I am running the current 1.4 release of Asterisk, it has a SIP config registered to my Voipfone account. I map UDP and TCP port 5060 to my asterisk server inside my NAT.
A lot of stuff is working correctly.
I am testing inbound, by calling my Voipfone 056 number from a landline.
The internal extension rings.
I get through to my voicemail.
As soon as I start recording a message, the Asterisk CLI gets flooded with this warning :
[Dec 9 16:09:34] WARNING: rtp.c:891 ast_rtcp_read: RTCP Read too short
Only the first second or two is recorded.
If I pick up the extension instead of letting it go to VM, audio seems fine.
Has anyone got an idea what is going wrong?
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