On 7 October 2010 10:10, Stefan Schmidt wrote:
> Am 07.10.10 10:52, schrieb Steve Davies:
>> Hi,
>>
>
>
> Hello,
>
> i just want to say something about point 4 which comes to my mind about
> security.
>
>>
>> 4) I am not sure whether it is worth dropping through and testing auth
>> against other
Am 07.10.10 10:52, schrieb Steve Davies:
> Hi,
>
Hello,
i just want to say something about point 4 which comes to my mind about
security.
>
> 4) I am not sure whether it is worth dropping through and testing auth
> against other peers if there is no username match. Can auth ever
> succeed und
Hi,
We have a scenario where we need multiple discrete SIP trunks (peers)
from/to a single endpoint. Because the authentication system starts by
matching IP address, it only ever matches on one of the SIP peer
entries, and ignores the others. This is documented behaviour and as
such is "correct".
Hi,
For IAX there is a fairly clear description of the authentication
process for inbound calls. A similar SIP document used to exist on the
voip-info wiki, but since 1.6.2 has a number of changes, I was
wondering how different (if at-all) 1.6 authentication might be in SIP
over 1.2. or 1.4 versio
my fault
regards
klaus
Klaus Darilion schrieb:
> Hi!
>
> Is it possible to define a peer where authentication is performed only
> on the auth-username in the Proxy-Authorization header? Thus allowing
> From username / auth user mismatch?
>
> thanks
> klaus
>
> __
Hi!
Is it possible to define a peer where authentication is performed only
on the auth-username in the Proxy-Authorization header? Thus allowing
From username / auth user mismatch?
thanks
klaus
___
-- Bandwidth and Colocation Provided by http://www.
Of Stojan Sljivic - GDS
> Sent: Tuesday, June 14, 2005 1:57 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] SIP Authentication
>
> Hi Olle,
>
> Do you have any idea why is Asterisk behaving like this?
> Did you t
TED] 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
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 extensi
2005 9:53
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP Authentication
>
>
> Stojan Sljivic - GDS wrote:
> > Hi,
> >
> > I have set autocreatepeer=no and it behaves just the same.
> > It seems that the d
Stojan Sljivic - GDS wrote:
> Hi,
>
> I have set autocreatepeer=no and it behaves just the same.
> It seems that the default value is no, or Asterisk does not understand
> this property.
> In which version of Asterisk was this property introduced? I use 1.0.5.
>
autocreatepeer is off by default
t: RE:
[Asterisk-Users] SIP Authentication
Race,
Are you saying that
the default is autocreatepeers=yes?
I was under the
impression that the default is no and yes must be explicitly
defined.
Same holds true for
insecure=, default no, optional yes or very.
P
Title: Message
Race,
Are you saying that the default is
autocreatepeers=yes?
I was under the impression that the
default is no and yes must be explicitly defined.
Same holds true for insecure=, default no,
optional yes or very.
Please tell me I am not mistaken so I do
not fe
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
---
Sent: Sunday, June 12, 2005 11:18
PM
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-
Fr
- 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 ([EMAIL PROTECTED]) with my SIP softphone,
Asterisk will not ask me for username/password
Title: Message
Hi,
I use
SIP softphone that is not registered at Asterisk.
When I
dial some extension defined in the dial plan ([EMAIL PROTECTED]) with my SIP softphone,
Asterisk will not ask me for username/password (will not return response 407) as
I expected.
The
response 407 - Authent
I think good practise is
* Not confuse [peer] names in sip.conf with extension numbers
(as you have learned). Extensions will match several peers
some times, and when using trunks, it will cause problems.
It is also much easier if the peer is a name and an extension
is a number. (But I migh
We are using a Cisco router with a T1 card plugged into a PRI provided
by a local telco (Allstream).
This Cisco accepts calls and sends them to a couple of servers running
Asterisk depending on which number was dialled.
But there is a problem.
When a call comes in to the Cisco from the PSTN it s
Hi,
Can anyone tell me how to authenticate against an external
SIP host when dialing out using the dial macro? I get “Forbidden
- wrong password on authentication for INVITE to '"XXX" ;tag=as06c232f1'
Thanks for your time
Guy
___
Kurt Bauer wrote:
Hi,
I have the following setup:
E100P
SER <> * <-> PBX
This works just fine, except when there are users on both boxes (ie. SER
and asterisk), whose usernames are the same, although the realm is
different.
At this point, Asterisk doesn't care about the re
Hi,
I have the following setup:
E100P
SER <> * <-> PBX
This works just fine, except when there are users on both boxes (ie. SER
and asterisk), whose usernames are the same, although the realm is
different.
An example:
user '[EMAIL PROTECTED]' wants to call some extension
At 5:27 PM -0500 on 7/14/04, Andres wrote:
wall with authentication requests and the actual characters inside
the request string - numeric characters seem to cause problems,
while alpha characters do not. I'm running
CVS-HEAD-07/13/04-21:34:00.
Here's the definition of my SER proxy in sip.conf
wall with authentication requests and the actual characters inside the request string
- numeric characters seem to cause problems, while alpha characters do not. I'm
running CVS-HEAD-07/13/04-21:34:00.
Here's the definition of my SER proxy in sip.conf:
[ser-to-tollfree]
type=peer
insecure=yes
h
[wrapping disabled to allow for easier review]
Yet another SIP authentication problem.
I have SER running, and passing calls to a PRI-enabled Asterisk server from a large
range of Media Terminal Adapters, and a few other Asterisk systems set up as
"clients". I have this PRI-enabled Asterisk se
Hi!
Searching at google, I found out that someone had a very similar problem to mine and posted it here under the subject "Outgoing calls to SIP provider".
Unfortunately I couldn't find how the problem was solved - if it was. Is it an Asterisk's bug? Have I done something wrong?
Thanks,
Chu
I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credential
Good day all
How do I get my asterisk and sip to use the password.I'm using x-lite.If
I use just the username and no password it still logs on?
Here is my sip.conf entry?
[101]
type=friend
callerid="Test User" <101>
context = test_1 ; Default context for incoming calls
username=101
secre
Hi,
I don't know whether only we are experiencing this problem but it
seems that if authentication is used on a couple of phones, and then
the authentication is removed (i.e. remove the secret parameter from
each of the extensions), then this isn't reflected in asterisk after
a reload. Instead
Might enter it in the new Asterisk bug tracker
Mark
On Mon, 21 Jul 2003, Tan Aks wrote:
> Hi,
>
> I don't know whether only we are experiencing this problem but it seems that if
> authentication is used on a couple of phones, and then the authentication is removed
> (i.e. remove the secret par
Hi,
I don't know whether only we are experiencing this
problem but it seems that if authentication is
used on a couple of phones, and then the authentication is removed (i.e. remove
the secret parameter from each of the extensions), then this isn't reflected in
asterisk after a reload. Ins
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