If I have an incoming call coming down a SIP trunk to a particular
internal SIP extension- I can answer the extension fine, all works

However, if I change extension.conf from dialling the internal
extension to forward the call to an external cell phone (up the same
trunk as the incoming leg of the call) I cannot get any audio and get
the following error message on the console:
[Jan 30 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short

i.e. change from
exten => s,1,Dial(SIP/203,20,t)

exten => s,1,Dial(SIP/07123123...@voipfone,20,t)

What's wrong?!


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