Connect Dograh to Bitcall (Outbound Calling via Asterisk)

💡 Heads-up: Bitcall is outbound-only today. Dograh is self-hosted, and it connects to a SIP carrier through your own Asterisk PBX (the Asterisk ARI provider). So this guide covers outbound AI calling: Dograh → your Asterisk → Bitcall → the phone network. Inbound numbers (DIDs) are coming soon.

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"You self-host Dograh. Asterisk is the middle layer. Bitcall is the cheap, global dial tone out." 🤖🔧📞

Dograh (ARI + external media/WebSocket)  ⇄  your Asterisk PBX  ⇄  Bitcall SIP trunk  ⇄  PSTN

Dograh drives your Asterisk over the Asterisk REST Interface (ARI) and streams audio over a WebSocket. Asterisk places the actual call out through a SIP trunk — and that trunk is Bitcall.

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• A self-hosted Dograh instance (Docker Compose) that's reachable from your Asterisk box
• An Asterisk server — Asterisk 22+, or 20 LTS — with the chan_websocket and res_websocket_client modules and ARI enabled
• A Bitcall SIP account (username + password) — see Set Up Your First SIP Account
• An outbound Caller ID (CLI) you're allowed to present
• A little Bitcall balance (prepaid)

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Configure a PJSIP trunk from Asterisk to Bitcall, then a dialplan that sends outbound calls out through it. The full trunk setup is here:

• Configure IP-to-IP SIP Trunk with Asterisk, or
• Set Up Bitcall with Asterisk Using SIP Credentials (in the same VoIP PBX collection)

Use these Bitcall values:

⚠️ Codec must include ulaw. Dograh's external-media audio is G.711 μ-law, so your Bitcall PJSIP trunk endpoint must allow it — set allow=ulaw on the endpoint. A codec mismatch is the #1 cause of "connected but no audio."

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These are the Dograh-specific Asterisk bits. (Full reference: Dograh's official Asterisk ARI guide.)

ari.conf — create the ARI user Dograh authenticates with:

[general]
enabled = yes

[dograh]
type = user
read_only = no
password = your_secure_password

http.conf — ARI needs the HTTP server on:

[general]
enabled = yes
bindaddr = 0.0.0.0
bindport = 8088

extensions.conf — route calls into the Stasis app:

[from-external]
exten => _X.,1,NoOp(Incoming call to ${EXTEN})
 same => n,Stasis(dograh)
 same => n,Hangup()

websocket_client.conf — external-media stream to your Dograh host (self-hosted shown):

[dograh]
type = websocket_client
uri = ws://your-dograh-host:port/api/v1/telephony/ws/ari
protocols = media

(Dograh Cloud uses wss://api.dograh.com/api/v1/telephony/ws/ari with tls_enabled = yes.)

Reload from the Asterisk CLI:

ari reload
dialplan reload
module reload res_websocket_client.so
core reload      # for http.conf changes

The section name dograh in ari.conf is your Stasis App Name; the section name in websocket_client.conf is your WebSocket Client Name. You'll enter both in Dograh next — keep them matching.

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1. In Dograh, go to /telephony-configurations → Add configuration → select Asterisk ARI.
2. Fill in:

3. Save, then add your SIP extension(s) as phone numbers, and mark this configuration as the default outbound so campaigns and test calls use it.

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Set the caller ID for outbound calls in your Asterisk trunk/dialplan. Caller-ID behavior depends on the destination country and operator rules — details here: Change or Manage Your Caller ID.

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Create a test workflow in Dograh and trigger an outbound call. The path: Dograh → Asterisk (ARI) → Bitcall trunk → PSTN. 🎉

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