Почему не работает группа номеров для входящих вызовов (issabel 4, asterisk 11)?

Question

[Назарий Луговой]

Первый раз настраиваю сервер ip телефонии. Система virtual-box, Issabel 4, asterisk 11, внешка - два модема e1550 модуль chan_dongle.
Настроил транки модемов, три внутренних номера 100, 101, 102. Также настроил Исходящие и входящие маршруты. Все ок, звонит во все стороны.
Проблема появилась когда создал и подключил группу вызовов (ring groups) на входящий маршрут (inbound routes).
Настройки группы:

Настройки входящего маршрута

При звонке на входящий маршрут, он сразу сбрасывается (кладет трубку), если в направлении звонка я указываю любой внутренний номер, вместо группы, звонок удачно проходит.
Если в настройках группы я указываю внутренний номер "если никто не ответил", то звонок сразу приходит на него.

Comments

[Drill]

Попрорбуйте изменить стратегию дозвона на "ringall"
А вообще, вместо гадать, почему не так, снимите дамп во время звонка и там все увидите.

[Назарий Луговой]

"Попрорбуйте изменить стратегию дозвона на "ringall" " - пробовал все варианты.
"снимите дамп во время звонка" - как это сделать? В CLI при входящем звонке ничего не выводится, просто со стороны АТС ложится трубка.

[Drill]

В CLI при входящем звонке ничего не выводится, просто со стороны АТС ложится трубка.

Чтобы сделать вывод логирования информативнее:
cli> core set verbose 3
(как минимум 3, можно больмше, вплоть до 9)

"снимите дамп во время звонка" - как это сделать?

cli> sip set debug on
И ловите INVITE и ответы на него.
дамп можно выложить здесь, только не простыню, а по существу.

[Drill]

И еще, прежде чем задавать, такого плана вопросы, следует приложить немного усилий для изучения тематики по существу вопроса. Все легко ищется в гугле.
1. Командная строка астериск
2. How To Debug and Troubleshoot VOIP
3. как понять что происходит на астериске

[Назарий Луговой]

Кусок Debug во время звонка, не ожидал что там всего так много..

spoiler

<------------->
Reliably Transmitting (no NAT) to 192.168.3.27:1042:
OPTIONS sip:101@192.168.3.27:1042;rinstance=47fbe190a6e5ef79 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK4d61c5f8
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as38365f1f
To: <sip:101@192.168.3.27:1042;rinstance=47fbe190a6e5ef79>
Contact: <sip:Unknown@192.168.3.151:5060>
Call-ID: 3a098cb85f913a1419719e150f57df29@192.168.3.151:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(11.25.3)
Date: Tue, 22 Jan 2019 16:45:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.3.27:1042 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK4d61c5f8
Contact: <sip:192.168.3.27:1042>
To: <sip:101@192.168.3.27:1042;rinstance=47fbe190a6e5ef79>;tag=1b78e377
From: "Unknown"<sip:Unknown@192.168.3.151>;tag=as38365f1f
Call-ID: 3a098cb85f913a1419719e150f57df29@192.168.3.151:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Supported: replaces
Allow-Events: presence, message-summary, tunnel-info
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '3a098cb85f913a1419719e150f57df29@192.168.3.151:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.3.27:1036 --->
REGISTER sip:192.168.3.151 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---c9ec176b5a12f111;rport
Max-Forwards: 70
Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
To: "102"<sip:102@192.168.3.151>
From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
CSeq: 23 REGISTER
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
User-Agent: X-Lite release 5.4.0 stamp 94388
Authorization: Digest username="102",realm="asterisk",nonce="62337258",uri="sip:192.168.3.151",response="9791ea3712f4b35a1c03370ccfeead2a",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.3.27:1036 (NAT)
Sending to 192.168.3.27:1036 (NAT)

