Try removing nat=yes On Aug 28, 2013 1:19 PM, "王鹏" <[email protected]> wrote:
> > HI,all > > my asterisk11 have a domain name with dynamic ipaddress. > i want to use asterisk and linphone make p2p call, i open icesupport in > sip.conf and rtp.conf, can call through but no voice. > > here > https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support > i see ICE support is only used for communication between a remote endpoint > and Asterisk. > > so i close icesupport in sip.conf and rtp.conf, but invite from caller > have ice paramter,but asterisk modify invite message, delete ice paramter, > then pass to callee, so phenomenon is also can call through but no voice. > > how to make asterisk not modify sdp of invite, or have other method to use > asterisk and linphone make p2p call. > > thanks!! > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz >
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