Connecting Asterisk to PSTN wall socket (Newbie)


#1

Hi all,

I’m trying to receive a call from the PSTN network on Asterisk. I have a PSTN wall socket (POTS line, not PRI/BRI). Is there any adapter that I can use for this purpose (adapter will receive SS7 call, convert to SIP, and send it to my Asterisk server)?

I have a Linksys PAP2T Internet Phone Adapter but, after reading the documentation, I realized that this adapter was not meant to be connected directly to a PSTN wall socket.

To provide a bit of background:

I’m doing a project by myself at uni to do some filtering to stop spam/robot calls. I implement this on Asterisk dial plan and, at the moment, in a simulation environment (i.e. using VMs / softphones). I’d like to make my project more realistic by being able to receive a real-world PSTN call. I did read about Digium cards that process calls coming from PRI/BRI lines. However, for my purpose, I was hoping I wouldn’t have to shell out around $100 to get those cards. Also, since those are PCI, I’m not sure how I can connect it to my laptop. Does anyone know if an adapter is sufficient and, if so, can you suggest an adapter?

Apologies if my question is a bit basic as I’m new to Asterisk (been a few weeks).

Also, on a side note, do I need DAHDI drivers to receive incoming PSTN calls? I had trouble installing them. It seems like the latest DAHDI driver (2.11.1) is incompatible/has issues with the latest Asterisk (14.3.0) and someone suggested, on a forum, to wait for the next release.

Thanks,
Clive


#2

SS7 is only used on PRI/BRI!

If SS7 was an error, you can use an FXO module on a Digium PCI[e] ananlogue interface card, or you can look for a black box called a SIP gateway.


#3

Thanks david551 for your reply.

  1. I thought SS7 was a family of protocols used in traditional telephony? If SS7 is only used for PRI/BRI, what does the analog line use?

  2. Is it necessary to get a Digium PCI (as I don’t think it will connect to my laptop)? Also, does the SIP gateway do the same function? I found an ATA that has an FXO port - https://www.myithub.com.au/Grandstream-HT503-1-Port-FXS-1-Port-FXO-ATA-built-in-router-PSTN-lifeline-P69738.aspx … Will this work?


#4

SS7 will probably be used between your public telephone operator’s exchanges. Incidentally, I think it is only used at PRI and above, not at BRI. However, the interface to your wall socket is analogue and based on a two wire system using 48V DC, on hook, and typically about 60V AC superimposed for ringing.

Although most PTOs will use similar standards for signalling an outgoing seize of the line, dialled digits, incoming ringing and answer, and station hangup of the line, there will be differences in detail and support for signalling answer on outgoing calls are caller release may not be present at all. A station hangup is likely to be treated as hold, rather than a release, especially if it is the called party. It cannot signal the richness of call failure reasons, or the distinction between hanugp (Clear) and Release.

If you are lucky, your PTO will provide a technical document specifying the interface, e.g. http://www.sinet.bt.com/sinet/SINs/pdf/351v4p6.pdf for BT in the UK.

Note that SS7 is not the only signalling system used on digital lines. Lines to PABXes and lines between PABXes are more likely to use protocols like Q.SIG or DPNSS.

On your second question, I would search the forum for the device you mention. It does sound like one that people have used.


#5

Thanks david551 for your answer. I’ll try out the Grandstream HT503. The world of traditional telephony seems complex to me. Do you have any recommendations on where I could get a clean and clear picture of how traditional telephony works and what protocols are used? From what I’ve read so far, there seem to be a variety of protocols being used and it has only left me confused.


#6

There is some information in http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id271202.html However, it doesn’t really get into details of the higher level signalling options for digital circuits, or the, typical, internal structure of the PSTN. You would probably need a whole book for that; maybe several.


#7

Thanks david551. Can you also tell me whether I need the DAHDI drivers installed (and working) on Asterisk in order to connect Asterisk to the PSTN network (using the Grandstream HT503 adapter - which has one FXO, one FXS, one WAN, and one LAN port)?


#8

Dahdi is used to:

  • handle PCI cards;
  • as one possible source of high resolution timing (although since about Asterisk 1.8 have been other options when you don’t have a PCI card;
  • implement the conference bridge used by the meetme application, either using a PCI card or in software. More recent conferencing applications do the bridge i user space.

Do you have any of these requirements?

I don’t use an HT203, so my original advice stands, which is to trawl the forum for postings from people who do.


#9

Thanks david551 for your explanations and advice. I would be using the adapter instead of the PCI (although I read on forums that the using a PCI gives better call quality) as this would be sufficient for the purposes of my project.