<--- Transmitting (no NAT) to 192.168.3.27:1036 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---c9ec176b5a12f111;received=192.168.3.27;rport=1036
From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
To: "102"<sip:102@192.168.3.151>;tag=as36cc7be4
Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
CSeq: 23 REGISTER
Server: IPBX-2.11.0(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f4a54a0"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.3.27:1036 --->
REGISTER sip:192.168.3.151 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---e099ef307998b60c;rport
Max-Forwards: 70
Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
To: "102"<sip:102@192.168.3.151>
From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
CSeq: 24 REGISTER
Expires: 3600
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
User-Agent: X-Lite release 5.4.0 stamp 94388
Authorization: Digest username="102",realm="asterisk",nonce="3f4a54a0",uri="sip:192.168.3.151",response="77826891b1876db5eb1c1cf864ccb4d4",algorithm=MD5
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.3.27:1036 (no NAT)
Reliably Transmitting (no NAT) to 192.168.3.27:1036:
OPTIONS sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK3ac2a7f8
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as28bdd887
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
Contact: <sip:Unknown@192.168.3.151:5060>
Call-ID: 25b7d5f45e13bb056cd0d2a02fa54cbe@192.168.3.151:5060
CSeq: 102 OPTIONS
User-Agent: IPBX-2.11.0(11.25.3)
Date: Tue, 22 Jan 2019 16:45:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (no NAT) to 192.168.3.27:1036 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---e099ef307998b60c;received=192.168.3.27;rport=1036
From: "102"<sip:102@192.168.3.151>;tag=604c8e4f
To: "102"<sip:102@192.168.3.151>;tag=as36cc7be4
Call-ID: 94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA
CSeq: 24 REGISTER
Server: IPBX-2.11.0(11.25.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;expires=3600
Date: Tue, 22 Jan 2019 16:45:28 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.3.27:1036:
NOTIFY sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK43f52759
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as23446dda
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
Contact: <sip:Unknown@192.168.3.151:5060>
Call-ID: 070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060
CSeq: 102 NOTIFY
User-Agent: IPBX-2.11.0(11.25.3)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:*97@192.168.3.151
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '94388MWRlNzM2ZWY0NDBlNThlNzViNDY3ZTRkY2QzZjQwMDA' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.3.27:1036 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK3ac2a7f8
Contact: <sip:192.168.3.27:1036>
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=6bdf1067
From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as28bdd887
Call-ID: 25b7d5f45e13bb056cd0d2a02fa54cbe@192.168.3.151:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Allow-Events: talk, hold
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '25b7d5f45e13bb056cd0d2a02fa54cbe@192.168.3.151:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.3.27:1036 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK43f52759
Contact: <sip:192.168.3.27:1036>
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=1ed0cc08
From: "Unknown" <sip:Unknown@192.168.3.151>;tag=as23446dda
Call-ID: 070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060
CSeq: 102 NOTIFY
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '070a43506ee481e75c4c7bc27936efb2@192.168.3.151:5060' Method: NOTIFY

<--- SIP read from UDP:192.168.3.27:1042 --->

[Назарий Луговой]

