[Asterisk-Users] Guest

Wed May 25 08:47:16 MST 2005

If I understand correctly what you are trying to do, I would suggest you set a default context in your SIP.CONF file.  That context would be an entry point into the dialplan that is secure (eg: no outside access) that can also forward directly to a single extensions.

Mike 

-----Original Message-----
From: Nabeel Jafferali [mailto:asterisk-lists at x2n.ca] 
Sent: Wednesday, May 25, 2005 9:00 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Guest

Anton Krall said:
> I think I once read something about creating a peer on sip.conf that 
> should be Guest in order to allow any server to connect without a 
> password to yours and go to the specified context.. Am I right?

Sorry Anton, I have no idea. Let me know if you do figure out how to have a guest peer.

--
Nabeel Jafferali
X2 Networks
www.x2n.ca
T: 1.647.722.6900
   1.877.VOIP.X2N
F: 1.866.655.6698
FWD: 46990

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- 
> bounces at lists.digium.com] On Behalf Of Anton Krall
> Sent: May 18, 2005 1:54 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Guest
> 
> 
> |-----Original Message-----
> |From: asterisk-users-bounces at lists.digium.com
> |[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Nabeel 
> |Jafferali
> |Sent: Miércoles, 18 de Mayo de 2005 08:02 a.m.
> |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |Subject: RE: [Asterisk-Users] Guest
> |
> |> For example, how does your dialplan look on the zap and sip
> |servers in
> |> order to route the call from a zap on server 1 to a sip on server 2?
> |
> |If you want any SIP server/client to be able to call you at 
> |anton at sip.intruder.com, for example, then in the context that is set 
> |in the [general] part of sip.conf (usually default), add something 
> |like:
> |
> |[default]
> |exten => anton,1,Goto(internal,200,1)
> |
> |Similarly, if you want a specific server to be able to do this, add a 
> |peer entry for that server that sets the context, and in that context 
> |put something like the above.
> |
> |Then, on that server, you would Dial(SIP/anton at server2).
> |
> |--
> |Nabeel Jafferali
> |X2 Networks
> |www.x2n.ca
> |T: 1.647.722.6900
> |   1.877.VOIP.X2N
> |F: 1.866.655.6698
> |FWD: 46990
> |
> |
> |> -----Original Message-----
> |> From: asterisk-users-bounces at lists.digium.com
> |[mailto:asterisk-users-
> |> bounces at lists.digium.com] On Behalf Of Anton Krall
> |> Sent: May 18, 2005 1:28 AM
> |> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |> Subject: RE: [Asterisk-Users] Guest
> |>
> |>
> |> |-----Original Message-----
> |> |From: asterisk-users-bounces at lists.digium.com
> |> |[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> |> |Jason Walker
> |> |Sent: Martes, 17 de Mayo de 2005 11:41 p.m.
> |> |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |> |Subject: RE: [Asterisk-Users] Guest
> |> |
> |> |I am a newbie to *, but if the far end of the call has no route to 
> |> |your phone, how do you think this could be accomplished?
> |> |
> |> |I have agents log into one SIP server (no ZAP cards, just
> |SIP). Calls
> |> |come through another * box with ZAP cards that are routed
> |to the SIP
> |> |only server via the extensions.conf file.
> |> |
> |> |It seems to me that the far end would need something in their 
> |> |dialplan to allow for calls to an extension to go to your
> |SIP server.
> |> |
> |> |I apologize if I am giving a "newbie" response - I am also in the 
> |> |process of learning.
> |> |
> |> |-----Original Message-----
> |> |From: asterisk-users-bounces at lists.digium.com
> |> |[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> |> |Anton Krall
> |> |Sent: Tuesday, May 17, 2005 9:08 PM
> |> |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |> |Subject: [Asterisk-Users] Guest
> |> |
> |> |Guys.
> |> |
> |> |What do I need to configure in order to let my Asterisk
> |receive calls
> |> |from sip phones, etc not registered with my server on my extension?
> |> |
> |> |For example, let people use their asterisks or sip phones to call 
> |> |blah111 at server.com?
> |> |
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