Hello, I’ve connected freeswitch with an asterisk server via sip trunk with the following configuration in my test environment: Freeswitch side: <gateway name="sip.testprovider.com"> <param name="username" value="MyCustomSipTrunk1"/> <param name="password" value="easy123"/> <param name="extension" value="trunk"/> <param name="register" value="true"/> <param name="from_domain" value="sip.testprovider.com"/> </gateway> Asterisk side: sip.conf [MyCustomSipTrunk1] type=peer callerid="MyCustomSipTrunk1" <MyCustomSipTrunk1> host=dynamic nat=no username=MyCustomSipTrunk1 fromdomain=sip.testprovider.com directmedia=no disallow=all allow=gsm allow=ulaw allow=alaw secret=easy123 context=kamailio extensions.conf [kamailio] exten => _X.,1,Dial(SIP/${EXTEN}@MyCustomSipTrunk1,60,tr) If I send a call from asterisk to freeswitch, I can see the following in the log: 2015-09-02 00:06:28.877729 [DEBUG] sofia.c:9001 IP 192.168.1.202 Rejected by acl "domains". Falling back to Digest auth. 2015-09-02 00:06:28.877729 [WARNING] sofia_reg.c:2827 Can't find user [MyCustomSipTrunk1@sip.testprovider.com] from 192.168.1.202 You must define a domain called 'sip.testprovider.com' in your directory and add a user with the id="MyCustomSipTrunk1" attribute and you must configure your device to use the proper domain in it's authentication credentials. If I create the user in the directory for the domain, it works - but why do I need to create this user however the gateway is already already registered and authenticated with the asterisk server? Thanks Markus