[Asterisk-Users] Vonage <---> Asterisk Working Config!

[Nitesh Divecha]

Hey Dean,

I followed your instructions and moved forward abit... Before it used to say
SIP 2.0 404 Not Found. But now with your settings I am getting SIP/2.0 407
Proxy Authentication Required.

And yea for some unknown reason the TRUNK name and USER CONTEXT has to be
same don't know why... 

But still no success for inbound calls... 

This is my sip.conf

[vonage]
username=<<VonageDID>>
type=friend
secret=blah
port=5061
nat=yes
insecure=very
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=inband
context=from-pstn
canreinvite=no
auth=md5

[vonage-out]
username=<<VonageDID>>
type=friend
secret=blah
port=5061
nat=yes
host=sphone.vopr.vonage.net
fromuser=<<VonageDID>>
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
auth=md5

Any help would be appreciated.... Thanks Dean!

Nitesh



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of dean collins
Sent: Wednesday, February 23, 2005 1:50 PM
To: nitesh at vipernetworks.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hi Nitesh, check out my config that I have for the Faktortel config in
the asterisk at home sourceforge forum, you'll probably be able to work out
how to set it up from there.


Cheers,
Dean


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 4:12 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Vonage <---> Asterisk Working Config!

Hello All,

I know this question has already been answered but for some reason I am
failing to receive inbound calls.

I got my Asterisk at home 0.6 setup for inbound and outbound and I can make
successful outbound calls. My Asterisk is registered with Vonage
successfully also. 

The only problem I am facing is inbound calls. I enabled the "sip debug"
and
I can see inbound calls coming in but then asterisk destroys it for some
unknown reason. 

Does anyone have a working config for inbound call?

Thanks Jay for all your help

Nitesh


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Content preview:  Hi Nitesh, check out my config that I have for the 
  Faktortel config in the asterisk at home sourceforge forum, you'll 
  probably be able to work out how to set it up from there. Cheers, Dean 

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