cli> core set verbose 3 при звонке

spoiler

== Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'Dongle/dongle0-0100000019'
    -- Executing [380935591582@from-pstn:1] Set("Dongle/dongle0-010000001a", "__FROM_DID=380935591582") in new stack
    -- Executing [380935591582@from-pstn:2] Gosub("Dongle/dongle0-010000001a", "app-blacklist-check,s,1()") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("Dongle/dongle0-010000001a", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("Dongle/dongle0-010000001a", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("Dongle/dongle0-010000001a", "") in new stack
    -- Executing [380935591582@from-pstn:3] Set("Dongle/dongle0-010000001a", "CDR(did)=380935591582") in new stack
    -- Executing [380935591582@from-pstn:4] ExecIf("Dongle/dongle0-010000001a", "0 ?Set(CALLERID(name)=[phone])") in new stack
    -- Executing [380935591582@from-pstn:5] Set("Dongle/dongle0-010000001a", "CHANNEL(musicclass)=default") in new stack
    -- Executing [380935591582@from-pstn:6] Set("Dongle/dongle0-010000001a", "__MOHCLASS=default") in new stack
    -- Executing [380935591582@from-pstn:7] Set("Dongle/dongle0-010000001a", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [380935591582@from-pstn:8] Set("Dongle/dongle0-010000001a", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [380935591582@from-pstn:9] Goto("Dongle/dongle0-010000001a", "ext-group,600,1") in new stack
    -- Goto (ext-group,600,1)
    -- Executing [600@ext-group:1] Macro("Dongle/dongle0-010000001a", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("Dongle/dongle0-010000001a", "TOUCH_MONITOR=1548176187.52") in new stack
    -- Executing [s@macro-user-callerid:2] Set("Dongle/dongle0-010000001a", "AMPUSER=[phone]") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("Dongle/dongle0-010000001a", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("Dongle/dongle0-010000001a", "1?Set(REALCALLERIDNUM=[phone])") in new stack
    -- Executing [s@macro-user-callerid:5] Set("Dongle/dongle0-010000001a", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("Dongle/dongle0-010000001a", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("Dongle/dongle0-010000001a", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("Dongle/dongle0-010000001a", "1?report") in new stack
    -- Goto (macro-user-callerid,s,15)
    -- Executing [s@macro-user-callerid:15] GotoIf("Dongle/dongle0-010000001a", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:16] Set("Dongle/dongle0-010000001a", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("Dongle/dongle0-010000001a", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,28)
    -- Executing [s@macro-user-callerid:28] Set("Dongle/dongle0-010000001a", "CALLERID(number)=[phone]") in new stack
    -- Executing [s@macro-user-callerid:29] Set("Dongle/dongle0-010000001a", "CALLERID(name)=dongle0") in new stack
    -- Executing [s@macro-user-callerid:30] Set("Dongle/dongle0-010000001a", "CDR(cnum)=[phone]") in new stack
    -- Executing [s@macro-user-callerid:31] Set("Dongle/dongle0-010000001a", "CDR(cnam)=dongle0") in new stack
    -- Executing [s@macro-user-callerid:32] Set("Dongle/dongle0-010000001a", "CHANNEL(language)=en") in new stack
    -- Executing [600@ext-group:2] Macro("Dongle/dongle0-010000001a", "blkvm-setifempty,") in new stack
    -- Executing [s@macro-blkvm-setifempty:1] GotoIf("Dongle/dongle0-010000001a", "1?init") in new stack
    -- Goto (macro-blkvm-setifempty,s,4)
    -- Executing [s@macro-blkvm-setifempty:4] Set("Dongle/dongle0-010000001a", "__BLKVM_CHANNEL=Dongle/dongle0-010000001a") in new stack
    -- Executing [s@macro-blkvm-setifempty:5] Set("Dongle/dongle0-010000001a", "SHARED(BLKVM,Dongle/dongle0-010000001a)=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:6] Set("Dongle/dongle0-010000001a", "GOSUB_RETVAL=TRUE") in new stack
    -- Executing [s@macro-blkvm-setifempty:7] MacroExit("Dongle/dongle0-010000001a", "") in new stack
    -- Executing [600@ext-group:3] GotoIf("Dongle/dongle0-010000001a", "1?skipov") in new stack
    -- Goto (ext-group,600,6)
    -- Executing [600@ext-group:6] Set("Dongle/dongle0-010000001a", "RRNODEST=") in new stack
    -- Executing [600@ext-group:7] Set("Dongle/dongle0-010000001a", "__NODEST=600") in new stack
    -- Executing [600@ext-group:8] GosubIf("Dongle/dongle0-010000001a", "0?sub-rgsetcid,s,1()") in new stack
    -- Executing [600@ext-group:9] Gosub("Dongle/dongle0-010000001a", "sub-record-check,s,1(rg,600,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] Set("Dongle/dongle0-010000001a", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:2] GotoIf("Dongle/dongle0-010000001a", "1?check") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [s@sub-record-check:7] Set("Dongle/dongle0-010000001a", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:8] GotoIf("Dongle/dongle0-010000001a", "1?next") in new stack
    -- Goto (sub-record-check,s,11)
    -- Executing [s@sub-record-check:11] ExecIf("Dongle/dongle0-010000001a", "0?Return()") in new stack
    -- Executing [s@sub-record-check:12] ExecIf("Dongle/dongle0-010000001a", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
    -- Executing [s@sub-record-check:13] GotoIf("Dongle/dongle0-010000001a", "0?rg,1") in new stack
    -- Executing [s@sub-record-check:14] Set("Dongle/dongle0-010000001a", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:15] Set("Dongle/dongle0-010000001a", "NOW=1548176187") in new stack
    -- Executing [s@sub-record-check:16] Set("Dongle/dongle0-010000001a", "__DAY=22") in new stack
    -- Executing [s@sub-record-check:17] Set("Dongle/dongle0-010000001a", "__MONTH=01") in new stack
    -- Executing [s@sub-record-check:18] Set("Dongle/dongle0-010000001a", "__YEAR=2019") in new stack
    -- Executing [s@sub-record-check:19] Set("Dongle/dongle0-010000001a", "__TIMESTR=20190122-185627") in new stack
    -- Executing [s@sub-record-check:20] Set("Dongle/dongle0-010000001a", "__FROMEXTEN=[phone]") in new stack
    -- Executing [s@sub-record-check:21] Set("Dongle/dongle0-010000001a", "__CALLFILENAME=rg-600-[phone]-20190122-185627-1548176187.52") in new stack
    -- Executing [s@sub-record-check:22] Goto("Dongle/dongle0-010000001a", "rg,1") in new stack
    -- Goto (sub-record-check,rg,1)
    -- Executing [rg@sub-record-check:1] GosubIf("Dongle/dongle0-010000001a", "0?record,1(rg,dontcare,[phone])") in new stack
    -- Executing [rg@sub-record-check:2] Return("Dongle/dongle0-010000001a", "") in new stack
    -- Executing [600@ext-group:10] Set("Dongle/dongle0-010000001a", "RingGroupMethod=ringall") in new stack
    -- Executing [600@ext-group:11] Macro("Dongle/dongle0-010000001a", "dial,80,Ttr,101-102-100") in new stack
    -- Executing [s@macro-dial:1] GotoIf("Dongle/dongle0-010000001a", "0?dial") in new stack
    -- Executing [s@macro-dial:2] Set("Dongle/dongle0-010000001a", "CHANNEL(musicclass)=default") in new stack
    -- Executing [s@macro-dial:3] AGI("Dongle/dongle0-010000001a", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Failed to execute '/var/lib/asterisk/agi-bin/dialparties.agi': Permission denied
    -- Executing [s@macro-dial:4] NoOp("Dongle/dongle0-010000001a", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
    -- Executing [600@ext-group:12] Gosub("Dongle/dongle0-010000001a", "sub-record-cancel,s,1()") in new stack
    -- Executing [s@sub-record-cancel:1] Set("Dongle/dongle0-010000001a", "__REC_POLICY_MODE=") in new stack
    -- Executing [s@sub-record-cancel:2] ExecIf("Dongle/dongle0-010000001a", "1?Return()") in new stack
    -- Executing [600@ext-group:13] Set("Dongle/dongle0-010000001a", "RingGroupMethod=") in new stack
    -- Executing [600@ext-group:14] GotoIf("Dongle/dongle0-010000001a", "0?nodest") in new stack
    -- Executing [600@ext-group:15] Set("Dongle/dongle0-010000001a", "__NODEST=") in new stack
    -- Executing [600@ext-group:16] Macro("Dongle/dongle0-010000001a", "blkvm-clr,") in new stack
    -- Executing [s@macro-blkvm-clr:1] Set("Dongle/dongle0-010000001a", "SHARED(BLKVM,Dongle/dongle0-010000001a)=") in new stack
    -- Executing [s@macro-blkvm-clr:2] Set("Dongle/dongle0-010000001a", "GOSUB_RETVAL=") in new stack
    -- Executing [s@macro-blkvm-clr:3] MacroExit("Dongle/dongle0-010000001a", "") in new stack
    -- Executing [600@ext-group:17] Goto("Dongle/dongle0-010000001a", "app-blackhole,hangup,1") in new stack
    -- Goto (app-blackhole,hangup,1)
    -- Executing [hangup@app-blackhole:1] NoOp("Dongle/dongle0-010000001a", "Blackhole Dest: Hangup") in new stack
    -- Executing [hangup@app-blackhole:2] Hangup("Dongle/dongle0-010000001a", "") in new stack
  == Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'Dongle/dongle0-010000001a'

