<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8" /></head><body><div>LuKaRo,<br /><br />The "sofia status profile my-sip reg" shows telephones registered to the profile, it doesn't show gateway connections.<br /><br />To see whether the call arrived you could look in the FreeSWITCH log (/var/log/freeswitch/freeswitch.log), there should be entries in there whenever something happens. In addition you could run tcpcump to capture the traffic and check for the incoming INVITE.<br /><br />Martin.<br /><br /><br /><br /><div style="mso-line-height-rule:exactly;-webkit-text-size-adjust:100%;"><table cellpadding="0" cellspacing="0" border="0" style="width:100%;"><tr style="font-size:0;"><td align="left" style="vertical-align:top;"><table cellpadding="0" cellspacing="0" border="0" style="font-size:0;"><tr style="font-size:0;"><td align="left" style="padding:0 0 14px;vertical-align:top;"><table cellpadding="0" cellspacing="0" border="0" style="font-size:0;line-height:normal;"><tr style="font-size:0;"><td align="left" style="vertical-align:top;"><img src="cid:image030223.png@03A9EEC6.163A9555" width="138" height="28" border="0" alt="" style="width:138px;min-width:138px;max-width:138px;height:28px;min-height:28px;max-height:28px;font-size:0;" /></td></tr><tr style="font-size:0;"><td align="left" style="vertical-align:top;"><table cellpadding="0" cellspacing="0" border="0" 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Sans Pro\'',Calibri,Arial;"><span style="font-size:13.33px;">Registered Office: CP House, Otterspool Way, Watford, WD25 8JJ, U.K <br />Registered in England No: 5964349 | VAT Number: GB 902 5369 37</span> <br /></td></tr></table></div>-----Original Message-----<br />From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> On Behalf Of LuKaRo<br />Sent: 14 July 2020 15:31<br />To: freeswitch-users@lists.freeswitch.org<br />Subject: [Freeswitch-users] Registering with external SIP provider<br /><br />Hi everyone,<br /><br />I'm unable to register freeswitch with an external SIP provider. I gave the pages on sofia on the wiki already a good read, but I can't get it to work. One special thing is that my machine has multiple network interfaces and needs to use a specific one to connect to the external SIP provider. Therefore, I copied "external.xml" in "/opt/freeswitch/conf/sip_profiles" to "my-sip.xml" and changed the values for "rtp-ip", "sip-ip", "ext-rtp-ip" and "ext-sip-ip" to the IP address of the interface freeswitch should use, and added the "my-sip" directory to the "<X-PRE-PROCESS cmd="include" header.<br /><br />Then I created "/opt/freeswitch/conf/sip_profiles/my-sip/my-sip.xml" with the following contents:<br /><br /><include><br /> <gateway name="my-telephony"><br /> <param name="proxy" value="<ip address of SIP provider>"/><br /> <param name="username" value="49xxxxxxxxxx"/><br /> <param name="extension" value="49xxxxxxxxxx"/><br /> <param name="password" value="passwordisnotneeded"/><br /> <param name="register" value="true"/><br /> <param name="register-transport" value="tcp"/><br /> <param name="context" value="public"/><br /> </gateway><br /></include><br /><br />"sofia status" on the fs_cli lists the profile as "REGED":<br /><br /> my-sip profile sip:mod_sofia@<local ip address>:5060 RUNNING (0)<br /> my-sip::my-sip gateway sip:49xxxxxxxxxx@<ip address of SIP provider> REGED<br /><br />ss -an|grep <ip address of SIP provider> also lists two established TCP-connections:<br /><br />tcp ESTAB 0 0 <local ip address>:46878 <ip address of SIP provider>:5060 tcp ESTAB 0 0 <local ip address>:5060 <ip address of SIP provider>:25976<br /><br />However, "sofia status profile my-sip reg" shows 0 registrations:<br /><br />Registrations:<br />=================================================================================================<br />Total items returned: 0<br />=================================================================================================<br /><br />And when issuing a call from outside to the extension registered at the SIP provider, I just get a busy signal on the phone. Logs of the external SIP provider report "recipient unavailable".<br /><br />Any ideas what could be wrong or how to further diagnose this issue? How can I check if the call reaches freeswitch?<br /><br />Thanks a lot in advance,<br /><br />LuKaRo<br /><br /></div></body></html>