Hi All, I have a question about how a particular situation would work between two PBX systems: If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same rack, same network), and then pass a call from the Mitel to Asterisk to perform some functions (lookups, maybe routing), and then pass the call back to the Mitel to be routed to it's endpoint, would Asterisk stay in that loop after the call was passed back to the Mitel? Or, does the call leave Asterisk completely when passed back? If it does leave/stay in the loop, is there a way to force it to leave/stay based on what my needs are? Thanks, Kenny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100401/c9c0e17e/attachment.htm
Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302. If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. Rodr?guez -----Original Message----- From: Kenneth Noisewater <noisewaterphd at gmail.com> Date: Thu, 1 Apr 2010 16:50:47 To: <asterisk-users at lists.digium.com> Subject: [asterisk-users] SIP Connection Question -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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