[Drill]

То что выложили (REGISTER, OPTIONS) - это все не относится к звонку.

И ловите INVITE и ответы на него.

Если не знаете, что это, сначала почитайте упрощенный вариантВзаимодействие клиентов SIP

[Назарий Луговой]

Спасибо за статью!
INVITE это примерно это:

spoiler

Audio is at 15586
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.3.27:1036:
INVITE sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
Max-Forwards: 70
From: "dongle0" <sip:[phone]@192.168.3.151>;tag=as74726125
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
Contact: <sip:[phone]@192.168.3.151:5060>
Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
CSeq: 102 INVITE
User-Agent: IPBX-2.11.0(11.25.3)
Date: Tue, 22 Jan 2019 17:32:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1265205272 1265205272 IN IP4 192.168.3.151
s=Asterisk PBX 11.25.3
c=IN IP4 192.168.3.151
t=0 0
m=audio 15586 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/102
[2019-01-22 19:32:10] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 22
[2019-01-22 19:32:10] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 22

<--- SIP read from UDP:192.168.3.27:1036 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
From: "dongle0" <sip:[phone]@192.168.3.151>;tag=as74726125
Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
[2019-01-22 19:32:11] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 33

<--- SIP read from UDP:192.168.3.27:1036 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
To: "102"<sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
From: "dongle0" <sip:[phone]@192.168.3.151>;tag=as74726125
Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 5.4.0 stamp 94388
Allow-Events: talk, hold
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
[2019-01-22 19:32:11] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 33
    -- SIP/102-00000014 is ringing

<--- SIP read from UDP:192.168.3.27:1036 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK6afc17f5
Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
From: "dongle0" <sip:[phone]@192.168.3.151>;tag=as74726125
Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
CSeq: 102 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 204

v=0
o=- 2754198276 3 IN IP4 192.168.3.27
s=X-Lite release 5.4.0 stamp 94388
c=IN IP4 192.168.3.27
t=0 0
m=audio 61040 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.3.27:61040
list_route: hop: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
set_destination: Parsing <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715> for address/port to send to
set_destination: set destination to 192.168.3.27:1036
Transmitting (no NAT) to 192.168.3.27:1036:
ACK sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.151:5060;branch=z9hG4bK0f593c5b
Max-Forwards: 70
From: "dongle0" <sip:[phone]@192.168.3.151>;tag=as74726125
To: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
Contact: <sip:[phone]@192.168.3.151:5060>
Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
CSeq: 102 ACK
User-Agent: IPBX-2.11.0(11.25.3)
Content-Length: 0


