Building on a stackoverflow question, how can I actually get the device to ring?thufir@mordor:~$ sudo sipsak -vv -s sip:6003@localhostNo SRV record: _sip._tcp.localhostNo SRV record: _sip._udp.localhostusing A record: localhostmessage received:SIP/2.0 200 OKVia: SIP/2.0/UDP 127.0.1.1:60831;branch=z9hG4bK.19ff6b4d;alias;received=127.0.0.1;rport=60831From: sip:sipsak@127.0.1.1:60831;tag=39c26336To: sip:6003@localhost;tag=as5c9393b3Call-ID: 969040694@127.0.1.1CSeq: 1 OPTIONSServer: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replacesContact: <sip:127.0.0.1:5060>Accept: application/sdpContent-Length: 0** reply received after 0.152 ms **   SIP/2.0 200 OK   final receivedAsterisk CLI output:mordor*CLI> mordor*CLI> sip show peersName/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description                      demo_alice/demo_alice     192.168.1.6                              D  Yes        Yes            5060     Unmonitored                                  demo_bob/demo_bob         192.168.1.8                              D  Yes        Yes            40962    Unmonitored                                  thufir/thufir             192.168.1.5                              D  Yes        Yes            5062     Unmonitored                                  3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]mordor*CLI> mordor*CLI> dialplan show internal[ Context 'internal' created by 'pbx_config' ]  '6001' =>         1. Dial(SIP/demo_alice,20)                    [pbx_config]  '6002' =>         1. Dial(SIP/demo_bob,20)                      [pbx_config]  '6003' =>         1. Dial(SIP/thufir,20)                        [pbx_config]-= 3 extensions (3 priorities) in 1 context. =-mordor*CLI> Nothing shows in the Asterisk CLI.  However, if I dial extension 6003 from an Android tablet:mordor*CLI>   == Using SIP RTP CoS mark 5    -- Executing [6003@internal:1] Dial(SIP/demo_alice-00000004, SIP/thufir,20) in new stack  == Using SIP RTP CoS mark 5    -- Called SIP/thufir    -- SIP/thufir-00000005 is ringing    -- SIP/thufir-00000005 answered SIP/demo_alice-00000004    -- Channel SIP/demo_alice-00000004 joined 'simple_bridge' basic-bridge <d5d9838f-a838-4a80-b3f9-15391fb4f2ab>    -- Channel SIP/thufir-00000005 joined 'simple_bridge' basic-bridge <d5d9838f-a838-4a80-b3f9-15391fb4f2ab>    -- Channel SIP/thufir-00000005 left 'native_rtp' basic-bridge <d5d9838f-a838-4a80-b3f9-15391fb4f2ab>    -- Channel SIP/demo_alice-00000004 left 'native_rtp' basic-bridge <d5d9838f-a838-4a80-b3f9-15391fb4f2ab>  == Spawn extension (internal, 6003, 1) exited non-zero on 'SIP/demo_alice-00000004'mordor*CLI> then demo_alice can dial thufir, the phone literally rings, and I can hang it up.To what extent can sipsak simulate a call?
How can sipsak simulate a call with usrloc?
command line;asterisk;voip;sip;telephony