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Are you sure? yes no
TDM410 PTSN line setup with 1 analog phone
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Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds
1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1
(not trying to use the gui, want to do everything by hand) with a TDM410 with
2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a
analog phone connected to port 3 (FXS). I compiled asterisk with asterisk
samples so I realize that may have messed me up.
This is all running on Ubuntu Server 12.04. I have been googling/researching
reading the book, etc. Everything I find is for SIP softphones etc. I just
want to start by getting the asterisk machine to provide dialtone to the analog
phone, and ring that phone when I call the PTSN line.
I must be missing something in the basic dahdi and dialplan to simple get the
analog phone to work. Can someone point me to a example of what I am trying to
accomplish? Not wanting handholding but a push in the right direction.
Thanks.
An FXO port needs to be connected to dial tone or your PSTN line. And an FXS port needs to be connected to the station equipment(ie. a physical phone). The TDM410 is basically a channel bank to Asterisk, so the channel type inside Asterisk is FXO to talk to the physical FXS card and FXS to talk to the physical FXO port. Lyle Giese LCR Computer Services, Inc.On 06/18/12 15:08, Joseph Towery wrote: > Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 > asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 > and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do > everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS > (PTNS) line plugged into port 1 (FXO) and a analog phone connected to > port 3 (FXS). I compiled asterisk with asterisk samples so I realize > that may have messed me up. > > This is all running on Ubuntu Server 12.04. I have been > googling/researching reading the book, etc. Everything I find is for > SIP softphones etc. I just want to start by getting the asterisk > machine to provide dialtone to the analog phone, and ring that phone > when I call the PTSN line. > > I must be missing something in the basic dahdi and dialplan to simple > get the analog phone to work. Can someone point me to a example of > what I am trying to accomplish? Not wanting handholding but a push in > the right direction. > > Thanks. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/aste...
Thanks Lyle,
Sorry to sound so much like a newb but in asterisk I am. I was initially trying
to do things by hand in the extensions.conf file and had no luck. I then got
from SVN checkout asterisk-gui and used it to simply try and get things started,
and created a trunk, users, incoming rule, etc. from the gui and finally got
dial tone, and can dial out, but I haven't got the analog phone ringing yet. I
will have more targeted questions in the near future. It is just hard to find
"google" help for analog answers. Most deal with SIP (which is my next step
once I have the analog lines working).
Thanks,
On 06/20/2012 08:44 AM, Joseph Towery wrote: > Sorry to sound so much like a newb but in asterisk I am. I was initially > trying to do things by hand in the extensions.conf file and had no luck. > I then got from SVN checkout asterisk-gui and used it to simply try and > get things started, and created a trunk, users, incoming rule, etc. from > the gui and finally got dial tone, and can dial out, but I haven't got > the analog phone ringing yet. I will have more targeted questions in the > near future. It is just hard to find "google" help for analog answers. > Most deal with SIP (which is my next step once I have the analog lines > working).Have you read any of the O'Reilly Asterisk books? They will help you learn quite a lot about Asterisk, and they are available online. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: | SIP: | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
Kevin, Thanks for the tip, the answer is yes, (I forgot I copy the first message in into the body below,) but I have read a lot in the http://cdn.oreilly.com/books/9780596510480.pd... and http://ofps.oreilly.com/titles/9780596517342/... pages. I was just wanting to get the very basic analog config working prior to jumping into SIP and other higher level things, and that is where I was having a stumbling block. I am making tiny steps forward at least right now. Thanks ________________________________ From: Kevin P. Fleming To: Sent: Wed, June 20, 2012 10:06:48 AM Subject: Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phoneOn 06/20/2012 08:44 AM, Joseph Towery wrote: > Sorry to sound so much like a newb but in asterisk I am. I was initially > trying to do things by hand in the extensions.conf file and had no luck. > I then got from SVN checkout asterisk-gui and used it to simply try and > get things started, and created a trunk, users, incoming rule, etc. from > the gui and finally got dial tone, and can dial out, but I haven't got > the analog phone ringing yet. I will have more targeted questions in the > near future. It is just hard to find "google" help for analog answers. > Most deal with SIP (which is my next step once I have the analog lines > working).Have you read any of the O'Reilly Asterisk books? They will help you learn quite a lot about Asterisk, and they are available online. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: | SIP: | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
On 06/20/2012 09:34 AM, Joseph Towery wrote: > Thanks for the tip, the answer is yes, (I forgot I copy the first > message in into the body below,) but I have read a lot in the > http://cdn.oreilly.com/books/9780596510480.pd... and > http://ofps.oreilly.com/titles/9780596517342/... > pages. I was just wanting to get the very basic analog config working > prior to jumping into SIP and other higher level things, and that is > where I was having a stumbling block. I am making tiny steps forward at > least right now. >Starting at page 79 of the 2nd Edition of the book, you'll find step-by-step instructions on setting up an FXS port for use with an analog telephone. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: | SIP: | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