---
[2019-01-22 19:32:13] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 33
[2019-01-22 19:32:13] WARNING[7523][C-00000036]: channel.c:1002 channel_indicate: [Dongle/dongle0-0100000021] Don't know how to indicate condition 22
    -- SIP/102-00000014 answered Dongle/dongle0-0100000021

<--- SIP read from UDP:192.168.3.27:1036 --->
BYE sip:[phone]@192.168.3.151:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.27:1036;branch=z9hG4bK-524287-1---70976675f8e00f05;rport
Max-Forwards: 70
Contact: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>
To: "dongle0" <sip:[phone]@192.168.3.151>;tag=as74726125
From: <sip:102@192.168.3.27:1036;rinstance=5c870cdd20add715>;tag=31158b0a
Call-ID: 2ff49f5d67e95131789afa962b474be1@192.168.3.151:5060
CSeq: 2 BYE
User-Agent: X-Lite release 5.4.0 stamp 94388
Content-Length: 0

примерно от invite до bye.
Это я зачитал когда позвонил на внутренний номер, при звонке на группу invite нет вообще.

[Drill]

Значит, что-то не договариваете.
1. Покажите настройку транка, через который идет входящий звонок.
2. Покажите вывод команды: "sip show peers"

[Назарий Луговой]

настройка trunk

'dongle show devices' и 'sip show peers'

[Назарий Луговой]

У меня еще есть trunk с купленным номером задарма.com, но я его отключил, чтоб логи не засорял. С ним логи не смотрел правда, но смысл идентичен: на внутренние он дозванивался сразу, на группу, отбивал сразу занято.

[Drill]

задарма.com, ... на внутренние он дозванивался сразу, на группу, отбивал сразу занято.

Это меняет диспозицию. Тогда, похоже проблема не в донгл.
1. Покажите dongle.conf
2. Покажите настройку транка задарма.com, только пароли не показывайте.
3. Входящий маршрут для транка задарма.com

[Drill]

Вдогонку:
4. Покажите вывод команды: "dialplan show ext-did-0002"

[Назарий Луговой]

dongle.conf

spoiler

[general]

interval=15			; Number of seconds between trying to connect to devices

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes			; Enables the use of a jitterbuffer on the receiving side of a
				; Dongle channel. Defaults to "no". An enabled jitterbuffer will
				; be used only if the sending side can create and the receiving
				; side can not accept jitter. The Dongle channel can't accept jitter,
				; thus an enabled jitterbuffer on the receive Dongle side will always
				; be used if the sending side can create jitter.

;jbforce = no			; Forces the use of a jitterbuffer on the receive side of a Dongle
				; channel. Defaults to "no".

;jbmaxsize = 200		; Max length of the jitterbuffer in milliseconds.

;jbresyncthreshold = 1000	; Jump in the frame timestamps over which the jitterbuffer is
				; resynchronized. Useful to improve the quality of the voice, with
				; big jumps in/broken timestamps, usually sent from exotic devices
				; and programs. Defaults to 1000.

;jbimpl = fixed			; Jitterbuffer implementation, used on the receiving side of a Dongle
				; channel. Two implementations are currently available - "fixed"
				; (with size always equals to jbmaxsize) and "adaptive" (with
				; variable size, actually the new jb of IAX2). Defaults to fixed.

;jbtargetextra = 40		; This option only affects the jb when 'jbimpl = adaptive' is set.
				; The option represents the number of milliseconds by which the new jitter buffer
				; will pad its size. the default is 40, so without modification, the new
				; jitter buffer will set its size to the jitter value plus 40 milliseconds.
				; increasing this value may help if your network normally has low jitter,
				; but occasionally has spikes.

;jblog = no			; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

[defaults]
; now you can set here any not required device settings as template
;   sure you can overwrite in any [device] section this default values

;context=default			; context for incoming calls
;context=from-gsm
context=from-pstn
group=0				; calling group
rxgain=0			; increase the incoming volume; may be negative
txgain=0			; increase the outgoint volume; may be negative
autodeletesms=yes		; auto delete incoming sms
resetdongle=yes			; reset dongle during initialization with ATZ command
u2diag=-1			; set ^U2DIAG parameter on device (0 = disable everything except modem function) ; -1 not use ^U2DIAG command
usecallingpres=yes		; use the caller ID presentation or not
callingpres=allowed_passed_screen ; set caller ID presentation		by default use default network settings
disablesms=no			; disable of SMS reading from device when received
				;  chan_dongle has currently a bug with SMS reception. When a SMS gets in during a
				;  call chan_dongle might crash. Enable this option to disable sms reception.
				;  default = no

language=ru			; set channel default language
smsaspdu=yes			; if 'yes' send SMS in PDU mode, feature implementation incomplete and we strongly recommend say 'yes'
mindtmfgap=45			; minimal interval from end of previews DTMF from begining of next in ms
mindtmfduration=80		; minimal DTMF tone duration in ms
mindtmfinterval=200		; minimal interval between ends of DTMF of same digits in ms

callwaiting=auto		; if 'yes' allow incoming calls waiting; by default use network settings
				; if 'no' waiting calls just ignored
disable=no			; OBSOLETED by initstate: if 'yes' no load this device and just ignore this section

initstate=start			; specified initial state of device, must be one of 'stop' 'start' 'remote'
				;   'remove' same as 'disable=yes'

