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Астериск + Acme Packet SBC 3820

Вопросы по использованию и настройке IP телефонов, шлюзов и всего прочего

Модераторы: april22, Zavr2008

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ded
Сообщения: 15628
Зарегистрирован: 26 авг 2010, 19:00

Re: Астериск + Acme Packet SBC 3820

Сообщение ded »

1xx responses are informational responses and in the case of 100 Trying are optional. SIP User Agent Servers (UAS) will generally respond with a 100 Trying response immediately when they receive an INVITE request to let the User Agent Client (UAC) know they are processing the request and to avoid retransmits. At some time later they will follow the 100 Trying response with a 180 Ringing or 183 Session Progress. Once someone or something answers the call a 2xx response needs to be sent, typically 200 Ok.

If your softphone software is only generating a 100 Trying response and not the subsequent 180 Ringing response then my guess is you have missed a step. At the very least if the softphone has a problem and can't generate a ringing response because there is nothing to ring it should generate a 4xx error response.
Всем привет.

Есть схема: Asterisk 1.6.2.17.3 <-SIP-> Iskratel Si2000v5 <-SS7/ISUP-> TDM

Со стороны TDM приходит вызов - нет КПВ. Причина - в ответ на входящий IAM сишка шлет как полагается, инвайт астериску, астериск отвечает 100 Trying, 180 Ringing и 183 Session Progress, в результате Si2000 пересылает в ISUP ACM Alerting, и далее CPG Progress, что в корне неправильно, ибо Alerting означает что вызываемый абонент найден и вызывается. Естетственно что TDM АТС, получая Progress после Alerting - затыкается.

Но тут, так как вызов транзитный, то астериск должен был ответить только с 183 Session Progress c SDP, чтобы станция послала в сторону TDM ACM c Progress Indicator=Inband early media, что подразумевает собой проключение тракта в предответном и начало передачи раннего аудио (КПВ, музычка, или голосовое сообщение).

Как заставить Астериск отвечать на инвайт только 183 Progress без 180 ?
Решение:

в sip.conf

prematuremedia = no
progressinband = never

далее в диалплане в любом месте перед вызовом, хоть перед Dial:

[context]
exten => _X.,1,Progress
exten => _X.,n,***********
http://asterisk-support.ru/question/156 ... iushchego/
Вернуться к началу
Internetchik
Сообщения: 107
Зарегистрирован: 10 июл 2012, 10:38
Откуда: Алматы

Re: Астериск + Acme Packet SBC 3820

Сообщение Internetchik »

Прошу прощения за долгое отсутствие.
Итак. Полные Вербоус + Дебаг:
Эластикс:

