hmm... now tring.. somone to know how can I redirect the output of the "sip debug" into file 'cause it is really hard to grasp (several pages is just one call)
> On Fri, 22 Oct 2004 01:24:24 +0300, raptor <[EMAIL PROTECTED]> wrote: > > On Fri, 22 Oct 2004 01:36:29 +0900 > > Benjamin on Asterisk Mailing Lists <[EMAIL PROTECTED]> wrote: > > > > |The problem is SIP NAT traversal > > | > > |The solution is 'canreinvite=no' > > > > ]- no it is not, the phones are on the same subnet :"( > > In that case, the phones are unable to talk to each other directly for > other reasons, ie configuration or incompatibility. To find out what's > going on, use SIP debug on the Asterisk console. > > rgds > benjk _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
