How to set up calls between asterisks?

Question

Asterisk

Merged 2 asteriska trunk iax2, the trunk rose, the status is OK. I set up outgoing call routing, dialing from one station 2XX, to another 1XX. During the call, they answer that "All lines are busy", I post the log from the call The output in the asterisk -vvvr command ... I don’t understand what he swears at.
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-- Executing [[email protected]:1] MacroExit("SIP/101-00000014", "" ) in new stack
-- Executing [[email protected]:18] GotoIf ("SIP/101-00000014", "0?bypass,1"
-- Executing [[email protected]:19] ExecIf("SIP/101-00000014", "1?Set(CONNECTEDL INE(num,i)=201)") in new stack
-- Executing [[email protected] macro-dialout-trunk:20] ExecIf("SIP/101-00000014", "1?Set(CONNECTEDL INE(name,i)=CID:101)") in new stack
-- Executing [[email protected] trunk:21] GotoIf("SIP/101-00000014", "0?customtrunk") in new stack
-- Executing [[email protected]:22] Dial("SIP/101-00000014", "IAX2/VLG/201,300,T tr") in new stack
-- Called IAX2/VLG/201
-- Hungup ' IAX2/VLG-16787'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [[email protected]:23] NoOp("SIP/101-00000014", " Dial failed for so me reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 50") in new stack
-- Executing [[email protected]:24] GotoIf("SIP/101-00000014", "0?continue,1:sC HANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk, s-CHANUNAVAIL,1)
-- Executing [[email protected]:1] Set("SIP/101-00000014", "RC=50") in new stack
-- Executing [[email protected] -dialout-trunk:2] Goto("SIP/101-00000014", "50,1") in new stack
-- Goto (macro-dialout-trunk,50,1)
-- Executing [[email protected]:1] Goto("SIP/101-00000014", "continue,1") in ne w stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [[email protected]:1] NoOp("SIP/101-00000014", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 50 - failing through to other trunks") in new stac k
-- Executing [[email protected]:2] Set("SIP/101-00000014", "CALLERID(numb er)=101") in new stack
-- Executing [[email protected]:8] Macro ("SIP/101-00000014", "outisbusy,") in new st ack
-- Executing [[email protected]:1] Progress("SIP/101-00000014", "") in new stack
-- Executing [ [email protected]:2] GotoIf("SIP/101-00000014", "0?emergency,1") in ne w stack
-- Executing [[email protected]:3] GotoIf("SIP/101-00000014", "1?intracompany,1") in new stack
-- Goto (macro-outisbusy,intracompany,1)
-- Executing [intracompany @macro-outisbusy:1] Playback("SIP/101-00000014", "all-circ uits-busy-now&pls-try-call-later, noanswer") in new stack
-- Playing 'all-circuits-busy-now .slin' (language 'ru')
-- Playing 'pls-try-call-later.slin' (language 'ru')
-- Executing [[email protected]:2] Congestion("SIP/101-00000014", "20") in new stack
[2016-01-28 16:50:01] WARNING[7548][C-00000013] : channel.c:4860 ast_prod: Prodding chann el 'SIP/101-00000014' failed
== Spawn extension (macro-outisbusy, intracompany, 2) exited non-zero on 'SIP/101-0000 0014' in macro 'outisbusy'
== Spawn extension (from-internal, 201, 8) exited non-zero on 'SIP/101-00000014'
-- Executing [[email protected]:1] Hangup("SIP/101-00000014", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101 -00000014'

Answer 1

Answer 2

-- Hungup 'IAX2/VLG-16787'
== Everyone is busy/congested at this time (1:0/0/1)

Asterisk on the second side gave out that all lines are busy. Show his log.

Answer 3

Lay out settings of trunks on both servers. do you have freepbx on both servers?