Код: Выделить всё

 == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [+77019650957@from-internal:1] Macro("SIP/1001-00000007", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/1001-00000007", "AMPUSER=1001") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1001-00000007", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1001-00000007", "1?Set(REALCALLERIDNUM=1001)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/1001-00000007", "AMPUSER=1001") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1001-00000007", "AMPUSERCIDNAME=Linux") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1001-00000007", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1001-00000007", "AMPUSERCID=1001") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/1001-00000007", "CALLERID(all)="Linux" <1001>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1001-00000007", "1?Set(CHANNEL(language)=ru)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1001-00000007", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/1001-00000007", "CALLERID(number)=1001") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/1001-00000007", "CALLERID(name)=Linux") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/1001-00000007", "Using CallerID "Linux" <1001>") in new stack
    -- Executing [+77019650957@from-internal:2] NoOp("SIP/1001-00000007", "Calling Out Route: Kcell") in new stack
    -- Executing [+77019650957@from-internal:3] Set("SIP/1001-00000007", "MOHCLASS=default") in new stack
    -- Executing [+77019650957@from-internal:4] Set("SIP/1001-00000007", "_NODEST=") in new stack
    -- Executing [+77019650957@from-internal:5] Macro("SIP/1001-00000007", "record-enable,1001,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1001-00000007", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/1001-00000007", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/1001-00000007", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/1001-00000007", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/1001-00000007", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:17] NoOp("SIP/1001-00000007", "Recording enable for 1001") in new stack
    -- Executing [s@macro-record-enable:18] Set("SIP/1001-00000007", "CALLFILENAME=OUT1001-20130530-122432-1369895072.7") in new stack
    -- Executing [s@macro-record-enable:19] Goto("SIP/1001-00000007", "record") in new stack
    -- Goto (macro-record-enable,s,23)
    -- Executing [s@macro-record-enable:23] MixMonitor("SIP/1001-00000007", "OUT1001-20130530-122432-1369895072.7.wav,,") in new stack
    -- Executing [s@macro-record-enable:24] Set("SIP/1001-00000007", "CDR(userfield)=audio:OUT1001-20130530-122432-1369895072.7.wav") in new stack
    -- Executing [s@macro-record-enable:25] MacroExit("SIP/1001-00000007", "") in new stack
    -- Executing [+77019650957@from-internal:6] Macro("SIP/1001-00000007", "dialout-trunk,4,+77019650957,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1001-00000007", "DIAL_TRUNK=4") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1001-00000007", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1001-00000007", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1001-00000007", "DIAL_NUMBER=+77019650957") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1001-00000007", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1001-00000007", "OUTBOUND_GROUP=OUT_4") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1001-00000007", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1001-00000007", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1001-00000007", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1001-00000007", "outbound-callerid,4") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1001-00000007", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1001-00000007", "0?Set(REALCALLERIDNUM=1001)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1001-00000007", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/1001-00000007", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/1001-00000007", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/1001-00000007", "TRUNKOUTCID=+77787460019") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1001-00000007", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1001-00000007", "1?Set(CALLERID(all)=+77787460019)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1001-00000007", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1001-00000007", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1001-00000007", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/1001-00000007", "0?sub-flp-4,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1001-00000007", "OUTNUM=+77019650957") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1001-00000007", "custom=SIP/Kcell") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1001-00000007", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1001-00000007", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1001-00000007", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1001-00000007", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1001-00000007", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1001-00000007", "SIP/Kcell/+77019650957,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 14314
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 11.11.11.11:5060:
INVITE sip:+77019650957@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK0172a5e4;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as29485211
To: <sip:+77019650957@11.11.11.11>
Contact: <sip:+77787460019@22.22.22.22:5060>
Call-ID: 0a4960ee61ba21a943e8c0ba027b3fdc@22.22.22.22:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.21.0)
Date: Thu, 30 May 2013 06:24:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 286

v=0
o=root 1531113489 1531113489 IN IP4 22.22.22.22
s=Asterisk PBX 1.8.21.0
c=IN IP4 22.22.22.22
t=0 0
m=audio 14314 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/Kcell/+77019650957

<--- SIP read from UDP:11.11.11.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0172a5e4;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as29485211
To: <sip:+77019650957@11.11.11.11>
Call-ID: 0a4960ee61ba21a943e8c0ba027b3fdc@22.22.22.22:5060
CSeq: 102 INVITE

<------------->
--- (6 headers 0 lines) ---
  == Begin MixMonitor Recording SIP/1001-00000007

<--- SIP read from UDP:11.11.11.11:5060 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0172a5e4;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as29485211
To: <sip:+77019650957@11.11.11.11>;tag=SDarpu199-1866102510
all-ID: 0a4960ee61ba21a943e8c0ba027b3fdc@22.22.22.22:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
    -- Got SIP response 400 "Bad Request" back from 11.11.11.11:5060
Transmitting (NAT) to 11.11.11.11:5060:
ACK sip:+77019650957@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK0172a5e4;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as29485211
To: <sip:+77019650957@11.11.11.11>;tag=SDarpu199-1866102510
Contact: <sip:+77787460019@22.22.22.22:5060>
Call-ID: 0a4960ee61ba21a943e8c0ba027b3fdc@22.22.22.22:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.21.0)
Content-Length: 0


