Why don't outgoing calls with goip+freepbx work?

Question

Asterisk

Hello!
Can't make an outgoing call via goip+asterisk+freepbx.
When I try to call, on the command line I see:

== Using SIP RTP CoS mark 5
    -- Called SIP/out1/[phone]
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [[email protected]:24] NoOp("SIP/111-00000011", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 1") in new stack
    -- Executing [[email protected]:25] GotoIf("SIP/111-00000011", "0?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [[email protected]:1] Set("SIP/111-00000011", "RC=1") in new stack
    -- Executing [[email protected]:2] Goto("SIP/111-00000011", "1,1") in new stack
    -- Goto (macro-dialout-trunk,1,1)
    -- Executing [[email protected]:1] Goto("SIP/111-00000011", "s-INVALIDNMBR,1") in new stack
    -- Goto (macro-dialout-trunk,s-INVALIDNMBR,1)
    -- Executing [[email protected]:1] NoOp("SIP/111-00000011", "Dial failed due to trunk reporting Address Incomplete - giving up") in new stack
    -- Executing [[email protected]:2] Progress("SIP/111-00000011", "") in new stack
    -- Executing [[email protected]:3] Playback("SIP/111-00000011", "ss-noservice,noanswer") in new stack
    -- <SIP/111-00000011> Playing 'ss-noservice.ulaw' (language 'ru')
    -- Executing [[email protected]:4] Busy("SIP/111-00000011", "20") in new stack
[2017-07-14 12:56:33] WARNING[16184][C-0000000a]: channel.c:5005 ast_prod: Prodding channel 'SIP/111-00000011' failed
  == Spawn extension (macro-dialout-trunk, s-INVALIDNMBR, 4) exited non-zero on 'SIP/111-00000011' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 79876543210, 5) exited non-zero on 'SIP/111-00000011'
    -- Executing [[email protected]:1] Macro("SIP/111-00000011", "hangupcall") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/111-00000011", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [[email protected]:3] ExecIf("SIP/111-00000011", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [[email protected]:4] Hangup("SIP/111-00000011", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/111-00000011' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/111-00000011'

Incoming/outgoing trunk successfully registered, sip show peers output:

localhost*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
1001/1001                 10.0.0.50                                D  No         No             5160     OK (3 ms)
111/111                   10.0.0.109                               D  No         No          A  5062     OK (12 ms)
112/112                   10.0.0.107                               D  No         No          A  60919    OK (2 ms)
out1/1001                 10.0.0.50                                   Yes        Yes            5160     OK (4 ms)
4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline]

An incoming call goes through without any problems.
The dialing settings, as I understand it, are also done correctly.

Though kill, but I don’t understand what to do next.
Everything that is advised on the links in Google, I either already tried to do, or was initially done correctly.

Answer 1

The issue is resolved :))
If anyone is interested, I can share the working options for setting up trunks and outgoing routes, which will allow at least a network of hundreds of goip to be assembled into telephony.

Answer 2

Ivan Eliseev
Good time of day.
Can you share working configs?

Answer 3

Do you really need to pass the "+" sign to your ISP?
I suspect not. And if you just want to be able to dial a number with a "+" sign, then move it in the route to the "prefix" field.