Hi all, We're currently running a PSTN -> SIP gateway with Asterisk. We also run IAX/SIP -> PSTN.
We have performed a test where the call is routed UK PSTN -> Digium E1 card -> Asterisk GW -> SIP G.711 -> FWD -> X-Ten softphone There is no echo at the softphone end, but severe echo on the PSTN side. We've also performed a test BRI PBX -> AVM Fritz CAPI -> Asterisk -> IAX -> Asterisk GW -> E1 -> UK PSTN Once again, no echo at the BRI side, but some echo at the UK PSTN side. We do have echocancel=yes / echocancel = 128 on the Asterisk machine with the E1 card. Is there any other options we can turn on / look at to test this? Also, are their any 'README's which document the best usage of the options in (for example) the zaptel makefile for MMX and more agressive echo cancel etc. We havnt tried these yet as we were unsure of any caveats? Linus _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
