No one is available to answer at this time (1:0/0/0) only on device

Good afternoon, as part of the Asterisk training, I set up the server, I have a DPH-150s device, it is registered under the number 102 on my server

Endpoint:  102_F48CEBF9AD18                                     Not in use    0 of inf
     InAuth:  102_F48CEBF9AD18/102_F48CEBF9AD18
        Aor:  102_F48CEBF9AD18                                   1
      Contact:  102_F48CEBF9AD18/sip:102_F48CEBF9AD18@10.1 859a8a676c Avail         9.637
  Transport:  transport-udp             udp      0      0

Endpoint:  202_18C04D699219                                     Not in use    0 of inf
     InAuth:  202_18C04D699219/202_18C04D699219
        Aor:  202_18C04D699219                                   1
      Contact:  202_18C04D699219/sip:202_18C04D699219@172. 7c5f425fa9 NonQual         nan
  Transport:  transport-udp             udp      0      0

 Endpoint:  203_9C2A705C3623                                     Not in use    0 of inf
     InAuth:  203_9C2A705C3623/203_9C2A705C3623
        Aor:  203_9C2A705C3623                                   1
      Contact:  203_9C2A705C3623/sip:203_9C2A705C3623@10.1 abef757a64 NonQual         nan
  Transport:  transport-udp             udp      0      0

subscribers 202 and 203 are Microsip client, when I make test calls between them everything is OK, but when I try to call 102 I get an error

> Executing [102@subDialUser:3] Dial("PJSIP/203_9C2A705C3623-00000039", "PJSIP/102_F48CEBF9AD18,30") in new stack
>     -- Called PJSIP/102_F48CEBF9AD18
>     -- No one is available to answer at this time (1:0/0/0)
>     -- Executing [102@subDialUser:4] Hangup("PJSIP/203_9C2A705C3623-00000039", "") in new stack

my /etc/asterisk/extensions.conf



exten => _XXX.,1,Verbose(Outcoming call to ${EXTEN})
 same => n,Dial(${Protocol}/${EXTEN}@operator-endpoint,,U(sub-recording,s,1))

exten => _[1-2]XX,1,Verbose(Outcoming call to internal number ${EXTEN})
 same => n,Dial(${Protocol}/${EXTEN})

exten => 100,hint,${Protocol}/${User_100}
exten => 100,1,Verbose(Route to local number 100)
 same => n,Gosub(subDialUser,${EXTEN},1(${Protocol}/${User_100},${mailBoxDefault},${mailBoxContext},${TimeoutDial}))

exten => 102,hint,${Protocol}/${User_102}
exten => 102,1,Verbose(Route to local number 102)
 same => n,Gosub(subDialUser,${EXTEN},1(${Protocol}/${User_102},${mailBoxDefault},${mailBoxContext},${TimeoutDial}))

exten => 201,hint,${Protocol}/${User_201}
exten => 201,1,Verbose(Route to local number 201)
 same => n,Gosub(subDialUser,${EXTEN},1(${Protocol}/${User_201},${mailBoxDefault},${mailBoxContext},${TimeoutDial}))

exten => 202,hint,${Protocol}/${User_202}
exten => 202,1,Verbose(Route to local number 202)
 same => n,Gosub(subDialUser,${EXTEN},1(${Protocol}/${User_202},${mailBoxDefault},${mailBoxContext},${TimeoutDial}))

exten => 203,hint,${Protocol}/${User_203}
exten => 203,1,Verbose(Route to local number 203)
 same => n,Gosub(subDialUser,${EXTEN},1(${Protocol}/${User_203},${mailBoxDefault},${mailBoxContext},${TimeoutDial}))

where to look to find the problem?

In the verbose output resulting from “pjsip set logger on”.

The device has returned a SIP status code that Asterisk interprets as Busy.

Sorry, I thought that was 1:1,0,0. I haven’t seen 1:0:0:0 before, but I guess it means that the call was abandoned or timed out. Unfortunately you seem to have screen scraped, when you should have used the log files. The log files contain time information, and it wouldn’t surprise me if the call ended after exactly 30 seconds.

The subscriber pressed the DND mode on the phone and forgot about it

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