> 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 [EMAIL PROTECTED], 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/[EMAIL PROTECTED]). -- 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: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] 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: [EMAIL PROTECTED] > |[mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] > |[mailto:[EMAIL PROTECTED] 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 [EMAIL PROTECTED] > | > |_______________________________________________ > |Asterisk-Users mailing list > |[email protected] > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > | > | > | > |-- > |No virus found in this outgoing message. > |Checked by AVG Anti-Virus. > |Version: 7.0.308 / Virus Database: 266.11.11 - Release Date: 5/16/2005 > | > |_______________________________________________ > |Asterisk-Users mailing list > |[email protected] > |http://lists.digium.com/mailman/listinfo/asterisk-users > |To UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > | > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
