<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt"><div>Yes, I have connected that, and the pci card has the lights on. I can now lift the receiver on the analog phone get dial tone and dial out. Next I need to get the phone to ring when called. Off to do more research.<br><br>Thanks for your help.<br></div><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><br><div style="font-family:arial, helvetica, sans-serif;font-size:14pt"><font face="Tahoma" size="2"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> Lyle Giese <lyle@lcrcomputer.net><br><b><span style="font-weight: bold;">To:</span></b> asterisk-users@lists.digium.com<br><b><span style="font-weight: bold;">Sent:</span></b> Wed, June 20, 2012 10:12:29 AM<br><b><span style="font-weight: bold;">Subject:</span></b> Re:
[asterisk-users] TDM410 PTSN line setup with 1 analog phone<br></font><br>
I have not use a TDM4xx card for a while, but I remember that in order <br>for ringing to work, you had to plug in an extra molex connector into <br>the card to supply power to the ringing generator portion.<br><br>If you forgot to do that...<br><br>Lyle<br><br>BTW, I know about being a noobie. I was there once myself and still am <br>there every day learning and working with new stuff. Sometimes not of <br>my own choosing, but one must do what they need to keep getting those <br>paychecks<GRIN>!<br><br>On 6/20/2012 8:44 AM, Joseph Towery wrote:<br>> Thanks Lyle,<br>><br>> Sorry to sound so much like a newb but in asterisk I am. I was<br>> initially trying to do things by hand in the extensions.conf file and<br>> had no luck. I then got from SVN checkout asterisk-gui and used it to<br>> simply try and get things started, and created a trunk, users, incoming<br>> rule, etc. from the gui and finally got
dial tone, and can dial out, but<br>> I haven't got the analog phone ringing yet. I will have more targeted<br>> questions in the near future. It is just hard to find "google" help for<br>> analog answers. Most deal with SIP (which is my next step once I have<br>> the analog lines working).<br>><br>> Thanks,<br>><br>> ------------------------------------------------------------------------<br>> *From:* Lyle Giese <<a ymailto="mailto:lyle@lcrcomputer.net" href="mailto:lyle@lcrcomputer.net">lyle@lcrcomputer.net</a>><br>> *To:* <a ymailto="mailto:asterisk-users@lists.digium.com" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>> *Sent:* Tue, June 19, 2012 9:29:12 PM<br>> *Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone<br>><br>> An FXO port needs to be connected to dial tone or your PSTN line. And an<br>> FXS port needs to be
connected to the station equipment(ie. a physical<br>> phone).<br>><br>> The TDM410 is basically a channel bank to Asterisk, so the channel type<br>> inside Asterisk is FXO to talk to the physical FXS card and FXS to talk<br>> to the physical FXO port.<br>><br>> Lyle Giese<br>> LCR Computer Services, Inc.<br>><br>> On 06/18/12 15:08, Joseph Towery wrote:<br>>> Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24<br>>> asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12<br>>> and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do<br>>> everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS<br>>> (PTNS) line plugged into port 1 (FXO) and a analog phone connected to<br>>> port 3 (FXS). I compiled asterisk with asterisk samples so I realize<br>>> that may have messed me up.<br>>><br>>> This is all running on
Ubuntu Server 12.04. I have been<br>>> googling/researching reading the book, etc. Everything I find is for<br>>> SIP softphones etc. I just want to start by getting the asterisk<br>>> machine to provide dialtone to the analog phone, and ring that phone<br>>> when I call the PTSN line.<br>>><br>>> I must be missing something in the basic dahdi and dialplan to simple<br>>> get the analog phone to work. Can someone point me to a example of<br>>> what I am trying to accomplish? Not wanting handholding but a push in<br>>> the right direction.<br>>><br>>> Thanks.<br>>><br>>><br>>> --<br>>> _____________________________________________________________________<br><span>>> -- Bandwidth and Colocation Provided by<a target="_blank" href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br>>> New to Asterisk?
Join us for a live introductory webinar every Thurs:<br><span>>> <a target="_blank" href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a></span><br>>><br>>> asterisk-users mailing list<br>>> To UNSUBSCRIBE or update options visit:<br><span>>> <a target="_blank" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span><br>><br>><br>><br>> --<br>> _____________________________________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>> <a href="http://www.asterisk.org/hello"
target="_blank">http://www.asterisk.org/hello</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><br>><br><br><br><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</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"
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