>> On 3/31/2010 12:16 PM, Philipp von Klitzing wrote: >> > >> > Maybe rtptimeout in sip.conf is involved (and not behaving as expected)? >> > >> I was suspecting something with either rtptimeout or sip registration >> timeout, but I'm not sure what.
Hi, I have had similar issue. I have downgraded from 1.6 to 1.4 and issue got solved. Never managed to find what is going on. It was happening only if all were true: - linksys phone or pap - asterisk 1.6 - use certain VOIP provider. Solution: moved to 1.4 I hope thsi helps. Peter -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
