The guide
Everything a phone system needs to send calls to a station: where the gateway is, how it will know you, the configuration for the systems we are asked about most, and what each refusal means.
The gateway has to know a call is yours before it answers. Pick whichever suits how your PBX reaches the internet; the audio and the dial codes are the same in all three.
The gateway recognises the address your PBX calls from and takes its calls without asking for anything.
We issue a user name and a password once. Your PBX answers the gateway’s challenge with them on each call, or registers.
The platform signs in to your PBX as one of its extensions, bound to one station. Calling that extension plays the station.
Choose how you will connect, the prefix your people will dial and a station to test with. What follows is written for exactly that.
No secrets in your configuration. Tell us the address your calls come from and we add it to your trunk.
Up to four keys: digits, * or #.
A dial code: a calling code, then five digits.
With this in place, 7 49 00001 from any extension plays the station at that code.
Plain Asterisk with chan_pjsip, version 16 or later. Edit the two files, then reload: “pjsip reload” and “dialplan reload” at the Asterisk console.
Where the gateway is, which codecs to offer and, if you connect with a user name, where it goes.
[sidekick-radio]
type = endpoint
context = from-sidekick-radio ; nothing calls back: this only has to exist
disallow = all
allow = opus,g722,ulaw,alaw ; Opus first
aors = sidekick-radio
direct_media = no
rtp_symmetric = yes
; no user name or password: we know you by the address you call from
[sidekick-radio]
type = aor
contact = sip:18.133.93.19:5060Whatever your people dial after the prefix is sent down the trunk with the prefix taken off.
; in the context your phones dial from
; 74900001 plays the station at 4900001
exten => _7X.,1,Dial(PJSIP/${EXTEN:1}@sidekick-radio)
same => n,Hangup()
; 70 is the browse menu
exten => 70,1,Dial(PJSIP/0@sidekick-radio)
same => n,Hangup()Call 74900001 from any phone. You should hear the station within a second or two. 70 is the browse menu.
The trunk agrees one codec per call, best first. Opus only wins if your PBX puts it in its offer, and it only reaches the listener unchanged if their phone speaks it too.
| Your side | What is heard |
|---|---|
| Trunk and phone both speak Opus | 20 kHz, as broadcast |
| Trunk in Opus, phone in G.722: your PBX transcodes | 7 kHz |
| Trunk offers G.722 but not Opus | 7 kHz |
| Trunk offers only G.711 | 3.4 kHz |
To check, look at the codec of a call in progress on your PBX. On Asterisk, “pjsip show channelstats” names it. The operator sees the same thing, per call, in the live view.
Calls go from your PBX to the gateway, so most firewalls need nothing opened: only outbound traffic allowed and its replies let back in.
| For | Port | Protocol |
|---|---|---|
| Signalling | 5060 | UDP · TCP |
| Signalling, encrypted | 5061 | TCP |
| Call audio | 42000–42199 | UDP |
All on 18.133.93.19. The audio range is the platform’s default; your operator confirms it with your trunk.
The gateway refuses a call before answering it, with a status that says why. Your PBX’s log shows it.
What arrived is not a dial code, a mapped number or the browse menu. Most often the prefix was not stripped: the gateway must receive 49 00001, not your prefix and then the code. If your PBX cannot strip it, ask us to strip it on your trunk.
The station’s queue has reached its limit of listeners, or your trunk its limit of simultaneous calls. Try again shortly, or ask for the limit to be raised.
The station’s own stream is not answering just now. Other stations are unaffected. Station pages on this site show each one’s health.
The call did not come from the address on your trunk, or the user name and password did not match. Check what address your PBX really sends from; behind NAT it is the router’s.
Your offer contained none of Opus, G.722 or G.711. Allow at least G.711 on the trunk.
Signalling got through and audio did not. Open the audio port range for UDP, switch off SIP ALG, and make sure your PBX is not advertising a private address for its audio.
This is an operator-run service: trunks are set up by a person, with you. Say which phone system you have and how you would like to connect, and we reply with what it needs.