 Звонок с Asterisk (номер 4006 и Ip 10.48.100.100)  на smg номер 3000 и ip 10.48.48.110 (неудачный вызов)

 IP-ATC*CLI> 
  == Using SIP VIDEO TOS bits 136
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [3000@from-internal:1] Macro("SIP/4006-00000311", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/4006-00000311", "AMPUSER=4006") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/4006-00000311", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/4006-00000311", "1?Set(REALCALLERIDNUM=4006)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/4006-00000311", "AMPUSER=4006") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/4006-00000311", "AMPUSERCIDNAME=asteriskTest") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/4006-00000311", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/4006-00000311", "AMPUSERCID=4006") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/4006-00000311", "CALLERID(all)="asteriskTest" <4006>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/4006-00000311", "1?Set(CHANNEL(language)=ru)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/4006-00000311", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/4006-00000311", "CALLERID(number)=4006") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/4006-00000311", "CALLERID(name)=asteriskTest") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/4006-00000311", "Using CallerID "asteriskTest" <4006>") in new stack
    -- Executing [3000@from-internal:2] NoOp("SIP/4006-00000311", "Calling Out Route: smg") in new stack
    -- Executing [3000@from-internal:3] Set("SIP/4006-00000311", "INTRACOMPANYROUTE=YES") in new stack
    -- Executing [3000@from-internal:4] Set("SIP/4006-00000311", "MOHCLASS=default") in new stack
    -- Executing [3000@from-internal:5] Set("SIP/4006-00000311", "_NODEST=") in new stack
    -- Executing [3000@from-internal:6] Macro("SIP/4006-00000311", "record-enable,4006,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/4006-00000311", "1?check") in new stack
    -- Goto (macro-record-enable,s,5)
    -- Executing [s@macro-record-enable:5] ExecIf("SIP/4006-00000311", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:6] GotoIf("SIP/4006-00000311", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,16)
    -- Executing [s@macro-record-enable:16] GotoIf("SIP/4006-00000311", "0?IN") in new stack
    -- Executing [s@macro-record-enable:17] ExecIf("SIP/4006-00000311", "1?MacroExit()") in new stack
    -- Executing [3000@from-internal:7] Macro("SIP/4006-00000311", "dialout-trunk,3,3000,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/4006-00000311", "DIAL_TRUNK=3") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/4006-00000311", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/4006-00000311", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/4006-00000311", "DIAL_NUMBER=3000") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/4006-00000311", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/4006-00000311", "OUTBOUND_GROUP=OUT_3") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/4006-00000311", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/4006-00000311", "1?skipoutcid") in new stack
    -- Goto (macro-dialout-trunk,s,12)
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/4006-00000311", "1?sub-flp-3,s,1") in new stack
    -- Executing [s@sub-flp-3:1] ExecIf("SIP/4006-00000311", "1?Return()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/4006-00000311", "OUTNUM=3000") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/4006-00000311", "custom=SIP/test_smg") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/4006-00000311", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)tr)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/4006-00000311", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/4006-00000311", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/4006-00000311", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/4006-00000311", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/4006-00000311", "SIP/test_smg/3000,300,tr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 13136
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.48.48.110:5060:
INVITE sip:3000@10.48.48.110 SIP/2.0
Via: SIP/2.0/UDP 10.48.100.100:5060;branch=z9hG4bK041c9e30;rport
Max-Forwards: 70
From: "asteriskTest" <sip:4006@10.48.100.100>;tag=as44bbba10
To: <sip:3000@10.48.48.110>
Contact: <sip:4006@10.48.100.100:5060>
Call-ID: 33b0f22d3f8ac04436beac8718f7924f@10.48.100.100:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Tue, 20 May 2014 05:23:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 1594832306 1594832306 IN IP4 10.48.100.100
s=Asterisk PBX 1.8.20.0
c=IN IP4 10.48.100.100
t=0 0
m=audio 13136 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/test_smg/3000

<--- SIP read from UDP:10.48.48.110:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.48.100.100:5060;branch=z9hG4bK041c9e30;received=10.48.100.100;rport=5060
From: "asteriskTest" <sip:4006@10.48.100.100>;tag=as44bbba10
To: <sip:3000@10.48.48.110>
Call-ID: 33b0f22d3f8ac04436beac8718f7924f@10.48.100.100:5060
CSeq: 102 INVITE
Contact: <sip:3000@10.48.48.110:5060>
P-Eltex-Info: {trunk,2} 786 <0.8687.0>
User-Agent: Eltex smg_pa_sip 2.15.1.32
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:10.48.48.110:5060 --->
INVITE sip:3000@10.48.100.100:5060 SIP/2.0
Via: SIP/2.0/UDP 10.48.48.110:5060;rport;branch=z9hG4bK-o258254441277969K198417
From: "asteriskTest" <sip:4006@10.48.48.110;user=phone>;tag=63478564522
To: <sip:3000@10.48.100.100;user=phone>
Call-ID: 1400-564522-64908
CSeq: 63467 INVITE
User-Agent: Eltex smg_pa_sip 2.15.1.32
Max-Forwards: 70
Contact: <sip:4006@10.48.48.110:5060>
Accept: multipart/mixed, application/sdp
Allow: INVITE, ACK, BYE, CANCEL, PRACK, REGISTER, INFO, REFER, NOTIFY, OPTIONS, UPDATE
Supported: timer, 100rel, replaces
Min-SE: 90
Session-Expires: 1800;refresher=uac
Category: 10
P-Eltex-Info: {trunk,2} 1207 <0.8689.0>
Content-Type: application/sdp
Content-Length: 183

v=0
o=- 1207 1207 IN IP4 10.48.48.110
s=SMG SIP session
c=IN IP4 10.48.48.110
t=0 0
m=audio 2104 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:30
a=sendrecv
<------------->
--- (18 headers 10 lines) ---
Sending to 10.48.48.110:5060 (NAT)
Using INVITE request as basis request - 1400-564522-64908
Found peer '4006' for '4006' from 10.48.48.110:5060

