Mickael MONSIEUR wrote: > > I have a standard Asterisk configuration: > > SIP friends (phones) <-----> Asterisk <-----> SIP gateway to PSTN converter > 80.236.215.61 109.69.217.6 internal IP ( > 10.4.0.10/255.255.255.0 ) > > When analyzing traffic on a SIP friend/phone I see this: > > INVITE sip:[email protected]:64946;ob SIP/2.0 > Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport > Max-Forwards: 70 > From: < sip:[email protected] >;tag=as15b47581 > To: "test" < sip:[email protected] >;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh > Contact: < sip:[email protected] > > Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM > CSeq: 102 INVITE > User-Agent: Asterisk > Require: timer > Session-Expires: 1800;refresher=uas > Min-SE: 90 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 217 > > v=0 > o=root 664087974 664087976 IN IP4 10.4.0.10 > s=Asterisk > c=IN IP4 10.4.0.10 > t=0 0 > m=audio 8652 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > My equipement IP 10.4.0.10 is visible to the user, why?
Mickael, What version of Asterisk are you running? Is the Asterisk server outside and the SIP gateway to PSTN converter inside of a NAT? What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