---
    -- SIP/Kcell-00000008 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/1001-00000007", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 127") in new stack
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/1001-00000007", "s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/1001-00000007", "RC=127") in new stack
    -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/1001-00000007", "127,1") in new stack
    -- Goto (macro-dialout-trunk,127,1)
    -- Executing [127@macro-dialout-trunk:1] Goto("SIP/1001-00000007", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1001-00000007", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,continue,3)
    -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1001-00000007", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 127 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/1001-00000007", "CALLERID(number)=1001") in new stack
    -- Executing [+77019650957@from-internal:7] Macro("SIP/1001-00000007", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/1001-00000007", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/1001-00000007", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/1001-00000007", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/1001-00000007", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/1001-00000007> Playing 'all-circuits-busy-now.alaw' (language 'ru')
Really destroying SIP dialog '0a4960ee61ba21a943e8c0ba027b3fdc@22.22.22.22:5060' Method: INVITE
    -- <SIP/1001-00000007> Playing 'pls-try-call-later.alaw' (language 'ru')
    -- Executing [s@macro-outisbusy:5] Congestion("SIP/1001-00000007", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/1001-00000007' in macro 'outisbusy'
  == Spawn extension (from-internal, +77019650957, 7) exited non-zero on 'SIP/1001-00000007'
    -- Executing [h@from-internal:1] Macro("SIP/1001-00000007", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1001-00000007", "0?endmixmoncheck") in new stack
    -- Executing [s@macro-hangupcall:2] Set("SIP/1001-00000007", "MIXMON_CALLFILENAME=/var/spool/asterisk/monitor/OUT1001-20130530-122432-1369895072.7.wav") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1001-00000007", "1?defaultmixmondir") in new stack
    -- Goto (macro-hangupcall,s,5)
    -- Executing [s@macro-hangupcall:5] System("SIP/1001-00000007", "test -e /var/spool/asterisk/monitor/OUT1001-20130530-122432-1369895072.7.wav") in new stack
    -- Executing [s@macro-hangupcall:6] NoOp("SIP/1001-00000007", "SYSTEMSTATUS = SUCCESS") in new stack
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1001-00000007", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/1001-00000007", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1001-00000007", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/1001-00000007", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/1001-00000007", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/1001-00000007", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/1001-00000007", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/1001-00000007", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/1001-00000007", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/1001-00000007", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/1001-00000007", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/1001-00000007", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/1001-00000007>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/1001-00000007", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/1001-00000007' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-00000007'
  == MixMonitor close filestream
  == End MixMonitor Recording SIP/1001-00000007
Reliably Transmitting (NAT) to 11.11.11.11:5060:
OPTIONS sip:11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK21d3ffb7;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@22.22.22.22>;tag=as4c6503c3
To: <sip:11.11.11.11>
Contact: <sip:Unknown@22.22.22.22:5060>
Call-ID: 14d1d15309bfe63363fa3990624c21dd@22.22.22.22:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.21.0)
Date: Thu, 30 May 2013 06:24:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:11.11.11.11:5060 --->
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK21d3ffb7;rport=5060
From: "Unknown" <sip:Unknown@22.22.22.22>;tag=as4c6503c3
To: <sip:11.11.11.11>;tag=aprqngfrt-mcq6lv20000c6
Call-ID: 14d1d15309bfe63363fa3990624c21dd@22.22.22.22:5060
CSeq: 102 OPTIONS

<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '14d1d15309bfe63363fa3990624c21dd@22.22.22.22:5060' Method: OPTIONS
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
Триксбокс:

