host=voipuser.org
fromuser=USERID
fromdomain=voipuser.org
insecure=very
qualify=no


In extensions.conf:

[incoming_voipuser]
exten => 448441234567,1,NoOp(--- ${CALLERID} calling on VoIPUser (${EXTEN}) ---)
exten => 448441234567,2,Dial(SIP/yoursipphone,20)
exten => 448441234567,3,Answer
exten => 448441234567,4,Wait,1
exten => 448441234567,5,Voicemail(u1)
exten => 448441234567,6,HangUp


To make calls through voipuser.org by dialing #81<number to call>:

[outgoing_voipuser]
exten => _#81.,1,Dial(SIP/${EXTEN:3}@voipuser,60)
exten => _#81.,2,Congestion


If your Asterisk box is behind a NAT router, you may need something like this in the top of your sip.conf otherwise it either won't work or you'll get something funny like 1-way audio (eg: remote end can hear you but you can't hear them):

[general]
nat=yes
externip=123.123.123.123 (replace with your external ip address)
localnet=10.0.0.0/255.0.0.0 (replace with your local network address and mask)
canreinvite=no


You may also want to tune your codec selection depending on your internet connection. If you have a high bandwidth link, use allow=ulaw. If you are low bandwidth, use allow=gsm (or buy some licences from Digium and use allow=g729).

In [general], allow the codecs you want available and in which order.

Then, in your voipuser section, say which codecs you want to use for that connection (order doesn't matter).

Here is what I use - i'm on a 750k/128k cable modem so I use the low bandwidth codec's 1st:

in [general]
disallow=all
allow=g729
allow=gsm
allow=ulaw
allow=alaw
allow=all

in [voipuser]
disallow=all
allow=g729
allow=gsm


If you have any questions or problems using Asterisk with voipuser.org, please feel free to post in this thread and we will try to help.

--ian


[edited 8th Jan 2005 to add nat and codec notes in]
[edited 14th Jan 2005 to add outgoing extension.conf snippet]

I am trying to use VoIP User with *, when i try to dialout any number, it gives me this error

-- Executing Dial("IAX2/uppal@uppal/5", "SIP/voipuser/1905XXXXX") in new stack
-- Called voipuser/19054XXXXX
-- Got SIP response 404 "This is not a valid user at this location" back from 216.127.66.119
-- SIP/voipuser-ab96 is circuit-busy
== Everyone is busy/congested at this time
ns1*CLI>

Heres my enteries in * config files.
---------------------
SIP.CONF
---------------------
register => uppal:XXXXX@voipuser.org/08449XXXXX

[voipuser]
type=friend
context=incoming_voipuser
username=uppal
secret=PASSWORD
host=voipuser.org
fromuser=uppal
fromdomain=voipuser.org
insecure=very
qualify=no

----------------------------------------

--------------------
extensions.conf
--------------------
exten => _55.,1,Dial(SIP/voipuser/${EXTEN:2})


Kindly help me out

thanks

~uppal

Hi,

uppal :

"SIP/voipuser/1905XXXXX") in new stack -- Called voipuser/19054XXXXX -- Got SIP response 404 "This is not a valid user at this location" back from 216.127.66.119

That means you dialed an invalid number. I dont know if Dean mentioned this in the outbound info but you need to dial the number in UK format - i.e:
- a uk number: 01234 123456
- a non-uk number 00 country_code number: eg 00 1 123 456 7890

uppal :

Executing Dial("IAX2/uppal@uppal/5"

--ian

Quote:

you need to dial the number in UK format

Thanks I knew there was something I forgot to mention - will add it now.

Dean

Iax's mentioned because i am using a iax client with my * setup.

Cheers

~uppal

Hi Dean,

Just an idea....is it possible for the outbound proxy be setup to use support different dialing formats? - i.e if you're setup for uk outbound you dial in uk format, but if you are setup for US then you dial in us format?

--ian

ichilton :

If your Asterisk box is behind a NAT router, you may need something like this in the top of your sip.conf otherwise it either won't work or you'll get something funny like 1-way audio (eg: remote end can hear you but you can't hear them): [general] nat=yes externip=123.123.123.123 (replace with your external ip address) localnet=10.0.0.0/255.0.0.0 (replace with your local network address and mask)

Hi,
Both my * server and SIP client are behind the same NAT-firewall. I do not have 'nat=yes' anywhere in my sip.conf, but I have both externip and localnet lines, and put 'canreinvite=no' for my SIP client to solve the 1-way audio problem.
I realized this when I tried an IAX client to place a call and audio worked fine. * had to stay in the middle to translate the IAX-SIP call, but with SIP-SIP calls, it was trying to let the two talk directly.
Which is the 'proper' way, nat=yes or canreinvite=no? Does it really make a difference?

Hi,

I use both nat=yes and canreinvite=no.

You're right, canreinvite=no stop's the phone trying to talk directly to the gateway and cutting asterisk out of the middle. Doing this helps in a nat setup because you dont have to worry about the firewall and port forwarding for your phone - just getting it right for asterisk.

--ian

Hey

I have just come to set this up in asterisk and I have used the following in extensions.conf

exten => _55X.,1,Dial(SIP/voipuser/{$EXTEN:2})

but i am still getting

-- Executing Dial("SIP/1000-5d28", "SIP/voipuser/{$EXTEN:2}") in new stack
-- Called voipuser/{$EXTEN:2}
-- Got SIP response 404 "This is not a valid user at this location" back from 216.127.66.119
-- SIP/voipuser-7c9e is circuit-busy
== Everyone is busy/congested at this time

when dialling 55016174847xx from my sip phone, can someone shed some light please?

riksta :

exten => _55X.,1,Dial(SIP/voipuser/{$EXTEN:2})

Try changing that to this:
exten => _55X.,1,Dial(SIP/{$EXTEN:2}@voipuser)

If that doesn't work, please post the output again and i'll try to help.

--ian

Hi, wow thanks for the quick response.

However, i tried adding the new extension and reloading asterisk, but i still get exactly the same error message

I have tried dialling the number in all possible formats including:

0044161...
44161...
0161...
161...

but to no avail....any suggestions please?

thanks again

Hi,

hummm....

Please could you try dialing it as 0161xxxxxxx (you should dial through voipuser.org as if you are in the UK) and paste the full output?


Thanks

--ian

-- Executing Dial("SIP/1000-7d64", "SIP/{$EXTEN:2}@voipuser") in new stack
-- Called {$EXTEN:2}@voipuser
-- Got SIP response 404 "This is not a valid user at this location" back from 216.127.66.119
-- SIP/voipuser-99e4 is circuit-busy
== Everyone is busy/congested at this time
pbx*CLI>


thats the output i get when dialing the 0161

Sorry, I made a mistake:

It's ${EXTEN:2}, not {$EXTEN:2} (I copied that bit from your post and didn't fix it )

--ian

ahh yeah of course, shoulda picked up on that

now im getting

-- Executing Dial("SIP/1000-bb20", "SIP/016174847xx@voipuser") in new stack
-- Called 016174847xx@voipuser
-- Got SIP response 488 "Not Acceptable Media" back from 216.127.66.119
== No one is available to answer at this time