Hi There,
I am IT guy but with limited knowledge in telephony/VoIP/SIP.
I have running FreePBX on Raspberry Pi. Extensions created, calls internally work fine. This will be a TEST server and never be in production.
I now need to create some basic inbound/outbound PSTN calls (again, test calls only) and I don’t know where to start. The biggest challenge probably is that I have absolutely no access to firewall or router configuration on this network. What kind of service (authentication) will I need to look for? Any recommended providers/guides?
For inbound calls, the goals to have prompt for extension to route the call. Thanks.
If you are behind an “only necessary services are permitted; everything else is blocked” firewall, you may be out of luck. If your Pi is new enough to have Wi-Fi, you could set up your mobile phone as a hotspot and connect the Pi to that, avoiding the corporate firewall.
Assuming that you are in US or Canada (if not, please post your country), you might try:
SignalWire; they give you a small credit at signup so you can test without making a payment. This is standard SIP, so it won’t work if your firewall blocks it.
Callcentric; also SIP, but uses nonstandard ports so less likely to be blocked. A free account can make toll-free calls and receive calls via SIPBroker. If you get past that, you can deposit as little as $5 to make and receive regular calls.
VoIP.ms; offers IAX connectivity, which might bypass attempts to block SIP. $25 minimum deposit but unused balance refunded if you close the account.
CallWithUs; offers a VPN through which you can do standard SIP.
Another possibility is setting up your PBX on a cloud server, where it will have unrestricted connectivity. If you can’t find a way to get your extensions through the company firewall, you could just use external numbers for your demo (calling an extension would ring to a corresponding mobile phone).
SW: A DID is only $0.08/mo., so no reason not to get one (though you are charged $0.00325/min. for incoming calls).
CC: Without a DID, your caller ID is one of theirs, or a number that you verify as yours.
VMS: Without a DID, you can send any valid number as caller ID.
CWU: They don’t even sell DIDs, but you can get one from a third party and route it to their system by SIP URI.
I was able to test with SignalWire test account by following their guide https://signalwire.com/resources/guides/freepbx. Outbound calls work but I can’t get anything inbound. Thoughts?
For inbound calls, Asterisk CLI shows this when I am trying to connect and call drops:
Setting global variable ‘SIPDOMAIN’ to ‘FreePBX IP Address’
Setting global variable ‘SIPDOMAIN’ to ‘FreePBX IP Address’
Setting global variable ‘SIPDOMAIN’ to ‘FreePBX IP Address’
Setting global variable ‘SIPDOMAIN’ to ‘FreePBX IP Address’
Setting global variable ‘SIPDOMAIN’ to ‘FreePBX IP Address’
(I’ve replaced the actual IP address).
Thoughts?
Don’t just look at the CLI, view the complete log at Reports -> Asterisk Logfiles. If there are insufficient clues, at the Asterisk command prompt, type pjsip set logger on
Make another attempted call and look at the INVITE coming in and the response given. If you have trouble interpreting them, paste the complete log for a call at https://pastebin.freepbx.org and post the link here.
SignalWire support helped me out. It was “Encryption: Optional” in SignalWire SIP endpoints that was the root cause. Changed to required, adjusted codecs and chipper and I am in bussines.
For anyone else that will be in my boots, SignalWire has great support. I tried FlowRoute and their supports was really bad (I assume they are more commercial oriented).