Код: Выделить всё

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
    -- Executing [+77019650957@from-internal:1] Macro("SIP/1101-00000000", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/1101-00000000", "AMPUSER=1101") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1101-00000000", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1101-00000000", "1?Set(REALCALLERIDNUM=1101)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/1101-00000000", "AMPUSER=1101") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/1101-00000000", "AMPUSERCIDNAME=1101") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1101-00000000", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1101-00000000", "AMPUSERCID=1101") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/1101-00000000", "CALLERID(all)="1101" <1101>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1101-00000000", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1101-00000000", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/1101-00000000", "Using CallerID "1101" <1101>") in new stack
    -- Executing [+77019650957@from-internal:2] Set("SIP/1101-00000000", "_NODEST=") in new stack
    -- Executing [+77019650957@from-internal:3] Macro("SIP/1101-00000000", "record-enable,1101,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1101-00000000", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/1101-00000000", "recordingcheck,20130530-170516,1369911916.0") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck,20130530-170516,1369911916.0: Outbound recording enabled.
 recordingcheck,20130530-170516,1369911916.0: CALLFILENAME=OUT1101-20130530-170516-1369911916.0
    -- <SIP/1101-00000000>AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:999] MixMonitor("SIP/1101-00000000", "OUT1101-20130530-170516-1369911916.0.wav,,") in new stack
  == Begin MixMonitor Recording SIP/1101-00000000
    -- Executing [+77019650957@from-internal:4] Macro("SIP/1101-00000000", "dialout-trunk,2,+77019650957,,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1101-00000000", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1101-00000000", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1101-00000000", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1101-00000000", "DIAL_NUMBER=+77019650957") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1101-00000000", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1101-00000000", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1101-00000000", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1101-00000000", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1101-00000000", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1101-00000000", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1101-00000000", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1101-00000000", "0?Set(REALCALLERIDNUM=1101)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1101-00000000", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/1101-00000000", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/1101-00000000", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/1101-00000000", "TRUNKOUTCID=+77787460019") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1101-00000000", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1101-00000000", "1?Set(CALLERID(all)=+77787460019)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1101-00000000", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1101-00000000", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1101-00000000", "0?AGI(fixlocalprefix)") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1101-00000000", "OUTNUM=+77019650957") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1101-00000000", "custom=SIP/KCELL") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1101-00000000", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1101-00000000", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1101-00000000", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1101-00000000", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1101-00000000", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1101-00000000", "SIP/KCELL/+77019650957,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 6
Audio is at 22.22.22.22 port 13680
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 11.11.11.11:5060:
INVITE sip:+77019650957@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK0a77ed19;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>
Contact: <sip:+77787460019@22.22.22.22>
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Thu, 30 May 2013 11:05:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1019165131 1019165131 IN IP4 22.22.22.22
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 22.22.22.22
t=0 0
m=audio 13680 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called KCELL/+77019650957
trixbox1*CLI> 
<--- SIP read from UDP://11.11.11.11:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0a77ed19;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 INVITE


<------------->
--- (6 headers 0 lines) ---
trixbox1*CLI> 
<--- SIP read from UDP://11.11.11.11:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0a77ed19;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>;tag=SD7djfc99-0069395621
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Content-Type: application/sdp
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 242

v=0
o=- 10160356 10160356 IN IP4 11.11.11.11
s=-
c=IN IP4 11.11.11.11
t=0 0
a=sendrecv
m=audio 17772 RTP/AVP 8 101
c=IN IP4 11.11.11.11
a=rtpmap:8 PCMA/8000
a=maxptime:40
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 11.11.11.11:17772
    -- SIP/KCELL-00000001 is making progress passing it to SIP/1101-00000000
trixbox1*CLI> 
<--- SIP read from UDP://11.11.11.11:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0a77ed19;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>;tag=SD7djfc99-0069395621
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 INVITE
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
    -- SIP/KCELL-00000001 is ringing
Scheduling destruction of SIP dialog '628b56486ffdcab1517014cf4816d022@22.22.22.22' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 11.11.11.11:5060:
CANCEL sip:+77019650957@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK0a77ed19;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Content-Length: 0


