Hi All,I tried to switch from SIP to PJSIP but I can't make any calls. Asterisk 15.4.0Clients: MicroSIP (based on the pjsip SIP stack) With sip.conf all functions OK (SIP instead of PJSIP in extensions.conf) I converted SIP to PJSIP with the script contrib/scripts/sip_to_pjsip/sip_to_pjsip.py and I removed sip.conf. I can see the clients with:CLI> pjsip show aors After client 61 calls 62 I get just:*CLI> == Setting global variable 'SIPDOMAIN' to '192.168.0.13' (This is Asterisk IP-Address) Call doesn't work! Can somebody tell me please what is wrong?What should I do to use PJSIP instead of SIP? Thank youRegardsMarko sip.conf--------------[general] [61]type=friendcanreinvite=nohost=dynamicsecret=123context=phones [62]type=friendcanreinvite=nohost=dynamicsecret=123context=phones pjsip.conf----------------[transport-udp]type = transportprotocol = udpbind = 0.0.0.0 [61]type = aormax_contacts = 1 [61]type = authusername = 61password = 123 [61]type = endpointcontext = phonesdirect_media = noauth = 61outbound_auth = 61aors = 61 [62]type = aormax_contacts = 1 [62]type = authusername = 62password = 123 [62]type = endpointcontext = phonesdirect_media = noauth = 62outbound_auth = 62aors = 62 extensions.conf---------------------------[general]autofallthrough=yes [default] [phones] exten => _.,1,Dial(PJSIP/${EXTEN},30) exten => _.,n,Hangup() *CLI> pjsip show aors Aor: <Aor..............................................> <MaxContact> Contact: <Aor/ContactUri............................> <Hash....> <Status> <RTT(ms)..>========================================================================================== Aor: 61 1 Contact: 61/sip:61 at 192.168.0.11:55238;ob af939754af Unknown nan Aor: 62 1 Contact: 62/sip:62 at 192.168.0.22:63508;rinstance=526f9 4bddb5801c Unknown nan *CLI> pjsip show endpoints Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.> I/OAuth: <AuthId/UserName...........................................................> Aor: <Aor............................................> <MaxContact> Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.........................................................> Match: <criteria.........................> Channel: <ChannelId......................................> <State.....> <Time.....> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>========================================================================================== Endpoint: 61 Not in use 0 of inf OutAuth: 61/61 InAuth: 61/61 Aor: 61 1 Contact: 61/sip:61 at 192.168.0.11:55238;ob af939754af Unknown nan Endpoint: 62 Not in use 0 of inf OutAuth: 62/62 InAuth: 62/62 Aor: 62 1 Contact: 62/sip:62 at 192.168.0.42:58658;rinstance=7bc 36def1b497 Unknown nan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180606/fa83c19e/attachment.html>