[Asterisk-Users] SIP Authentication
Race Vanderdecken
asteriskusers at codetyrant.com
Mon Jun 13 07:52:16 MST 2005
Greetings,
You have stumbled on to one of the most troublesome flag for newbies;
autocreatepeer.
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer
in your sip.conf file add a line in the [general] section
"autocreatepeer=no"
Now people can only use your Asterisk SIP connection if you create a
peer entry for them in your sip.conf file.
Your sip.conf file should be located in "/etc/asterisk" directory.
"cd /etc/asterisk"
"vi sip.conf"
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stojan
Sljivic - GDS
Sent: Monday, June 13, 2005 2:18 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SIP Authentication
Hi,
Does anyone know the solution to this issue?
Regards,
Stojan Sljivic
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stojan
Sljivic - GDS
Sent: Friday, June 10, 2005 13:21
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] SIP Authentication
Hi,
I use SIP softphone that is not registered at Asterisk.
When I dial some extension defined in the dial plan (100 at asterisk-ip)
with my SIP softphone, Asterisk will not ask me for username/password
(will not return response 407) as I expected.
The response 407 - Authentication required will be returned if username
defined in the softphone's setting matches one of the SIP peers defined
in sip.conf.
This means that anyone can dial extension at my Asterisk and that is not
good, since that person could then dial over my ZAP line.
How can I configure Asterisk to allow only peers defined in sip.conf to
register and dial?
Regards,
Stojan Sljivic
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