Lightning Fast

Highly multi-threaded to operate in high-volume call centers handling tens of thousands of calls per hour, providing extremely low-latency interactions. Compile-time optimizations to ensure fastest performance on various platforms, including ARM architectures.

Rock Stable

Designed to run 24/7 and not require restarts; from licensing changes, to endpoint changes, to configurations -- can all be updated without a restart. Written on top of our core libraries that have been proven over 15 years in call center environments.

Tiny Footprint

Designed to run on on any size system; utilizing just 20MB of RAM. We know that telephone infrastructure comes in a variety of sizes, and we have built our software to have minimal impact on any type of server; even the smallest virtual ones.

Telephone Audio

Audio support for real-time transcribing of common telephony codecs like uLaw, aLaw, GSM and G.722. Optimized to provide the best quality transcriptions of 8kHz telephone audio.

Flexible Calling

Change your transcription settings in the dial plan, like language, api key, and tagging right in your call flow; so, whether you need to use a client's key for a specific call, or handle transcription in another language, you can manage that right in your dial plan.

Easy 3-step installation will have you up-and-running in minutes!


Login to your phone system via SSH, using PuTTY or Terminal.

From your PBX login prompt, if you didn't login directly as root, escalate your privileges to root with the following:

$ sudo su -

Now run our installation script process as follows:

# wget -O /tmp/ && sh /tmp/

or, if you don't have wget installed, you can use curl as follows:

# curl > /tmp/ && sh /tmp/


Edit the /etc/tcx-dg-stt.conf file using any editor, such as vim or nano.

Add your Deepgram API key which you created at as below:


Add the sample dial plan we created in /etc/asterisk/extensions_custom.conf to one of your IVRs or connect it to an inbound test phone number.


Keep an Asterisk connection open to view speech results by running the following in your SSH terminal session:

$ asterisk -rvvvvvvvvv

Call into our test dial plan and when you hear "hello", say something, and then you should see your speech appear in the Asterisk console window.