I have migrated from Asterisk 1.4 to Asterisk 11 (11.6-cert13).
This is the problematic scenario:
- A call enters from a SIP client (through Kamailio Proxy) to Asterisk extension.
- The dialplan executes and the call is answered and redirected to the queue application.
- The caller receives the correct RTP packets for the “ring” sound of the queue.
- The call is assigned to an agent and the agent answers.
- SIP call is re-invited for directmedia.
- The caller receives the correct RTP packets from the agent.
- The agent sets the call to hold (SDP: sendonly).
- In Asterisk verbose you can see: “Started music on hold, class ‘default’, on SIP/(…)”
- The caller does not receive any RTP packets from Asterisk. So there is no music on hold playing.
- In Debug log I can see the following traces repeating until I exit the “hold” of the agent:
DEBUG[C-00000005] res_rtp_asterisk.c: Remote address is null, most likely RTP has been stopped
This worked before in version 1.4. But neither Asterisk 1.8 nor Asterisk 11 transmit the music correctly.
I suspect some error in my configuration. Where can I begin?