[Asterisk-Users] SIP Authentication

Stojan Sljivic - GDS stojan.sljivic at gdspartners.com
Tue Jun 14 05:56:36 MST 2005


Hi Olle,

Thanks for the info.
I have put sip.conf [general] section's context=default.
In default context I hang-up all calls.
This way only peers registered in sip.conf with context=sip will go to the
sip context that will allow them to call some extensions.

However now I have a problem with Vonage. I cannot get any call from Vonage
now since all calls from Vonage go to the [default] context, where they are
hanged-up.

My sip.conf has these lines for Vonage registration:

register => XXX:XXX at atlas-east.vonage.net:5060

[vonage]
type=friend
username=XXX
secret=XXX
host=atlas-east.vonage.net
port=5060
allow=all
maxexpirey=15
dtmfmode=rfc2833
fromuser=XXX
fromdomain=atlas-east.vonage.net
context=sip
nat=yes

How should I configure Asterisk to allow calls from Vonage?

Regards,
Stojan Sljivic 


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Olle E. Johansson
> Sent: Tuesday, June 14, 2005 9:57
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP Authentication
> 
> 
> Stojan,
> You have to check what extensions you have enabled in the 
> context specified in the sip.conf [general] section. All of 
> those will be reachable without authentication by anyone.
> 
> If there's no "context=" setting, Asterisk defaults to the 
> context named "[default]" in your dialplan. Any extensions in 
> the [default] context is in that case reachable without 
> authentication.
> 
> I hope this is solving your problem. If not, to get help with 
> SIP configuration issues you need to provide readers of the 
> list with your configuration and in some cases also SIP debug 
> output. For SIP debug output, we always need you to turn 
> verbosity and debugging levels to a maximum so we really see 
> what is going on inside your Asterisk box.
> 
> Regards,
> /Olle
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