---
Scheduling destruction of SIP dialog '628b56486ffdcab1517014cf4816d022@22.22.22.22' in 32000 ms (Method: INVITE)
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/1101-00000000' in macro 'dialout-trunk'
trixbox1*CLI> 
<--- SIP read from UDP://11.11.11.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0a77ed19;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>;tag=SD7djfc99-0069395621
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 CANCEL


<------------->
--- (6 headers 0 lines) ---
trixbox1*CLI> 
<--- SIP read from UDP://11.11.11.11:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0a77ed19;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>;tag=SD7djfc99-0069395621
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 INVITE
Reason: Q.850;cause=31
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 11.11.11.11:5060:
ACK sip:+77019650957@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK0a77ed19;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as04a90d8d
To: <sip:+77019650957@11.11.11.11>;tag=SD7djfc99-0069395621
Contact: <sip:+77787460019@22.22.22.22>
Call-ID: 628b56486ffdcab1517014cf4816d022@22.22.22.22
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Content-Length: 0


---
  == Spawn extension (from-internal, +77019650957, 4) exited non-zero on 'SIP/1101-00000000'
    -- Executing [h@from-internal:1] Macro("SIP/1101-00000000", "hangupcall") in new stack
Really destroying SIP dialog '628b56486ffdcab1517014cf4816d022@22.22.22.22' Method: INVITE
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1101-00000000", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1101-00000000", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1101-00000000", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/1101-00000000", "") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1101-00000000' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1101-00000000'
  == MixMonitor close filestream
  == End MixMonitor Recording SIP/1101-00000000
Дебаг с Acme:

Эластикс - Acme:

Код: Выделить всё

\00\00Q\00\00\00\00\00\00\00\00\00\00%\00|\00\00\00\00`F\00E\00\00\00U\00\009\00Y\00\00\00\00\00\00\00\00-INVITE sip:+77012112416@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK0fc81984;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as158946c6
To: <sip:+77012112416@11.11.11.11>
Contact: <sip:+77787460019@22.22.22.22:5060>
Call-ID: 4ced40eb2de53b2500117500533e85a5@22.22.22.22:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.21.0)
Date: Wed, 29 May 2013 09:43:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 860817334 860817334 IN IP4 22.22.22.22
s=Asterisk PBX 1.8.21.0
c=IN IP4 22.22.22.22
t=0 0
m=audio 12818 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
\00\00Q\00f\00\00[\00\00[\00\00\00\00\00\00\00\00\00bk\00E\00M\00\00\00\00+\00/\00wY\00\00\00\009\00SIP/2.0 100 Trying
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0fc81984;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as158946c6
To: <sip:+77012112416@11.11.11.11>
Call-ID: 4ced40eb2de53b2500117500533e85a5@22.22.22.22:5060
CSeq: 102 INVITE

\00\00Q}\00\00\00\00\00\00\00\00\00\00\00\00\00\00\00\00bk\00E\00~\00\00\00\00\00\00/\00wY\00\00\00\00joaSIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK0fc81984;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as158946c6
To: <sip:+77012112416@11.11.11.11>;tag=SD7tc8599-1122544651
Call-ID: 4ced40eb2de53b2500117500533e85a5@22.22.22.22:5060
CSeq: 102 INVITE
Content-Length: 0

\00\00QX\00\00\00\00\00\00\00\00\00%\00|\00\00\00\00`F\00E\00דV\00\009\00\00Y\00\00\00\00\00\00\00\00\00ACK sip:+77012112416@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK0fc81984;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as158946c6
To: <sip:+77012112416@11.11.11.11>;tag=SD7tc8599-1122544651
Contact: <sip:+77787460019@22.22.22.22:5060>
Call-ID: 4ced40eb2de53b2500117500533e85a5@22.22.22.22:5060
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.21.0)
Content-Length: 0
Триксбокс - Acme:

