We are integrating Retell AI with Asterisk/Issabel.
We tested both chan_sip/UDP and PJSIP/TCP using register-phone-call and dynamic SIP URI dialing.
SIP signaling works correctly:
- Asterisk sends INVITE to call_xxx@sip.retellai.com
- Retell replies 100 / 180 / 200 OK with SDP
- Asterisk ACKs successfully
- Codec negotiated: PCMU/ulaw
Asterisk publishes public RTP correctly:
- c=IN IP4 45.161.109.142
- RTP port is within 10000-20000
Retell publishes media IP/port correctly:
- 161.115.x.x
- dynamic RTP port
However, Asterisk receives zero RTP packets from Retell.
Evidence:
- chan_sip channelstats: Receive packets = 0, Transmit packets > 0
- PJSIP channelstats: Receive Count = 0, Transmit Count = 1182
- tcpdump -i any shows RTP from 45.161.109.142 to 161.115.x.x
- tcpdump does not show RTP from 161.115.x.x back to 45.161.109.142
- Local firewall allows UDP 10000-20000
- direct_media is disabled
- rtp_symmetric/force_rport enabled in PJSIP test
Retell recording contains bot audio only, but the called phone does not hear the bot.
Please verify why Retell/LiveKit is not sending RTP back to our Asterisk RTP address.