Thanks. I will go back and use that reference. I was using examples on web pages I was trying to use and just got confused with too much information. ________________________________ From: Kevin P. Fleming To: Sent: Wed, June 20, 2012 10:48:14 AM Subject: Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phoneOn 06/20/2012 09:34 AM, Joseph Towery wrote: > Thanks for the tip, the answer is yes, (I forgot I copy the first > message in into the body below,) but I have read a lot in the > http://cdn.oreilly.com/books/9780596510480.pd... and > http://ofps.oreilly.com/titles/9780596517342/... > pages. I was just wanting to get the very basic analog config working > prior to jumping into SIP and other higher level things, and that is > where I was having a stumbling block. I am making tiny steps forward at > least right now. >Starting at page 79 of the 2nd Edition of the book, you'll find step-by-step instructions on setting up an FXS port for use with an analog telephone. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: | SIP: | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
I have not use a TDM4xx card for a while, but I remember that in order for ringing to work, you had to plug in an extra molex connector into the card to supply power to the ringing generator portion. If you forgot to do that... Lyle BTW, I know about being a noobie. I was there once myself and still am there every day learning and working with new stuff. Sometimes not of my own choosing, but one must do what they need to keep getting those paychecks<GRIN>!On 6/20/2012 8:44 AM, Joseph Towery wrote: > Thanks Lyle, > > Sorry to sound so much like a newb but in asterisk I am. I was > initially trying to do things by hand in the extensions.conf file and > had no luck. I then got from SVN checkout asterisk-gui and used it to > simply try and get things started, and created a trunk, users, incoming > rule, etc. from the gui and finally got dial tone, and can dial out, but > I haven't got the analog phone ringing yet. I will have more targeted > questions in the near future. It is just hard to find "google" help for > analog answers. Most deal with SIP (which is my next step once I have > the analog lines working). > > Thanks, > > ------------------------------------------------------------------------ > *From:* Lyle Giese > *To:* > *Sent:* Tue, June 19, 2012 9:29:12 PM > *Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone > > An FXO port needs to be connected to dial tone or your PSTN line. And an > FXS port needs to be connected to the station equipment(ie. a physical > phone). > > The TDM410 is basically a channel bank to Asterisk, so the channel type > inside Asterisk is FXO to talk to the physical FXS card and FXS to talk > to the physical FXO port. > > Lyle Giese > LCR Computer Services, Inc. > > On 06/18/12 15:08, Joseph Towery wrote: >> Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 >> asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 >> and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do >> everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS >> (PTNS) line plugged into port 1 (FXO) and a analog phone connected to >> port 3 (FXS). I compiled asterisk with asterisk samples so I realize >> that may have messed me up. >> >> This is all running on Ubuntu Server 12.04. I have been >> googling/researching reading the book, etc. Everything I find is for >> SIP softphones etc. I just want to start by getting the asterisk >> machine to provide dialtone to the analog phone, and ring that phone >> when I call the PTSN line. >> >> I must be missing something in the basic dahdi and dialplan to simple >> get the analog phone to work. Can someone point me to a example of >> what I am trying to accomplish? Not wanting handholding but a push in >> the right direction. >> >> Thanks. >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/aste... > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/aste... >
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. Thanks for your help. ________________________________ From: Lyle Giese To: Sent: Wed, June 20, 2012 10:12:29 AM Subject: Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone I have not use a TDM4xx card for a while, but I remember that in order for ringing to work, you had to plug in an extra molex connector into the card to supply power to the ringing generator portion. If you forgot to do that... Lyle BTW, I know about being a noobie. I was there once myself and still am there every day learning and working with new stuff. Sometimes not of my own choosing, but one must do what they need to keep getting those paychecks<GRIN>!On 6/20/2012 8:44 AM, Joseph Towery wrote: > Thanks Lyle, > > Sorry to sound so much like a newb but in asterisk I am. I was > initially trying to do things by hand in the extensions.conf file and > had no luck. I then got from SVN checkout asterisk-gui and used it to > simply try and get things started, and created a trunk, users, incoming > rule, etc. from the gui and finally got dial tone, and can dial out, but > I haven't got the analog phone ringing yet. I will have more targeted > questions in the near future. It is just hard to find "google" help for > analog answers. Most deal with SIP (which is my next step once I have > the analog lines working). > > Thanks, > > ------------------------------------------------------------------------ > *From:* Lyle Giese > *To:* > *Sent:* Tue, June 19, 2012 9:29:12 PM > *Subject:* Re: [asterisk-users] TDM410 PTSN line setup with 1 analog phone > > An FXO port needs to be connected to dial tone or your PSTN line. And an > FXS port needs to be connected to the station equipment(ie. a physical > phone). > > The TDM410 is basically a channel bank to Asterisk, so the channel type > inside Asterisk is FXO to talk to the physical FXS card and FXS to talk > to the physical FXO port. > > Lyle Giese > LCR Computer Services, Inc. > > On 06/18/12 15:08, Joseph Towery wrote: >> Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 >> asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 >> and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do >> everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS >> (PTNS) line plugged into port 1 (FXO) and a analog phone connected to >> port 3 (FXS). I compiled asterisk with asterisk samples so I realize >> that may have messed me up. >> >> This is all running on Ubuntu Server 12.04. I have been >> googling/researching reading the book, etc. Everything I find is for >> SIP softphones etc. I just want to start by getting the asterisk >> machine to provide dialtone to the analog phone, and ring that phone >> when I call the PTSN line. >> >> I must be missing something in the basic dahdi and dialplan to simple >> get the analog phone to work. Can someone point me to a example of >> what I am trying to accomplish? Not wanting handholding but a push in >> the right direction. >> >> Thanks. >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/aste... > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/aste... >