Код: Выделить всё

m\00\00Q\00\00\00\00\00\00\00\00\00\00\00\00bk\00\00\00\00\00E\00\00\00\00\00\00:`\00Y\00\00\00/\00w\00\00u\00\00INVITE sip:+77012112416@11.11.11.11 SIP/2.0
Via: SIP/2.0/UDP 22.22.22.22:5060;branch=z9hG4bK17e1c184;rport
Max-Forwards: 70
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as3856e4a6
To: <sip:+77012112416@11.11.11.11>
Contact: <sip:+77787460019@22.22.22.22>
Call-ID: 7dac6ade4e97a5260b237d9e0a982d3a@22.22.22.22
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
Date: Wed, 29 May 2013 14:42:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 2068256181 2068256181 IN IP4 22.22.22.22
s=Asterisk PBX 1.6.0.26-FONCORE-r78
c=IN IP4 22.22.22.22
t=0 0
m=audio 12180 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m\00\00Q\00\00V\00\00V\00\00\00\00\00\00\00\00\00bk\00E\00H\00\00\00\000\00/\00wY\00\00\00\004c\00SIP/2.0 100 Trying
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK17e1c184;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as3856e4a6
To: <sip:+77012112416@11.11.11.11>
Call-ID: 7dac6ade4e97a5260b237d9e0a982d3a@22.22.22.22
CSeq: 102 INVITE

p\00\00Qsl\00\00\00\00\00\00\00\00\00\00\00\00bk\00E\00\00\00\00\00\00\00\00\00/\00wY\00\00\00\00ᅋSIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK17e1c184;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as3856e4a6
To: <sip:+77012112416@11.11.11.11>;tag=SDeojb299-1960457372
Call-ID: 7dac6ade4e97a5260b237d9e0a982d3a@22.22.22.22
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,PRACK,UPDATE
Content-Type: application/sdp
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 242

v=0
o=- 13025816 13025816 IN IP4 11.11.11.11
s=-
c=IN IP4 11.11.11.11
t=0 0
a=sendrecv
m=audio 29656 RTP/AVP 8 101
c=IN IP4 11.11.11.11
a=rtpmap:8 PCMA/8000
a=maxptime:40
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
q\00\00Q\00<\00\00\00\00\00\00\00\00\00\00\00\00\00\00\00\00bk\00E\00\00w\00\00\00>\00\00\00\00Y\00\00s\00/\00\00\00\00\00\00\00"+\00\00\005\00\00\00՗\00\00\00\00\00\00\00\00`\00	kΐ\00\00\00\00\00\00\00\00i	\00b\00\00\00\00\00\00\00\00\00\00	dក\00\00\00\00\00\00G
p\00\00\00\00\00\00s
Z\00\00\00\00\00\00\00\00\00d	ޖ\00\00\00\00\00\00\00\00m\00	n\00\00\00\00\00\00\00j	\00`圁\00\00\00\00\00\00q\00\00Q\00\00\00\00\00\00\00\00\00\00\00\00\00\00\00\00\00bk\00E\00ȇO\00\00>\00^\00/\00wY\00\00s\00/\00\00\00\00\00\00\00#+\00\00D5\00\00\00W	x㙀\00\00\00\00\00\00H

J蚃\00\00\00\00\00\00~Q\00\00\00\00\00\00\00\00\00a\00	jÐ\00\00\00\00\00\00\00\00n	\00l\00\00\00\00\00\00\00\00\00\00	g瞁\00\00\00\00\00\00X
r웃\00\00\00\00\00\00q

