[Asterisk-Users] Guest
Anton Krall
akrall-lists at intruder.com.mx
Wed May 18 10:54:03 MST 2005
Nabeel
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?
|-----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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|> |
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