Hello community, I have successfully set up asterisk free PBX server and I am also able to connect to it by softphone. Now as next step I want to extend this to PSTN , My Required scenario: I need a number which will connect outside PSTN world to my PBX and by applying extension particular softphone or connected normal phone should get connected. Which hardware I need for it. Also please explain a bit of dial plans. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101014/b88bdab3/attachment.htm
Gopalakrishnan A.N
2010-Oct-14 14:09 UTC
[asterisk-users] How to connect asterisk PBX to PSTN
Hi Joshi, To connect with PSTN line you need FXO / FXS card. FXO is used to connect CO line and FXS is used to connect internal station line. With help of FXO you can connect the outside world and with help of FXS you can connect normal analog phones. Inspite of normal analog phones you can connect SIP phones (soft phones) also. Some vendors are there for these PSTN cards like Digium, Sangoma, Openvox. Good luck....:) On Thu, Oct 14, 2010 at 7:05 PM, Jigar Joshi <jigarjm at gmail.com> wrote:> Hello community, > > I have successfully set up asterisk free PBX server and I am also able to > connect to it by softphone. > > Now as next step I want to extend this to PSTN , > > My Required scenario: > > I need a number which will connect outside PSTN world to my PBX and by > applying extension particular softphone or connected normal phone should get > connected. > > Which hardware I need for it. > Also please explain a bit of dial plans. > > 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/asterisk-users >-- Thank you with regards, Gopalakrishnan A.N, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101014/2c5005d0/attachment.htm
hi
check chan_dahdi
you need digium hardware or external gateway
to use external gateway you can use only sip
for digium hardware use dahdi
otherwise use a sip provider :P
----- Original Message -----
From: Jigar Joshi
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, October 14, 2010 3:35 PM
Subject: [asterisk-users] How to connect asterisk PBX to PSTN
Hello community,
I have successfully set up asterisk free PBX server and I am also able to
connect to it by softphone.
Now as next step I want to extend this to PSTN ,
My Required scenario:
I need a number which will connect outside PSTN world to my PBX and by
applying extension particular softphone or connected normal phone should get
connected.
Which hardware I need for it.
Also please explain a bit of dial plans.
Thanks
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I will have a closer look at this book My Question is : Is it possible with asterisk to resolve all the code to one extension and with the extension no. For example person one calls it and enters code 1 person two calls and enters code 2 both the call should received by a single extension say 1001. and there I should be able to differenciate both the calls using code entered. in the example: both the call will be given to extension 1001 and at 1001 there will be an app running that will make this into two calls.i mean each packet contains the code entered. I hope this answer would be helpful . Thanks. On Mon, Oct 18, 2010 at 9:39 PM, Steve Edwards <asterisk.org at sedwards.com>wrote:> On Mon, 18 Oct 2010, Jigar Joshi wrote: > > > @Gilles here are my requirement.can you please help me . > > Are you putting this "out to bid" or are you just too lazy to read ATFOT > (http://downloads.oreilly.com/books/9780596510480.pdf)? > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _____________________________________________________________________ > -- 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/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101019/b0b6da21/attachment.htm
Un-top-posting...> On Mon, 18 Oct 2010, Jigar Joshi wrote: > > > @Gilles here are my requirement.can you please help me .> On Mon, Oct 18, 2010 at 9:39 PM, Steve Edwards > <asterisk.org at sedwards.com> wrote: > > Are you putting this "out to bid" or are you just too lazy to read ATFOT > (http://downloads.oreilly.com/books/9780596510480.pdf)?On Tue, 19 Oct 2010, Jigar Joshi wrote:> I will have a closer look at this book > > My Question is : > > Is it possible with asterisk to resolve all the code to one extension > and with the extension no. > > For example person one calls it and enters code 1 person two calls and > enters code 2 ? > > both the call should?received?by a single extension say 1001. and there > I should be able to differenciate both the calls using code entered. > > in the example: both the call will be given to extension 1001 and at > 1001 there will be an app running that will make this into two calls.i > mean each packet contains the code entered.Reading the book will help you understand the terminology so you will either learn how to write a dialplan to suit your needs or ask questions that can be answered. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
_____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jigar Joshi Sent: Tuesday, October 19, 2010 1:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to connect asterisk PBX to PSTN I will have a closer look at this book My Question is : Is it possible with asterisk to resolve all the code to one extension and with the extension no. For example person one calls it and enters code 1 person two calls and enters code 2 both the call should received by a single extension say 1001. and there I should be able to differenciate both the calls using code entered. in the example: both the call will be given to extension 1001 and at 1001 there will be an app running that will make this into two calls.i mean each packet contains the code entered. I hope this answer would be helpful . Thanks. <snip> #1. Steve (as usual) is correct #2. We eat this kind of "simple dialplanning" for lunch #3. After you read the book, ask it again if needed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101019/876937f3/attachment.htm