Hello, I have installed freeSwitch 1.1beta from the git repository (FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000). My SIP provider (Netelip) is defined in conf/directory/default/example.conf.xml: <include> <user id="$${default_provider}"> <gateways> <gateway name="$${default_provider}"> <param name="username" value="$${default_provider_username}"/> <param name="password" value="$${default_provider_password}"/> <param name="from-user" value="$${default_provider_username}"/> <param name="from-domain" value="$${default_provider_from_domain}"/> <param name="expire-seconds" value="600"/> <param name="register" value="$${default_provider_register}"/> <param name="retry-seconds" value="30"/> <param name="extension" value="$${default_provider_contact}"/> <param name="auth-calls" value="false"/> <!--<param name="contact-params" value="domain_name=$${domain}"/>--> <param name="context" value="public"/> </gateway> </gateways> <params> <param name="password" value="$${default_provider_password}"/> </params> </user> </include> and doesn't authenticate inbound calls (insecure=invite in asterisk language). When I receive an inbound call, I receive the following error: freeswitch@internal> recv 1000 bytes from udp/[194.140.135.80]:5060 at 23:23:10.163377: ------------------------------------------------------------------------ INVITE sip:34810101189@sukmad.dyndns-home.com SIP/2.0 Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport From: "916495242" <sip:916495242@194.140.135.80>;tag=as7a2a97e5 To: <sip:34810101189@sukmad.dyndns-home.com> Contact: <sip:916495242@194.140.135.80> Call-ID: 6c47a4064ef8d0641da415d2121e3226@194.140.135.80 CSeq: 102 INVITE User-Agent: Netelip Max-Forwards: 70 Date: Fri, 13 Apr 2012 23:23:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 425 v=0 o=root 1249 1249 IN IP4 194.140.135.80 s=session c=IN IP4 194.140.135.80 t=0 0 m=audio 16474 RTP/AVP 8 0 97 3 18 4 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ send 382 bytes to udp/[194.140.135.80]:5060 at 23:23:10.164409: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060 From: "916495242" <sip:916495242@194.140.135.80>;tag=as7a2a97e5 To: <sip:34810101189@sukmad.dyndns-home.com> Call-ID: 6c47a4064ef8d0641da415d2121e3226@194.140.135.80 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000 Content-Length: 0 ------------------------------------------------------------------------ 2012-04-14 01:23:10.162331 [DEBUG] sofia.c:7598 IP 194.140.135.80 Rejected by acl "domains". Falling back to Digest auth. send 871 bytes to udp/[194.140.135.80]:5060 at 23:23:10.166026: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 194.140.135.80:5060;branch=z9hG4bK6bb16a10;rport=5060 From: "916495242" <sip:916495242@194.140.135.80>;tag=as7a2a97e5 To: <sip:34810101189@sukmad.dyndns-home.com>;tag=SaBHcvmm7evNF Call-ID: 6c47a4064ef8d0641da415d2121e3226@194.140.135.80 CSeq: 102 INVITE User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-d2edcad 2012-03-31 23-42-27 +0000 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="194.140.135.80", nonce="bc1571d4-0957-4609-a887-7c3359b9ae9c", algorithm=MD5, qop="auth" Content-Length: 0 I have read in forums that the solution to this problem is to add the IP to the acl.conf.xml, like this: <list name="domains" default="deny"> <!-- domain= is special it scans the domain from the directory to build the ACL --> <node type="allow" domain="$${domain}"/> <!-- use cidr= if you wish to allow ip ranges to this domains acl. --> <node type="allow" cidr="194.140.135.80/24"/> </list> But my SIP provider have a dynamic IP, and it's behind a NAT. If my SIP provider will use any IP, ¿Where can I do this without enabling all IPs to enter calls in my system? Thanks in advance, Regards.