I had been having trouble getting my SIP connections to persist for more than a few seconds following a restart. I was pursuing this in another call (freePBX trunk to inbound route works - for about 20 seconds only).
I thought I’d attempt a real basic “Hello World” type install / config in order to isolate the problem.
So… I have a windows host with a Linux guest, running Asterisk.
All starts ok.
Host dnsmgr Username Refresh State Reg.Time
sip.voipfone.net:5060 Y hidden 45 Registered Sun, 31 Mar 2019 22:55:32
1 SIP registrations.
which seems ok.
All I want to do is see Asterisk / PBX pickup a call and do something with it. I don’t care what. I thought the simplest of “Hello World” tests would be to do a Text to Speech announcement or maybe forward to a misc destination.
I can’t get Asterisk to pick up anything.
18.104.22.168:5060 192.168.86.46:5160│Via: SIP/2.0/UDP 22.214.171.124:5060;branch=z9hG4bK742fbcb1;received=126.96.36.199;rport=5060
──────────┬───────── ──────────┬─────────│From: “07951408???” sip:firstname.lastname@example.org;tag=VF5e4104d5faa3ffeb1cdb7c929e8f
│ INVITE (SDP) │ │To: sip:email@example.com.???:5160;tag=as31f9e7cc
22:53:21.978033 │ ──────────────────────────> │ │Call-ID: VFb6aad53abf56d550f3c14b7a870aa2@voipfone
+0.026518 │ 403 Forbidden │ │CSeq: 102 INVITE
22:53:22.004551 │ <────────────────────────── │ │Server: FPBX-188.8.131.52(13.22.0)
+0.020813 │ ACK │ │Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
22:53:22.025364 │ ──────────────────────────> │ │Supported: replaces, timer
+0.003228 │ CANCEL │ │Content-Length: 0
22:53:22.028592 │ ──────────────────────────> │ │
+0.000486 │ 481 Call leg/transaction d │ │
22:53:22.029078 │ <────────────────────────── │ │
So the connection is being “forbidden”. I thought I’d hide some IP/account/password info with “???” just in case! Is this paranoia!
I’m hoping as this is a real basic install that I’ve done nothing to mess with, we can solve this one quickly so I might proceed (cautiously) building on what’s gone before. Unfortunately I’m falling at the first hurdle!
I really thought I could just specify a trunk, specify the SIP settings and then handle the Inbound route accordingly.
External calls to the DID receive an immediate not here -> voicemail message.
REGISTER and NOTIFY calls seem to be made ok - it’s the INVITE which is “forbidden”.
Please help! Between the other call and this I’ve spent over a week on this and not really got anywhere.
VOIP services are from voipfone (sip.voipfone.net)