[asterisk-users] intermittent problem on 1.4

[John Taylor]

We're trying to forward an incoming SIP call from voipfone (UK ITSP) that
originated from a UK landline back up a SIP trunk to the same ITSP and on to
another UK landline number.

UK Landline->voipfone->asterisk 1.4->voipfone->UK landline

About 1 in 3 times the call at the final landline is silent and we see "RTP
Read too short" scrolling on the console log.

Where do we start working out what's going on? Other than that the server is
working well

John
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