Hi, I am breaking my fingers in configuring an asterisk (1.6) to successfully transmit audio with the following setup:
asterisk, resides in local network, ip is 10.26.208.252 versatel business router (directly connected to a dsl, configured by sip-provider), WAN ip 89.244.13.25 versatel sip-proxy ip 89.244.13.10 in sip.conf I have: [general] bindaddr=0.0.0.0 externip=89.244.13.25 localnet=10.26.208.0/255.255.252.0 nat=yes qualify=yes the local sip phones register correctly and can make calls between each other with audio. the local sip phones CAN make outbound calls via the sip-provider... will say, destination phone rings, but there is no audio (on both legs) after pickup... external phones can call my sip-number... the call comes into the asterisk, the sip-extension rings, but after pickup... no audio at all. even if i route the call from external to a queue or something else... i see, that asterisk is playing voicefiles, but the caller does not hear anything. because sip-signalling works in any ways, but audio not, i think its got something to do with nat... but there is no firewall between asterisk and the router or between the router and the internetconnection from versatel... and i already tried millions of combinations of using nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m stuck as i was never ever stuck before :-((((( any hints? anybody? thanks, yves -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
