Phone Line Coming In

Sorry if this sounds like a basic question, but I can’t seem to find a strait answer online, and you guys have been great helping me out so far as I am very new to Asterisk.

So I have set up Asterisk with extensions etc. and have it working nicely with the free sip softphone X-Lite 4. I am now looking to be able to get into my system with outside numbers (not sure of actual name eg. dialed from physical phone). What exsactly am I looking for, a DID, SIP Trunk, VoiP Trunk?

I also need to be able to have many different phone numbers (e.g… 1-888-111-2222, and 1-888-1111-3333) all coming into my Asterisk, this is possible correct?


SIP Trunk is a subset of VoIP Trunk. You can also use an ISDN trunk, or, given you haven’t specified more than one simultaneous call, a single analogue line; these both require special hardware.

You actually need Direct In Dialing, however, in the VoIP world, DID seems to be misused to refer to someone who has ISDN direct in dialing and then forwards numbers over a VoIP connection, regardless of whether they actually forward any of the dialed digits. As such, in the VoIP world, DID tends to be a synonym for VoIP Trunk, so, if you use a VoIP provider of incoming PSTN connections, make sure that they are actually providing you with dialed digits.

I’m not sure if Asterisk supports DID over a simple analogue line, but it will certainly support it over ISDN, as well as over VoIP (SIP, H.323, IAX, etc.).

It is more likely that you will get direct in dialing if you have a static IP address, as the registration process needed for fully dynamic access tends to force the contents of the dialed digits, although I’m sure some providers must have got round that somehow.

Thanks for the reply. So it appears I need to add a little more info. I won’t have a physical telephone line going into the Asterisk server. I need it to all be done over the internet. And I would need to be able to do simultanius calls, both on multiple calls on the same number, and multiple calls from different numbers all at the same time. It probably wouldn’t happen often, but I need to be able to accommodate it.

So would I be looking for a SIP Trunk then?

SIP or IAX or H.323, the first two being the most likely ones to be available. However you need to be more specific, as you want part of the original dialed digits to be forwarded, so you will need a service designed for PABX users, not for direct use with SIP phones.

Assuming that the 1111 was a typo, I assume you are in the NANP area. I don’t know who are the good business SIP providers there. (To be honest, I don’t really know much about the UK market, either.)

I’m surprised you couldn’t find this information. Most people start with all this as given and are trying to do it on the cheap or find a reliable provider.

Yes “1111” was a typo. So is SIP and SIP Trunk the same thing? And yes I need it to do caller id. I will have no outbound, I just need inbound calls.

Please read the book at

I’m not sure that it says much about service providers, but I think you need quite a bit of the background material.

SIP is a protocol. A SIP trunk is where two VoIP devices communicate using SIP in a way that allows an arbitrary number of simultaneous calls between them (although commercial considerations may impose limits, and available bandwidth will impose limits). Asterisk itself doesn’t really make much distinction between SIP being used as a trunk between PABXes and being used to connect to a phone, although one would normally use the term trunk when neither end was a phone.

I think most providers provide both incoming and outgoing services.

I’m not sure how the US market works, but in the UK, the way that landline calls are charged means that is possible to provide an incoming service free of charge to the recipient.