Hi! I thought it was the SIP device too, but I have looked for avery litle comand of this device and I can't find this Ip address, and I see that its Ip is Ok, and I have configurated the REGISTRAR section too... I don't know what's happening, and I don't understand that, if the IP is wrong, why can I hear the callee phone ringing and the call only goes off when I pick it up?
it's so strange...I think! Michelle >On Mon, 16 Jun 2003, michelle matis litio wrote: >> to 229.159.241.112:5060 >> Retransmitting #5 (no NAT): >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK- 3a5246f7- >> 8c6b606-10eb >> From: ;tag=0-13c4-3a5246f7- 8c6b604-c3a >> To: ;tag=as52ed0a6a >> Call-ID: <A href="javascript:sendMsg('f93b00-0-13c4-3a5246f7-8c6b602- [EMAIL PROTECTED]');">f93b00-0-13c4-3a5246f7-8c6b602- [EMAIL PROTECTED]</A> >> CSeq: 1 INVITE >> User-Agent: Asterisk PBX >> Contact: >> Content-Type: application/sdp >> Content-Length: 135 >> >> v=0 >> o=root 11673 11673 IN IP4 188.208.12.237 >> s=session >> c=IN IP4 188.208.12.237 >> t=0 0 >> =audio 13532 RTP/AVP 0 >> a=rtpmap:0 PCMU/8000 >Hi, >Its being sent to that IP address, because that is that the >originating SIP device put in its Via header. >Also, your SIP device didn't put any From or To in its INVITE. >Perhaps you could send a sip debug from the start of a SIP call >attempt. >But I'm sure that the trouble is with your SIP Gateway device's >setup. >Steve >_______________________________________________ >Asterisk- Users mailing list ><A href="javascript:sendMsg('Asterisk- [EMAIL PROTECTED]>http://lists.digium.com/mailman/listinfo/asterisk- users');">[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users</A> ----- Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com Ya.com ADSL Home 24 h, Módem + Alta ĄGratis! http://acceso.ya.com/adslhome24h/ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users