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. <br><br>UK Landline->voipfone->asterisk 1.4->voipfone->UK landline<br>
<br>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.<br><br>Where do we start working out what's going on? Other than that the server is working well<br>
<br>John<br>