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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Adtran TA9xx series get a lot of love, and for good reason: they're a solid piece of kit. A TA904 could probably do what you need (Asterisk <- SIP -> Adtran <- E1 -> Panasonic). You might need a different model of Adtran if you need 60 channels (2x E1 interfaces), maybe a TA900e model of some stripe.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I haven't used one, but I just recently found out about Yeastar and their line of VoIP gateways. They seem to be competitively priced, and although their marketing materials do not mention it, they run Asterisk internally, and it looks like they also have a serial console and SSH support (so you can edit the Asterisk and DAHDI config files by hand, use an Asterisk remote console, etc.). You would probably need at least a TE200: <a href="http://www.yeastar.com/pri-voip-gateway-te200/">http://www.yeastar.com/pri-voip-gateway-te200/</a><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Finally, according to another respondent, your Panasonic phone system supports H.323. I believe Asterisk has an H.323 channel driver. So even though Panasonic does not support SIP itself, it does support VoIP trunking, just with a different protocol. Have you thought about trying to get Asterisk and the Panasonic talking to each other over IP using H.323? I have no experience with H.323 so I wouldn't be much help, but just thought I'd throw that out there.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>-- Nathan<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Ikka Tirtawidjaja<br><b>Sent:</b> Tuesday, September 13, 2016 9:16 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Panasonic PBX connect to Asterisk (Ikka Tirtawidjaja)<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>Dear Mc Grath R,<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thx for the information.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>If that TDA-600 can not use SIP Trunking, can we use "SIP to E1 converter/gateway" between my Asterisk Server and Panasonic KX-TDA600 ?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>or from asterisk we convert the sip phone line with ATA Converter / FXS Gateway, and then we connect the output (RJ11) to C0 on Panasonic TDA600 ?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Any recomendation for SIP to E1 gateway ? <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Thx in advance <o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Ikka<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Wed, Sep 14, 2016 at 12:13 AM, Mc GRATH Ricardo <<a href="mailto:mcgrathr@mail2web.com" target="_blank">mcgrathr@mail2web.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it only support H323 Trunk through 4 or 16 channels gateway card and TDM technology with ISDN BRI and PRI card.<br><br>Mc GRATH Ricardo<br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016<br> <a href="http://www.asterisk.org/community/astricon-user-conference" target="_blank">http://www.asterisk.org/community/astricon-user-conference</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></body></html>