;exten=[phone]		; exten for start incoming calls, only in case of Subscriber Number not available!, also set to CALLERID(ndid)

dtmf=relax			; control of incoming DTMF detection, possible values:
				;   off	   - off DTMF tones detection, voice data passed to asterisk unaltered
				;              use this value for gateways or if not use DTMF for AVR or inside dialplan
				;   inband - do DTMF tones detection
				;   relax  - like inband but with relaxdtmf option
				;  default is 'relax' by compatibility reason

[dongle0] 
audio=/dev/ttyUSB1
data=/dev/ttyUSB2

[dongle1]
audio=/dev/ttyUSB4
data=/dev/ttyUSB5

задарма

dialplan show ext-did-0002

spoiler

issabel*CLI> dialplan show ext-did-0002
[ Context 'ext-did-0002' created by 'pbx_config' ]
  '380935591582' => 1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                    3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                    4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                    5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                    6. Set(__MOHCLASS=default)                    [pbx_config]
                    7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                    8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
     [dest-ext]     9. Goto(from-did-direct,102,1)                [pbx_config]
  'fax' =>          1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [pbx_config]
  Include =>        'ext-did-0002-custom'                         [pbx_config]

-= 2 extensions (10 priorities) in 1 context. =-

так-же проверил функцию очереди, работает.

[Drill]

Не показали

Входящий маршрут для транка задарма.com

Давайте проверим, как через задарма ходит звонок.
1. Транк задарма - "контекст USER" и "опции USER" уберите вообще. Оставьте только "опции PEER". Проверьте, что в "опции PEER" есть "context=from-trunk". Сохраните изменения.
2. Включите транк задарма.
3. Во входящем маршруте для задарма установите направление на группу. Сохраните изменения.
4. Еще раз покажите вывод команды: "dialplan show ext-did-0002"
5. Включите дамп.
6. Сделайте входящий звонок через задарма и покажите дамп.

[Назарий Луговой]

0.

1. сделал
2. сделал
3. сделал
4.

spoiler

issabel*CLI> dialplan show ext-did-0002
[ Context 'ext-did-0002' created by 'pbx_config' ]
  '134628' =>       1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                    3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                    4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                    5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                    6. Set(__MOHCLASS=default)                    [pbx_config]
                    7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                    8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
     [dest-ext]     9. Goto(from-did-direct,102,1)                [pbx_config]
  'fax' =>          1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [pbx_config]
  Include =>        'ext-did-0002-custom'                         [pbx_config]

-= 2 extensions (10 priorities) in 1 context. =-

5. сделал
6.

spoiler

<--- SIP read from UDP:185.45.152.174:5060 --->
<--- SIP read from UDP:185.45.152.174:5060 --->
INVITE sip:134628@192.168.3.1:5060 SIP/2.0
Record-Route: <sip:185.45.152.174;lr=on;ftag=as228d9596;nat=yes>
Via: SIP/2.0/UDP 185.45.152.174;branch=z9hG4bKaed4.aacd7fc1cbaa307edc993b466321f392.0
Via: SIP/2.0/UDP 185.45.152.131:5060;rport=5060;branch=z9hG4bK7fea108e
Max-Forwards: 69
From: "[phone]" <sip:[phone]@sip.задарма.com>;tag=as228d9596
To: <sip:134628@185.45.152.174>
Contact: <sip:[phone]@185.45.152.131:5060>
Call-ID: 27e390fc011aade23271ed911e86621b@185.45.152.131:5060
CSeq: 102 INVITE
User-Agent: задарма Voip
Date: Wed, 23 Jan 2019 10:56:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
CALLED_DID: 380947117050
Content-Type: application/sdp
Content-Length: 411
hostname: sipbalancer-2
cc_num: 134628
cc_counter: 1

v=0
o=root 80984630 80984630 IN IP4 185.45.152.176
s=задарма Voip
c=IN IP4 185.45.152.176
t=0 0
m=audio 18946 RTP/AVP 8 0 18 110 117 119 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

в логе задарма переписал на русский чтоб система пропустила

[Drill]

пакет INVITE вижу, а где остальные? (100 Trying, 180 Ringing, 200 Ok, ACK, BYE)
в общем все пакеты из одной последовательности (смотри поле CSeq: 102 INVITE)
Звонок вообще прошел?

