Hello,
I configured asterisk in sip.conf like that: ===== register => username:[email protected]:5060/number [sipgate-out] port=5060 type=friend insecure=invite nat=yes username=username fromuser=username fromdomain=sipgate.de secret=secret host=sipgate.de qualify=5000 canreinvite=no ===== But all I get on CLI is: [Jan 19 15:59:55] NOTICE[1928]: chan_sip.c:12104 sip_reg_timeout: -- Registration for '[email protected]' timed out, trying again (Attempt #1) My first impression was that there must be a NAT related issue, so I captured some SIP debug packets and got: ===== Retransmitting #6 (NAT) to 217.10.79.9:5060: OPTIONS sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.23.30:5060;branch=z9hG4bK4c5e9a7f;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as669e138a To: <sip:sipgate.de> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.21 Date: Thu, 19 Jan 2012 15:00:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 ===== As you can see, asterisk writes the local ip address in the VIA header. Is this the supposed behaviour? As far as my understanding of the NAT problem goes, this is exactly what I don't want to be happening. I'm familiar with the general NAT problem with layer 7 protocols relying on ip addresses in the content of a layer 3 packet, but I thought, asterisk would replace the local ip with the WAN ip. Even setting "externip=xx.xx.xx.xx" does not change anything. Anyone can give me advice? kind regards, Ruben -- _____________________________________________________________________ -- 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
