No Audio After Answer¶
Symptom: call answers but both sides are silent. No "hello?" gets through either way. Call stays connected for the full duration of someone's patience.
Different from one-way audio (one side hears) and from "call disconnects immediately" (the call doesn't stay up). This is bidirectional silence with the dialog still alive.
Most Likely Causes¶
- Codec mismatch — both sides agreed on a codec in SDP that one side can't actually encode/decode. Audio frames sent, decoded as garbage / discarded.
- MoH playing instead of bridge — the call is on hold (one side hit hold) and the other hears MoH (or silence if MoH is empty).
- Wrong RTP port range firewalled — RTP is silently dropped at a firewall mid-path. SIP works fine.
callcounterglitch / call parked accidentally — UnifiedBX bug that parks rather than bridges.- Phone microphone hardware-muted on both sides — rarer but happens, especially with new headsets.
Diagnose¶
# During the silent call:
asterisk -rx "core show channels"
asterisk -rx "core show channel <channel-id>"
# Look at "Bridged with" — should show the other leg.
# Watch RTP stats:
asterisk -rx "rtp set debug on"
# Then in another terminal:
tail -f /var/log/asterisk/full | grep -i rtp
# Should see "Sending packet" and "Received packet" entries from both sides.
# If you only see Send-and-no-Receive, RTP is unidirectional — firewall.
# If you see neither, codec or call state issue.
Fix¶
- Lock both sides to one codec for testing. Trunk and extension both set to
ulawonly. If audio returns, walk codecs back one at a time to find the offender. - Verify the call isn't parked / on hold —
core show channelsshould show both legsUp, notHold. - Check System Admin → Firewall RTP rules; verify with
iptables -L -n | grep 1000. - Restart RTP module if state is corrupted:
asterisk -rx "module reload res_rtp_asterisk.so".
When to Escalate¶
- RTP packets visible in both directions in
rtp set debug onoutput, but still silent. That's a phone or carrier endpoint issue, not UnifiedBX.