hi Gopal, A PRI and E1 are basically the same thing. Some will also call it PRA. Firstly you will need the same hardware interface on both sides, if you use a E1 you need to connect to a E1 interface on both sides. Secondly you will need the same SS7 stack on both sides - with Asterisk you need MTP and ISUP with a matching configuration. The two main configs are ETSI and ANSI and with a E1 you usually use ETSI version. But, you need to get an exact description of the protocol versions used on the switch and match that on Asterisk. And to bring the link up on MTP2 and MTP3 you will need to know the point codes,link types and link circuits and CIC number scheme. If you connect with a single E1 you will a F-Link. A-Link is redundancy, but you need 2 E1's (or T1's). You need to know a minimum of 2 point-codes. A point code is SS7's version of an IP address. An E1 have 32 x 64kbps data streams. Channel 0 is used for sync, which leaves 31 "circuits" that you can use for signalling. The most common configuration is to use channel 16 for SS7 signalling and leave the others for voice. To get voice through you will need matching CIC numbering schemes on both ends. For a single E1 it usually is 1 to 30. Start by getting an exact description of what protocol and configuration you will be talking to. The "switch" is usually called a "MSC". Hope this helps and good luck. Jan From: saigop@xxxxxxxxx Date: Fri, 17 Jun 2011 19:40:09 +0530 To: asterisk-ss7 at lists.digium.com Subject: SS7 End to End Network Connectivity... Hi users, I want to know about end to end connectivity of SS7 network from the switch side to customer premises equipment. For example in PRI in the exchange side we have ASMI RAD modem and in customer premises we have one ASMI RAD modem. In the same way I would like to know what kind of equipment we have in customer premises and the line connectivity to asterisk server E1 card. Can some one direct me on this. I tried googling, but I got like the complete protocol stack or the SS7 layer architecture, not getting the connectivity in customer premises side. Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-ss7 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20110617/db8fad59/attachment.htm>
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