<div style="font-family:Arial;font-size:14px"><div style="font-family: Arial; font-size: 14px;">I am doing what u wanna atm but instead of an Alcatlet with SIP support i have to <br />struggle with an Avaya CM without SIP but with H.323.<br />So far putting a trunk over Ethernet with SIP is the way I gonna go.<br />I havent run in to any show-stopper so far with my CM H.323 - Asterisk integration.<br /></div>
<p>On Mon, 23 Nov 2009 11:17:22 -0500, Ryan Wagoner <rswagoner@gmail.com> wrote:</p>
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<p>Either use SIP or PRIs to do the integration. FXO and FXS interfaces<br />are a single port, where as a PRI will provide you with 23 channels.<br />Use QSIG signaling over the PRI so Caller ID names will show between<br />the systems.<br /><br />I just integrated a Toshiba CIX with Asterisk due to the cost for SIP<br />licensing and the reliability of the Toshiba VOIP Phones. They were<br />having hardware failures every few months. I went with Sangoma PRI<br />cards using QSIG.<br /><br />Everything has been working great and I have rolled out 12 Snom 370<br />phones to work with the 150 Toshiba Digital phones. To the end users<br />the experience is seamless as they can 4 digit dial any extension and<br />the call will be routed to the correct system. This does take a bit of<br />duplicate setup on the two systems, but was worth the hassle for the<br />end result.<br /><br />Ryan<br /><br />On Mon, Nov 23, 2009 at 6:17 AM, Alex Balashov<br /> wrote:<br />> PRI is likely the simplest and most reliable.<br />><br />> Xavier Mesquida wrote:<br />><br />>><br />>> Hi, I want to interconnect a Alcatel OmniPCX PBX with SIP support with<br />>> an Asterisk PBX. My intention is Alcatel PBX manage all external calls<br />>> and analog extensions and Asterisk manage all the SIP users (because I<br />>> have to pay for every SIP license in Alcatel PBX and I can’t edit<br />>> configuration or password in that PBX)<br />>><br />>> What’s the best way to interconnect the 2 PBX? With SIP, with a FXO<br />>> interface or FXS? How can I do that? Thanks<br />>><br />>><br />>><br />>><br />>> ------------------------------------------------------------------------<br />>><br />>> _______________________________________________<br />>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br />>><br />>> asterisk-users mailing list<br />>> To UNSUBSCRIBE or update options visit:<br />>> http://lists.digium.com/mailman/listinfo/asterisk-users<br />><br />><br />> --<br />> Alex Balashov - Principal<br />> Evariste Systems<br />> Web : http://www.evaristesys.com/<br />> Tel : (+1) (678) 954-0670<br />> Direct : (+1) (678) 954-0671<br />><br />> _______________________________________________<br />> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br />><br />> asterisk-users mailing list<br />> To UNSUBSCRIBE or update options visit:<br />> http://lists.digium.com/mailman/listinfo/asterisk-users<br /><br />_______________________________________________<br />-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br /><br />asterisk-users mailing list<br />To UNSUBSCRIBE or update options visit:<br /> http://lists.digium.com/mailman/listinfo/asterisk-users</p>
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