Hi all, I’m a newbie to Asterisk. I have a need and I don’t know if Asterisk can meet that need. I rented an E1 trunk line from a telecom provider for calls, E1 trunk connected to my Avaya G650 gateway, and my analog phone made incoming and outgoing calls through the G650. I want to change my analog phone to a SIP phone, can I replace the G650 with asterisk? If so, what hardware and software are needed in addition to a computer with Asterisk installed? How do I configure Asterisk?
Howdy! Welcome to the forums.
Probably, yes. A system running Asterisk with the right hardware can be configured to do something like this:
Telco ↔ E1 ↔ Asterisk PBX ↔ SIP ↔ Phone
For E1, the Sangoma A100 series of interface cards is a very Asterisk-compatible choice.
There are some drivers for the card (wanpipe and dahdi), and you can try Asterisk 22 !
You can find lots of getting started with Asterisk information on the documentation site.
A solution where no drivers are required would be an external gateway. Manufacturers such as Patton or Beronet offer such gateways.
Sure, you could do that, but OP’s question was about replacing an Avaya box with an Asterisk box.
Another answer to the question might be one of a graduated pass thru approach (although it requires a 2 port E1 card instead of 1 port E1 card).
So in addition to the scenario described in the earlier post:
Telco ↔ E1 ↔ Asterisk PBX ↔ SIP ↔ Phone
You could consider a pass thru scenario:
Telco ↔ E1 ↔ port 1 on card ↔ Asterisk PBX ↔ port 2 on card ↔ E1 ↔ Avaya G650 ↔ analog ↔ phone
…which allows doing “nothing” initially but routing the calls thru Asterisk, then, as you get comfortable, start adding more services in Asterisk, like a SIP phone, or an IVR menu.
Hi,
I thought, as a newbie to asterisk he maybe wants starts with a Asterisk “Box” in a VM. Than the external gateways would be a better solution as no drivers are needed (all SIP traffic to Asterisk).
Yes, that would also be my recommendation. This way the RTP traffic can remain on the card/gateway and the call can be signaled via Asterisk.
Oh,great! That’s just I wanted. Thanks very much. I’m also studying the documentation on the website, but it feels like the documentation is a bit difficult for me, I don’t know if the configuration of this requirement scenario is simple or complex for asterisk? Is there a similar configuration case on the website? I want to learn as I build it so that it can be easier to understand.
Forwarding the call is trivial. Handling the SIP side is something nearly every user does. The E1 side will not need a lot of configuration, but circuit switched telephony is becoming obsolescent and traditionally Digium’s position was that you should get configuration help from the manufacturer of the the E1 card, so there will not be much old information on the forum, and I think only on current active responder on the forum has any expertise in DAHDI at all.
While I am not that active on here I have used Sangoma, Digium and Xorcom cards. The last time I purchased any hardware was in 2018 and we went with Xorcom for some SS7 links (that are sadly still in use). This is running Asterisk 15. I used Sangona “back in the day” when they had better name than Digium (circa 2004). I have found that Digium cards were the easiest to setup as they “just worked with Dahdi”. When I used Sangoma it required additional steps such as patching Dahdi. I would see if there was anyway to get rid of the physical line and go with SIP. If you must keep the E1 I would go with Xorcom as there was native support for their devices in Dahdi. From what I recall their support was also pretty helpful.
Thank you. Now this is the case:
Telco ↔ E1 ↔ Avaya G650 ↔phone
Carriers can also provide SIP trunk services, so can I change it to the following, add SIP phones, and keep the original phone unchanged.
Telco ↔ SIP ↔ Asterisk PBX ↔ E1 ↔ Avaya G650 ↔phone
Preformatted text|_________________________ IP phone
^^^ that is a big change - might require that you be ready to go with “Telco A” and “Telco B”. But even if the same Telco, you will likely be in a totally different department within that Telco. Probably the preferred option would be “Telco B” because then you can have the systems running side-by-side with some test phone numbers, so…
This:
- Telco A ↔ E1 ↔ Avaya ↔ old phone
And this at the same time for testing:
- Telco B ↔ SIP ↔ Asterisk ↔ new phone
Eventually, all sorts of things are possible simultaneously:
- Telco A ↔ E1 ↔ Asterisk ↔ E1 ↔ Avaya ↔ old phone
- Telco B ↔ SIP ↔ Asterisk ↔ SIP ↔ new phone
- Telco A ↔ E1 ↔ Asterisk ↔ SIP ↔ Telco B
- old phone ↔ Avaya ↔ E1 ↔ Asterisk ↔ SIP ↔ new phone
- new phone ↔ SIP ↔ Asterisk ↔ E1 ↔ Telco A
etc.
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