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