Topic/Body:
I’m building a call center using Asterisk 18 with ARI (Asterisk REST Interface). The call flow works (calls connect), but two-way audio is not bridging properly.
Setup:
-
Customer calls in → enters
myappStasis application
-
Agent (extension 100/101) is a PJSIP endpoint with
context=ari-agents -
When agent answers, I create a Local channel:
Local/100@local-bridge -
The
local-bridgecontext answers the channel before Stasis
-
I create a bridge and add both customer and agent channels
Issue:
After bridging, no audio flows in either direction. The channels are in the bridge, but silence.
What I’ve tried:
-
Added
Answer()in
local-bridgecontext before
Stasis() -
Updated Java code to use
Local/+ extension +
@local-bridge -
Added
isChannelInBridge()polling every 2 seconds
-
Verified bridge membership via ARI
Configuration:
-
pjsip.conf: agents use
context=ari-agents -
extensions.conf:
[ari-agents]and
[local-bridge]contexts
-
Java: AriWebSocketClient handles bridging logic
Question:
What am I missing in the bridge setup? Is the Local channel approach correct, or should I use a different method to connect the agent to the customer bridge?