Aꅂ\00\00\00\00\00\00e	r\00\00Q\00%\00\00\00\00\00\00\00\00\00\00\00\00\00\00bk\00E\00\00\00\00\00\00\00\00/\00wY\00\00\00\00\00"\00SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 22.22.22.22:5060;received=22.22.22.22;branch=z9hG4bK17e1c184;rport=5060
From: "+77787460019" <sip:+77787460019@22.22.22.22>;tag=as3856e4a6
To: <sip:+77012112416@11.11.11.11>;tag=SDeojb299-1960457372
Call-ID: 7dac6ade4e97a5260b237d9e0a982d3a@22.22.22.22
CSeq: 102 INVITE
Contact: <sip:172.16.240.20:5060;transport=udp>
Content-Length: 0
Общий вид (с начала 2 звонка с эластикса, потом с триксбокса): http://sendfile.su/818958
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Internetchik
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Re: Астериск + Acme Packet SBC 3820

Сообщение Internetchik »

ddkprog писал(а):вам сказали о дебаге астериска
а вы почему то нам показываете tcpdump

полные вербоус и дебаг + сип дебаг на 11 версии = где?

хотя я бы смотрел дебаг на вашем АЦМЕ
это он отбивает
и оптионс отбивает
и инвайты сначала говорит траинг а потом опять отбивает

дебажде ваш АЦМЕ

Ацме не наш, а провайдера. Выложил то, что дали. 11 версии сейчас тоже нет. На ее месте висит триксбокс.
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Internetchik
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Re: Астериск + Acme Packet SBC 3820

Сообщение Internetchik »

ded писал(а):
1xx responses are informational responses and in the case of 100 Trying are optional. SIP User Agent Servers (UAS) will generally respond with a 100 Trying response immediately when they receive an INVITE request to let the User Agent Client (UAC) know they are processing the request and to avoid retransmits. At some time later they will follow the 100 Trying response with a 180 Ringing or 183 Session Progress. Once someone or something answers the call a 2xx response needs to be sent, typically 200 Ok.

If your softphone software is only generating a 100 Trying response and not the subsequent 180 Ringing response then my guess is you have missed a step. At the very least if the softphone has a problem and can't generate a ringing response because there is nothing to ring it should generate a 4xx error response.
Всем привет.

Есть схема: Asterisk 1.6.2.17.3 <-SIP-> Iskratel Si2000v5 <-SS7/ISUP-> TDM

Со стороны TDM приходит вызов - нет КПВ. Причина - в ответ на входящий IAM сишка шлет как полагается, инвайт астериску, астериск отвечает 100 Trying, 180 Ringing и 183 Session Progress, в результате Si2000 пересылает в ISUP ACM Alerting, и далее CPG Progress, что в корне неправильно, ибо Alerting означает что вызываемый абонент найден и вызывается. Естетственно что TDM АТС, получая Progress после Alerting - затыкается.

Но тут, так как вызов транзитный, то астериск должен был ответить только с 183 Session Progress c SDP, чтобы станция послала в сторону TDM ACM c Progress Indicator=Inband early media, что подразумевает собой проключение тракта в предответном и начало передачи раннего аудио (КПВ, музычка, или голосовое сообщение).

Как заставить Астериск отвечать на инвайт только 183 Progress без 180 ?
Решение:

в sip.conf

prematuremedia = no
progressinband = never

далее в диалплане в любом месте перед вызовом, хоть перед Dial:

[context]
exten => _X.,1,Progress
exten => _X.,n,***********
http://asterisk-support.ru/question/156 ... iushchego/
ded, мне кажется это немного не мой вариант. В Вашем примере - проблема в пакетах с пометкой 180-183. А у меня до этого даже не доходит. Мне кажется проблема именно в немного измененном синтаксисе отправки инвайтов. А если это аналогия, извините я ее не понял.
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ded
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Re: Астериск + Acme Packet SBC 3820

Сообщение ded »

А попробовать? Зачем гадать - кажеться, не кажеться....
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jugatsu
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Re: Астериск + Acme Packet SBC 3820

Сообщение jugatsu »

Так провайдеру скидывал дамп или нет? Я заметил лишь небольшое различие в контакт хедере и сдп.
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jugatsu
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Re: Астериск + Acme Packet SBC 3820

Сообщение jugatsu »

