One-way audio when forwarding SIP call from Primary Asterisk to Secondary Asterisk

Asterisk Asterisk SIP
1

I have a setup with two Asterisk servers: Primary and Secondary.

Incoming SIP calls first arrive at the Primary Asterisk server.
If the called number matches a special DID, the call is forwarded to the Secondary Asterisk server using SIP (UDP).

The Secondary Asterisk:

  • Plays a welcome / IVR message

  • Sends the call into a queue

  • Agents are registered on the Secondary server

Call Flow

Customer / SIP Trunk
        |
        v
Primary Asterisk
        |
        v
Secondary Asterisk
        |
        v
Agent IP Phone

Problem

When the agent answers the call:

  • Customer can hear the agent :white_check_mark:

  • Agent cannot hear the customer :cross_mark:

  • Call signaling is successful

  • Call stays connected

  • Issue is one-way audio

All signaling and RTP are using UDP.

Additional Notes

  • SIP protocol: PJSIP

  • Codecs are compatible

  • No call drop

how fix this issue. I can provide if needed any logs or settings.

Thank you.

2

You’ve offered no proof of this.

PJSIP is an implementation of the SIP protocol. not a different protocol/

We meed to see the actual signalling.

3

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