Hot answers tagged pstn
8
IPX? :-)
Doing a google search on the phrase you're asking about seems to indicate that you can't send calls over this VoIP link if it originates on POTS. If someone calls into your HQ office from a POTS line, and you then route it to one of your branch offices over your provider's VoIP link, you're in breach of your contract, and they're probably in ...
5
Phone numbers and "SIP URIs like me@domain.com" are different kinds of URIs. A phone number can be represented as a URI like this:
tel:+12125551212
whereas a "SIP URI like me@domain.com" can be represented as a URI like this:
sip:me@domain.com
Many SIP user agents don't actually use tel: URIs. Instead they just take the number that the user dialed and ...
4
In your output, there is an error from the format_wav.c module saying the input is not a wav file. Check your format and make sure that you're saving it in the format Asterisk expects.
As a workaround, you can use the open source audio tool "sox" to make things a little easier:
sox /var/lib/asterisk/sounds/custom/Untitled24.wav -r 8000 -c 1 ...
3
There are a variety of terminal servers on the market -- you telnet/ssh to them and upon connection, you can interact with one or more serial ports on the console server, which are typically hooked via a null-modem cable to other servers. If the server has a serial port, this is the best way to go. You can then use a mobile broadband device to get a ...
3
I suspect there are lots of ways to do this. One that comes to mind is callback.
You register a dialin ISP account for the modem on the server; that's simple enough, and non-contentious. You will need an ISP that will allocate you a static address; if this costs extra, pay it.
You then configure something like diald to bring the link up and down on ...
2
You're correct that you need an FXO/FXS card presuming that your phone lines will be analog. I've used cards from both Digium and Sangoma. I'd recommend sticking with the major manufacturers of these cards for Asterisk as this is your first setup and you want it as simple as possible. Digium will be a little simpler but I'm partial to Sangoma cards as ...
2
I'd add this as a comment but I don't have enough rep yet:
Where I've seen the term Non-IP traffic in the past generally refers to how providers will charge you for incoming calls from the PSTN. Basically, hosted VOIP providers are trying to mimic the charge conditions you'd get with an in-house PBX. Corporate PSTN accounts generally have a per-minute ...
1
If it's a single physical location, the cheapest and simplest solution would be to use a traditional analogue PBX. They all support internal calling and don't need to be connected to PSTN.
If you must use a software solution then consider Asterisk - but you'll need to buy some FXO hardware to terminate your handset connections.
1
I personally would use Vitelity. I like being able to provision numbers on the fly via the web interface. There is no upselling (no humans to talk to, for that matter), and although they specialize in SIP termination, it definitely is possible to configure any number to "failover" to a PSTN number if no SIP device is available to ring when a call comes in. ...
1
Not sure this belongs to SF but you could get most modems using Intel 537P (AKA Ambient MD3200 ) chipset to be recognized as Digium Wildcard X100P clone by Asterisk. Do not confuse it with Intel 537EP (newer but unsupported chip).
537P is rectangular chip
537EP is square 64-pin TQFP
Also pay attention to which version G(global) or U(USA) you ...
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