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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110119/71a101af/attachment.htm>
John Taylor wrote: [snip]> Where do we start working out what's going on? Other than that the > server is working well > > Johncould you please ilustrate a little bit more your scenario?, (if you want, use fake IPs). Note: What's the exactly version number of your Asterisk box? -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs
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