У вас там авторизация судя по всему по ip, поскольку нет 401/407. Можно попробовать данный инвайт кастануть SIP INSPECTOR'ом или scapy и поиграться с хедерами.
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Internetchik
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Re: Астериск + Acme Packet SBC 3820

Сообщение Internetchik »

jugatsu писал(а):Так провайдеру скидывал дамп или нет? Я заметил лишь небольшое различие в контакт хедере и сдп.
Да скидывал! 2 недели с ними промаялись, они лишь руками разводят, мол "Непоняяятно..."
jugatsu писал(а):У вас там авторизация судя по всему по ip, поскольку нет 401/407. Можно попробовать данный инвайт кастануть SIP INSPECTOR'ом или scapy и поиграться с хедерами.
Хорошо займусь.
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Internetchik
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Re: Астериск + Acme Packet SBC 3820

Сообщение Internetchik »

ded писал(а):А попробовать? Зачем гадать - кажеться, не кажеться....
ded, извините, но я не могу разобраться куда подставить данную строку. В extensions* эластикса столько разных макро- и from- , что глаза разбегаются. Вы, случаем, не знаете название контекста?
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ded
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Re: Астериск + Acme Packet SBC 3820

Сообщение ded »

Пример

Код: Выделить всё

[context]
exten => _X.,1,Progress
exten => _X.,n,***********
является годным, но лабораторным. Для freePBX, Trixbox, Elastix, etc нужно вкуривать в структуру инклюдов в диалпланах.
В общих чертах: все исходящие оригинируются из dialplan from-internal

Код: Выделить всё

*CLI> dialplan show from-internal
[ Context 'from-internal' created by 'pbx_config' ]
  Include =>        'from-internal-noxfer'                        [pbx_config]
  Include =>        'from-internal-xfer'                          [pbx_config]
  Include =>        'bad-number'                                  [pbx_config]
Смотрим дальше -последовательно опрашивая эти диалпланы и добираемся до
*CLI> dialplan show from-internal-additional
где перечислены все app-а также внутренний план - ext-local
а в конце - искомое -
Include => 'outbound-allroutes'

То есть все исходящие из станции выбегают из 'outbound-allroutes
Если же посмотреть на него

Код: Выделить всё

*CLI> dialplan show outbound-allroutes
[ Context 'outbound-allroutes' created by 'pbx_config' ]
  'foo' =>          1. Noop(bar)                                  [pbx_config]
  Include =>        'outbound-allroutes-custom'                   [pbx_config]
  Include =>        'outrt-1'                                     [pbx_config]
  Include =>        'outrt-2'                                     [pbx_config]
  Include =>        'outrt-3'                                     [pbx_config]
То есть все исходящие маршруты - outrt-1, outrt-2, outrt-3, и все они имеют зародыши для создания кастомных трюков. Например outrt-1-custom

Код: Выделить всё

*CLI> dialplan show outrt-1
[ Context 'outrt-1' created by 'pbx_config' ]
  '_X.' =>          1. Macro(user-callerid,LIMIT,)                [pbx_config]
                    2. Set(INTRACOMPANYROUTE=YES)                 [pbx_config]
                    3. Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})}) [pbx_config]
                    4. Set(_NODEST=)                              [pbx_config]
                    5. Gosub(sub-record-check,s,1(out,${EXTEN},)) [pbx_config]
                    6. Macro(dialout-trunk,2,${EXTEN},)           [pbx_config]
                    7. Macro(outisbusy,)                          [pbx_config]
  Include =>        'outrt-1-custom'                              [pbx_config]
теоретически нам бы добавить этот Progress только для нужного роута, но где гарантии, что мы не поменяем приоритеты маршрутов через пол-года, и всё съедет вбок?
Я бы попробовал сделать в outbound-allroutes-custom, только для определённой маски (шаблона):

Код: Выделить всё

exten => _123X.,1,Progress
exten => _123X.,n,Goto(outrt-1,${EXTEN},1)
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