<--- Reliably Transmitting (NAT) to 10.48.48.110:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.48.48.110:5060;branch=z9hG4bK-o258254441277969K198417;received=10.48.48.110;rport=5060
From: "asteriskTest" <sip:4006@10.48.48.110;user=phone>;tag=63478564522
To: <sip:3000@10.48.100.100;user=phone>;tag=as53696c41
Call-ID: 1400-564522-64908
CSeq: 63467 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="517f9a57"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1400-564522-64908' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.48.48.110:5060 --->
ACK sip:3000@10.48.100.100:5060 SIP/2.0
Via: SIP/2.0/UDP 10.48.48.110:5060;rport;branch=z9hG4bK-o258254441277969K198417
From: "asteriskTest" <sip:4006@10.48.48.110;user=phone>;tag=63478564522
To: <sip:3000@10.48.100.100;user=phone>;tag=as53696c41
Call-ID: 1400-564522-64908
CSeq: 63467 ACK
User-Agent: Eltex smg_pa_sip 2.15.1.32
Max-Forwards: 70
Contact: <sip:4006@10.48.48.110:5060>
P-Eltex-Info: {trunk,2} 1207 <0.8689.0>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:10.48.48.110:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.48.100.100:5060;branch=z9hG4bK041c9e30;received=10.48.100.100;rport=5060
From: "asteriskTest" <sip:4006@10.48.100.100>;tag=as44bbba10
To: <sip:3000@10.48.48.110>;tag=53073564522
Call-ID: 33b0f22d3f8ac04436beac8718f7924f@10.48.100.100:5060
CSeq: 102 INVITE
P-Eltex-Info: REL from ss7 layer, ss7 cause: {isup,<<128,149>>}
P-Eltex-Info: {trunk,2} 786 <0.8687.0>
Contact: <sip:3000@10.48.48.110:5060>
User-Agent: Eltex smg_pa_sip 2.15.1.32
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 10.48.48.110:5060:
ACK sip:3000@10.48.48.110 SIP/2.0
Via: SIP/2.0/UDP 10.48.100.100:5060;branch=z9hG4bK041c9e30;rport
Max-Forwards: 70
From: "asteriskTest" <sip:4006@10.48.100.100>;tag=as44bbba10
To: <sip:3000@10.48.48.110>;tag=53073564522
Contact: <sip:4006@10.48.100.100:5060>
Call-ID: 33b0f22d3f8ac04436beac8718f7924f@10.48.100.100:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0


---
Scheduling destruction of SIP dialog '33b0f22d3f8ac04436beac8718f7924f@10.48.100.100:5060' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/4006-00000311", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/4006-00000311", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/4006-00000311", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/4006-00000311", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("SIP/4006-00000311", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/4006-00000311", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/4006-00000311", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/4006-00000311", "CALLERID(number)=4006") in new stack
    -- Executing [3000@from-internal:8] Macro("SIP/4006-00000311", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/4006-00000311", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/4006-00000311", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/4006-00000311", "1?intracompany,1") in new stack
    -- Goto (macro-outisbusy,intracompany,1)
    -- Executing [intracompany@macro-outisbusy:1] Playback("SIP/4006-00000311", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/4006-00000311> Playing 'all-circuits-busy-now.slin' (language 'ru')
    -- <SIP/4006-00000311> Playing 'pls-try-call-later.slin' (language 'ru')
    -- Executing [intracompany@macro-outisbusy:2] Congestion("SIP/4006-00000311", "20") in new stack
  == Spawn extension (macro-outisbusy, intracompany, 2) exited non-zero on 'SIP/4006-00000311' in macro 'outisbusy'
  == Spawn extension (from-internal, 3000, 8) exited non-zero on 'SIP/4006-00000311'
    -- Executing [h@from-internal:1] Macro("SIP/4006-00000311", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/4006-00000311", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] NoOp("SIP/4006-00000311", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:11] GotoIf("SIP/4006-00000311", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,29)
    -- Executing [s@macro-hangupcall:29] NoOp("SIP/4006-00000311", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:30] GotoIf("SIP/4006-00000311", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,35)
    -- Executing [s@macro-hangupcall:35] NoOp("SIP/4006-00000311", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:36] GotoIf("SIP/4006-00000311", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,42)
    -- Executing [s@macro-hangupcall:42] NoOp("SIP/4006-00000311", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:43] GotoIf("SIP/4006-00000311", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,46)
    -- Executing [s@macro-hangupcall:46] GotoIf("SIP/4006-00000311", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,49)
    -- Executing [s@macro-hangupcall:49] GotoIf("SIP/4006-00000311", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,51)
    -- Executing [s@macro-hangupcall:51] AGI("SIP/4006-00000311", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/4006-00000311>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:52] Hangup("SIP/4006-00000311", "") in new stack
  == Spawn extension (macro-hangupcall, s, 52) exited non-zero on 'SIP/4006-00000311' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4006-00000311'
Really destroying SIP dialog '1400-564522-64908' Method: ACK
