Registering with external SIP provider
Hi everyone, 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. Then I created "/opt/freeswitch/conf/sip_profiles/my-sip/my-sip.xml" with the following contents: <include> <gateway name="my-telephony"> <param name="proxy" value="<ip address of SIP provider>"/> <param name="username" value="49xxxxxxxxxx"/> <param name="extension" value="49xxxxxxxxxx"/> <param name="password" value="passwordisnotneeded"/> <param name="register" value="true"/> <param name="register-transport" value="tcp"/> <param name="context" value="public"/> </gateway> </include> "sofia status" on the fs_cli lists the profile as "REGED": my-sip profile sip:mod_sofia@<local ip address>:5060 RUNNING (0) my-sip::my-sip gateway sip:49xxxxxxxxxx@<ip address of SIP provider> REGED ss -an|grep <ip address of SIP provider> also lists two established TCP-connections: 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 However, "sofia status profile my-sip reg" shows 0 registrations: Registrations: ================================================================================================= Total items returned: 0 ================================================================================================= 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". Any ideas what could be wrong or how to further diagnose this issue? How can I check if the call reaches freeswitch? Thanks a lot in advance, LuKaRo
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LuKaRo