[Asterisk-Users] Guest

[Nabeel Jafferali]

> 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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