asterisk users - Aug 2005 - Asterisk and Vonage - Can't call out but can receive calls

Hi,

I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me
to place and receive
calls.  I have successfully configured Asterisk to route inbound calls and send
them to the correct extension, but I can't get outbound calls to work.  I
have
Asterisk successfully registering with Vonage, but when an INVITE is sent out, I
get a "404 Not Found" back from Vonage and thus Asterisk shows

    "SIP/atlas-east.vonage.net is circuit-busy"
    
Below are my sip.conf and extensions.conf files that I am using.  I saw several
other sample configuration
files that claimed connectivity to Vonage, but I have not been able to get them
to work.  My Asterisk server is behind a NAT firewall.
The files below are the closest I have gotten to complete connectivity.  Any
feedback is appreciated!

One note:
I was able to receive calls only if I used atlas-east as the Vonage server.  If
I used sphone.vopr.vonage.net, no calls came in.


sip.conf:
[general]
externip=X.X.X.X
port=5060
bindaddr=X.X.X.X
context=vonage-out
disallow=all
allow=ulaw
allow=alaw
nat=yes

register=####:PASSWORD@atlas-east.vonage.net:5060/201

[vonage]
username=####
type=peer
secret=PASSWORD
port=5060
nat=yes
host=atlas-east.vonage.net
fromdomain=vonage.net
canreinvite=no
fromuser=####
dtmfmode=rfc2833
context=vonage-out

[201]
type=friend
username=201
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
defaultip=X.X.X.X
mailbox=201
callerid=NAME
progressinband=no
context=from-sip

extensions.conf (relevant part)
[vonage-out]
exten => ####,1,Goto(from-sip,201,1)

[from-sip]
exten => _9.,1,Dial(SIP/${EXTEN-1}@atlas-east.vonage.net)
exten => 201,1,Dial(SIP/201)
exten => 202,1,Dial(SIP/202)



Thanks,
-- Brian

Brian Deep wrote:
> 
> [from-sip]
> exten => _9.,1,Dial(SIP/${EXTEN-1}@atlas-east.vonage.net)
> exten => 201,1,Dial(SIP/201)
> exten => 202,1,Dial(SIP/202)
try exten => _9.,1,Dial(SIP/${EXTEN:1}@atlas-east.vonage.net)

You are calling a sip host that you do not have defined in sip.conf.

I think the line should look like this

exten => _9.,1,Dial(SIP/${EXTEN:1}@vonage)

This will force * to look in its sip.conf file for a stanza called 
[vonage] which you have rather than [atlas-east.vonage.net] which you 
don't have.

Brian Deep wrote:
> Hi,
> 
> I'm trying to get Asterisk to connect to Vonage (softphone acct) to
allow me to place and receive
> calls.  I have successfully configured Asterisk to route inbound calls and
send
> them to the correct extension, but I can't get outbound calls to work. 
I have
> Asterisk successfully registering with Vonage, but when an INVITE is sent
out, I
> get a "404 Not Found" back from Vonage and thus Asterisk shows
> 
>     "SIP/atlas-east.vonage.net is circuit-busy"
>     
> Below are my sip.conf and extensions.conf files that I am using.  I saw
several other sample configuration
> files that claimed connectivity to Vonage, but I have not been able to get
them
> to work.  My Asterisk server is behind a NAT firewall.
> The files below are the closest I have gotten to complete connectivity. 
Any feedback is appreciated!
> 
> One note:
> I was able to receive calls only if I used atlas-east as the Vonage server.
If
> I used sphone.vopr.vonage.net, no calls came in.
> 
> 
> sip.conf:
> [general]
> externip=X.X.X.X
> port=5060
> bindaddr=X.X.X.X
> context=vonage-out
> disallow=all
> allow=ulaw
> allow=alaw
> nat=yes
> 
> register=####:PASSWORD@atlas-east.vonage.net:5060/201
> 
> [vonage]
> username=####
> type=peer
> secret=PASSWORD
> port=5060
> nat=yes
> host=atlas-east.vonage.net
> fromdomain=vonage.net
> canreinvite=no
> fromuser=####
> dtmfmode=rfc2833
> context=vonage-out
> 
> [201]
> type=friend
> username=201
> secret=PASSWORD
> host=dynamic
> dtmfmode=rfc2833
> defaultip=X.X.X.X
> mailbox=201
> callerid=NAME
> progressinband=no
> context=from-sip
> 
> extensions.conf (relevant part)
> [vonage-out]
> exten => ####,1,Goto(from-sip,201,1)
> 
> [from-sip]
> exten => _9.,1,Dial(SIP/${EXTEN-1}@atlas-east.vonage.net)
> exten => 201,1,Dial(SIP/201)
> exten => 202,1,Dial(SIP/202)
> 
> 
> 
> Thanks,
> -- Brian
> 
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-- 

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com