и я не вижу, что установлено направлениее на группу, остался маршрут на номер 102.
[dest-ext] 9. Goto(from-did-direct,102,1)
а должно быть
Goto(ext-group,600,1)

[Drill]

Не надо постить логи в ответы темы.
Только все пакеты из одной последовательности (смотри поле CSeq: 102 INVITE), а не CSeq: 102 OPTIONS
Чтобы удобнее было снимать дамп, поставьте утилиту Sngrep: установка и использование

[Drill]

И покажите вывод команды: "dialplan show 134628@from-trunk"

[Назарий Луговой]

Не надо постить логи в ответы темы.

дело не в этом, почему-то не мог сюда отправить, кнопка "отправить" активна, нажимаю но ничего не происходит. чистил кеш, пробовал разные браузеры, без толку. Вот и написал в ответы.

spoiler

issabel*CLI> dialplan show 134628@from-trunk
[ Included context 'ext-did-catchall' created by 'pbx_config' ]
  '_.' =>           1. Noop(Catch-All DID Match - Found ${EXTEN} - You probably want a DID for this.) [pbx_config]
                    2. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    3. Goto(ext-did,s,1)                          [pbx_config]

-= 1 extension (3 priorities) in 1 context. =-

[Назарий Луговой]

[Drill]

Еще раз. Во входящем маршруте для задарма установите направление на группу.
покажите вывод команд:
1. "dialplan show from-pstn"
2. "dialplan show ext-did"
3. "dialplan show ext-did-0001"
4. "dialplan show ext-did-0002"

[Назарий Луговой]

Еще раз. Во входящем маршруте для задарма установите направление на группу.
покажите вывод команд:

Может я не верно понимаю смысл этих слов?

вывод команд:

spoiler

issabel*CLI> dialplan show from-pstn
[ Context 'from-pstn' created by 'pbx_config' ]
  'sms' =>          1. Verbose(Incoming SMS from ${CALLERID(num)} ${BASE64_DECODE(${SMS_BASE64})}) [pbx_config]
                    2. System(echo '${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${DONGLENAME} - ${CALLERID(num)}: ${BASE64_DECODE(${SMS_BASE64})}' >> /var/log/asterisk/sms.txt) [pbx_config]
                    3. System(echo '${SMS} === ${CALLERID(num)} - ${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)}' | mail -s IncomingSMS [e-mail]) [pbx_config]
                    4. Hangup()                                   [pbx_config]
  'ussd' =>         1. Verbose(Incoming USSD: ${BASE64_DECODE(${USSD_BASE64})}) [pbx_config]
                    2. System(echo "${DONGLENAME} - ${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} - ${BASE64_DECODE(${USSD_BASE64})}" > /var/log/asterisk/ussd_${DONGLENAME}.txt) [pbx_config]
                    3. Hangup()                                   [pbx_config]
  Include =>        'from-pstn-custom'                            [pbx_config]
  Include =>        'ext-did'                                     [pbx_config]
  Include =>        'ext-did-post-custom'                         [pbx_config]
  Include =>        'from-did-direct'                             [pbx_config]
  Include =>        'ext-did-catchall'                            [pbx_config]

-= 2 extensions (7 priorities) in 1 context. =-

issabel*CLI> dialplan show ext-did
[ Context 'ext-did' created by 'pbx_config' ]
  'foo' =>          1. Noop(bar)                                  [pbx_config]
  Include =>        'ext-did-custom'                              [pbx_config]
  Include =>        'ext-did-0001'                                [pbx_config]
  Include =>        'ext-did-0002'                                [pbx_config]

-= 1 extension (1 priority) in 1 context. =-

issabel*CLI> dialplan show ext-did-0001
[ Context 'ext-did-0001' created by 'pbx_config' ]
  'fax' =>          1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [pbx_config]
  Include =>        'ext-did-0001-custom'                         [pbx_config]

-= 1 extension (1 priority) in 1 context. =-

issabel*CLI> dialplan show ext-did-0002
[ Context 'ext-did-0002' created by 'pbx_config' ]
  '134628' =>       1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                    3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                    4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                    5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                    6. Set(__MOHCLASS=default)                    [pbx_config]
                    7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                    8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
     [dest-ext]     9. Goto(ext-group,600,1)                      [pbx_config]
  'fax' =>          1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [pbx_config]
  Include =>        'ext-did-0002-custom'                         [pbx_config]

-= 2 extensions (10 priorities) in 1 context. =-

[Drill]

DID правильный, и направление тоже.

