Hi, can someone please look at it and tell my what I am doing wrong? NOTICE] sofia_reg.c:448 Registering sip-trunk.telekom.de [WARNING] sofia_reg.c:484 Timeout Registering sip-trunk.telekom.de [WARNING] sofia_reg.c:505 sip-trunk.telekom.de Failed Registration [908], setting retry to 30 seconds. My setup: ######## I have a static IP address for my router. My router does not have UPnP or NAT-PMP. Phones and Freeswitch server are on the same local network. The local network is behind NAT PHONE -> FS -> NAT -> Public Internet I only want to connect to the PSTN via my sip gateway provider. According to this manual I did the following setting: https://freeswitch.org/confluence/display/FREESWITCH/NAT+Traversal vars.xml ###### <X-PRE-PROCESS cmd="set" data="external_rtp_ip=X.X.X.X“/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=X.X.X.X"/> internal.xml ######### <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> external.xml ########## <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> ( What is a little bit strange though is that nat_map status is not working [ERR] mod_commands.c:751 nat_map API called while NAT not initialized ) Is the reason why it does not work the port specification in the Contact field (sngrep output)? Contact: <sip:gw+sip-trunk.telekom.de@X.X.X.X:5080;transport=tcp;gw= sip-trunk.telekom.de> I do not want to setup port forwarding in the firewall. The connection should remain open via a keep alive signal. sngrep output ########### 2018/06/30 13:52:37.211946 10.0.200.2:51813 -> 217.0.26.165:5060 REGISTER sip:reg.sip-trunk.telekom.de;transport=tcp SIP/2.0 Via: SIP/2.0/TCP X.X.X.X:5080;rport;branch=z2hG4bK588cZXSvrN1De Max-Forwards: 70 From: <sip:username@reg.sip-trunk.telekom.de>;tag=9aldsj8n67a To: <sip:username@reg.sip-trunk.telekom.de> Call-ID: CSeq: 111844514 REGISTER Contact: <sip:gw+sip-trunk.telekom.de@X.X.X.X:5080;transport=tcp;gw= sip-trunk.telekom.de> Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~32bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 Provider settings: ############# output proxy: reg.sip-trunk.telekom.de registrar: sip-trunk.telekom.de sip_profiles/external/telekom_voip.xml ############################## <include> <gateway name="sip-trunk.telekom.de"> <param name="username" value=„XXXXXX“/> <param name="password" value=„XXXXXX“/> <param name="realm" value="sip-trunk.telekom.de"/> <param name="proxy" value="reg.sip-trunk.telekom.de"/> <param name="register" value="true"/> <param name="register-transport" value="tcp"/> <param name="expire-seconds" value="3600"/> </gateway> </include> Firewall settings ############# # block all traffic in all directions block log # loopback interface is not filtered set skip on lo0 voip_tcp_client_out = "{ssh, ftp, domain, http, https, sip}" voip_udp_client_out = "{domain, ntp, sip, 3478}" match out on $wan_if from any nat-to ($wan_if) pass on $voip_if inet proto tcp from any to port $voip_tcp_client_out pass on $voip_if inet proto udp from any to port $voip_udp_client_out pass out on $wan_if inet proto tcp from any to port $tcp_client_out pass out on $wan_if inet proto udp from any to port $udp_client_out Thank you for helping :) Regards, Paul