Hello Matthew, My version is Asterisk 1.6.2.9.
Or have you seen NAT? I have no NAT on my network. Have you seen my little diagram above? Here it is: 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 ) My Asterisk server has two NIC/interfaces. - 1 interface with public IP (109.69.217.6 to talk with SIP friends) - 1 interface with internal ip (10.4.0.1 to talk with SIP gateway's) SIP friend should not even know that the call is routed to the SIP/PSTN gateway. It could be a SIP trunk to a SIP provider Internet, the user does not have to know... Best regards, Mickael 2013/6/13 Matthew J. Roth <mr...@imminc.com> > 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:xxxx@80.236.215.61:64946;ob SIP/2.0 > > Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport > > Max-Forwards: 70 > > From: < sip:xxxx@109.69.217.6 >;tag=as15b47581 > > To: "test" < sip:xxxx@109.69.217.6>;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh > > Contact: < sip:xxxxx@109.69.217.6 > > > 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 >
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