- When you perform the attended-transfer(particularly the SIP REFER method) by the transfer button on the IP phone for the incoming call. After the transfer of the call, it would have the one-way audio or no audio on both sides for the caller and the transfer target party.
- The incoming call is forwarded by the extension always forwarding settings.
- This article also applies to the situation with the calls via SIP trunk become no audio after a specific time like 15 mins.
1. Try to disable the Allow RTP Re-Invite option in the SIP Advanced setting.
2. Try to disable the Jitter Buffer for the trunk that involves the forwarding or transfer.
If this solution doesn't help, try to seek help from our support.
This article also applies to the situation with no audio after PBX receives a Re-INVITE from the SIP trunk or external SIP device.
I came across this article because I got the problem that an incoming call is forwarded by the extension always forwarding settings - the forwarding itself works, but I got no audio.
I tried the solutions in this article, but nothing worked.
Maybe there is another suggestion?
I exactly the same problem has Vera Reisner
When a inbound route by DID call extension directly if an always forwarding is enable on phone, the forwarding itself works, but I got no audio.
If exension is call by IVR or queue the always forwarding is enable on phone, all is ok.
One of our customers has a one-way audio issue after call transfer. We tried all settings that mention here + Modem/Firewall NAT configuration. Discussed with ITSP so and so and problem not solved after few days spend on it.
After all, today the customer enabled the Jitter buffer, strangely the problem solved! I want to share here. Maybe someone faces the same issue.
I've been struggling with the same problem for a coiuple of days and have tried all solutions mentioned here. Turning on the jitter buffer has also worked for me.
Hello Paul Glenton
I'm happy to see others benefiting. I'm glad the issue was resolved. Good work
Thanks a lot, my problem is solved with disable the Allow RTP Re-Invite.