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SIP setup, without the guesswork

Connect your own trunk, check it, place a test call, and read a trace of your recent SIP activity, so a failure is something you can read rather than something you report.

Any SIP provider
G.722, G.711 offered on your trunk — HD first, standard otherwise
Measured region latency, probed by us rather than reported by a carrier

Point the trunk

Choose Connect a trunk on Numbers & SIP and enter what your provider's portal shows: host, port, transport, username and password, and your numbers. Check the connection, which rings nobody, then place a test call. The trunk's recent SIP activity shows each call setup, the SIP response and the codec negotiated. A row we recorded ourselves rather than observed is labelled as such, because a trace you cannot trust is worse than no trace.

Receive conversations, and call out

WeTalk generates sign-in details for your provider and shows the password once; it accepts a call only from the addresses you allow, signed in with those details. Connect a trunk number to an agent on its Channels tab, and the same trunk carries your campaigns and Ring a number now, from your own caller-ID numbers. At launch we ring Greek numbers only.

Media and latency

We offer wideband G.722 first and fall back to G.711. HD is only heard when your provider and every network in between carry G.722, and most calls to ordinary phone numbers are narrowband, because most phone networks are. Each conversation's History shows the codec it actually used. Opus is for browser voice — the test in the console and the website widget — not for SIP. We report round-trip and time to first audio per conversation, so you can see whether a problem is ours or the network's. An address that keeps failing to sign in is locked out, so a leaked password alone does not let someone else's phone system call in as you.

Connect a trunk and test it

Nothing goes live until you place the call yourself.