Hi!
I’m not very familiar with VoIP telephoning.
I want to make a simple call to a phone number in my country (Bulgaria, +359) and play some music.
Until now I used a SIP accont from Flowroute (using sip.conf configuration).
Everything works, but now I want to be my own SIP provider.
Is it possible?
Can I just attach some VoIP device to my linux server and start making calls?
Thank you!
Use Asterisk with a Digium FXO card. I think that is basically how Asterisk started. (There are standalone gateway devices on the market, as well.)
I need, for example, to connect my local phone line to the gateway.
Then gateway to Asterisk’s server.
Then use Asterisk to make calls (via local phone).
Is this going to work: cisco.com/en/US/products/ps11977/index.html
There are a lot of SIP gatways in market . GrandStream . Quintum Tenor - Audio Codes - ATCOM , etc .
SPA112 is FXS Analog terminal adapter . It means you can connect it to the Phone and not PSTN line . If I were you I would choose some products that have documentation on how to configure it with asterisk . All scenario are the same but for the first use I will prefer not to use linksys ATAs .
[quote=“htorbov”]
Is this going to work: cisco.com/en/US/products/ps11977/index.html[/quote]
That appears to be an ATA, and FXS interface device.
If you don’t want to do it just with Asterisk, there are FXO gateways, but they are less common than FXS ones.