RE: [Asterisk-Users] Asterisk as a SIP/H.323 Router

[Abdul Hakeem]

Title: Message

Hi, I took a look at the architecture. The way the Cisco boxes will work if Asterisk in the middle is a Proxy, but it's not. You cannot re-direct an incoming Voip call from one gateway to another, only a proxy can do that. The Asterisk in this mode can only terminate the calls via the PSTN. If it attempts to re-direct the call to the Cisco, it has to be via it's PRI interface (i.e. Cisco PRI0-3 is connected to the PRI interface of the Asterisk Location B. Cheers, Abdul -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich Sent: Monday, March 17, 2003 10:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk as a SIP/H.323 Router Hi All,   I've been spending the last month experimenting with Asterisk, and I must say that all results point to a very positive outcome.   Now, i've been asked the following question: Is it possible to put an Asterisk box between 2 Cisco routers or other SIP complianet equiment, then routing SIP/H.323 calls between the two routers?     Here's a drawing that will explain:         +--------------+                   +----------+                 +--------------+  PRI  |              |1.1.1.2     1.1.1.3|          |1.1.2.1   1.1.2.2|              |  ---->+ Cisco Router +-------------------+ Asterisk +--------+--------+ Cisco Router +--> PRI0-3       | Location A   |                   |Location B|        |        |              |       +--------------+                   +----------+        |        +--------------+                                                              |                                                              |        +--------------+                                                              | 1.1.2.3|              |                                                              +--------+ Cisco Router +--> PRI4-7                                                                       |              |                                                                       +--------------+   The question here is this:   A phone call is is made to the router in location A. The Cisco router routes the call via SIP to the Asterisk Box at Location B. According to a set of rules on the Asterisk box, the box would route the SIP/H.323 to one of the the other Cisco boxes, in order to terminate the call.   What do you think, is this possible?   Nir Simionovich   P.S.   Excuse me for the poor ASCII art, I didn't want to attach a Visio or a PDF file.