А теперь
dialplan show ext-did-catchall
dialplan show 134628@from-pstn
sip show peers

[Drill]

и еще
dialplan show ext-group

[Назарий Луговой]

spoiler

issabel*CLI> dialplan show ext-did-catchall
[ Context 'ext-did-catchall' created by 'pbx_config' ]
  'h' =>            1. Hangup()                                   [pbx_config]
  's' =>            1. Noop(No DID or CID Match)                  [pbx_config]
     [a2]           2. Answer()                                   [pbx_config]
                    3. Wait(2)                                    [pbx_config]
                    4. Playback(ss-noservice)                     [pbx_config]
                    5. SayAlpha(${FROM_DID})                      [pbx_config]
                    6. Hangup()                                   [pbx_config]
  '_.' =>           1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Noop(Received an unknown call with DID set to ${EXTEN}) [pbx_config]
                    3. Goto(s,a2)                                 [pbx_config]
  Include =>        'ext-did-catchall-custom'                     [pbx_config]

-= 3 extensions (10 priorities) in 1 context. =-

issabel*CLI> dialplan show 134628@from-pstn
[ Included context 'ext-did-0002' created by 'pbx_config' ]
  '134628' =>       1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                    3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                    4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                    5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                    6. Set(__MOHCLASS=default)                    [pbx_config]
                    7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                    8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
     [dest-ext]     9. Goto(from-did-direct,102,1)                [pbx_config]

[ Included context 'ext-did-catchall' created by 'pbx_config' ]
  '_.' =>           1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Noop(Received an unknown call with DID set to ${EXTEN}) [pbx_config]
                    3. Goto(s,a2)                                 [pbx_config]

-= 2 extensions (12 priorities) in 2 contexts. =-

issabel*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
100/100                   192.168.3.10                             D  No         No          A  33732    OK (429 ms)
101/101                   192.168.3.27                             D  No         No          A  1028     OK (104 ms)
102/102                   192.168.3.27                             D  No         No          A  4420     OK (3 ms)
задарма/134628            185.45.152.174                              Yes        Yes            5060     OK (75 ms)
4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline]

issabel*CLI> dialplan show ext-group
[ Context 'ext-group' created by 'pbx_config' ]
  '600' =>          1. Macro(user-callerid,)                      [pbx_config]
                    2. Macro(blkvm-setifempty,)                   [pbx_config]
                    3. GotoIf($["${GOSUB_RETVAL}" = "TRUE"]?skipov) [pbx_config]
                    4. Macro(blkvm-set,reset)                     [pbx_config]
                    5. Set(__NODEST=)                             [pbx_config]
     [skipov]       6. Set(RRNODEST=${NODEST})                    [pbx_config]
     [skipvmblk]    7. Set(__NODEST=${EXTEN})                     [pbx_config]
                    8. GosubIf($[${DB_EXISTS(RINGGROUP/600/changecid)} = 1 & "${DB(RINGGROUP/600/changecid)}" != "default" & "${DB(RINGGROUP/600/changecid)}" != ""]?sub-rgsetcid,s,1()) [pbx_config]
                    9. Gosub(sub-record-check,s,1(rg,600,dontcare)) [pbx_config]
                    10. Set(RingGroupMethod=ringall)              [pbx_config]
     [DIALGRP]      11. Macro(dial,80,${DIAL_OPTIONS},101-102-100) [pbx_config]
                    12. Gosub(sub-record-cancel,s,1())            [pbx_config]
                    13. Set(RingGroupMethod=)                     [pbx_config]
                    14. GotoIf($["foo${RRNODEST}" != "foo"]?nodest) [pbx_config]
                    15. Set(__NODEST=)                            [pbx_config]
                    16. Macro(blkvm-clr,)                         [pbx_config]
                    17. Goto(app-blackhole,hangup,1)              [pbx_config]
     [nodest]       18. Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST}) [pbx_config]
  '601' =>          1. Macro(user-callerid,)                      [pbx_config]
                    2. Macro(blkvm-setifempty,)                   [pbx_config]
                    3. GotoIf($["${GOSUB_RETVAL}" = "TRUE"]?skipov) [pbx_config]
                    4. Macro(blkvm-set,reset)                     [pbx_config]
                    5. Set(__NODEST=)                             [pbx_config]
     [skipov]       6. Set(RRNODEST=${NODEST})                    [pbx_config]
     [skipvmblk]    7. Set(__NODEST=${EXTEN})                     [pbx_config]
                    8. GosubIf($[${DB_EXISTS(RINGGROUP/601/changecid)} = 1 & "${DB(RINGGROUP/601/changecid)}" != "default" & "${DB(RINGGROUP/601/changecid)}" != ""]?sub-rgsetcid,s,1()) [pbx_config]
                    9. Macro(prepend-cid,test-group2)             [pbx_config]
                    10. Gosub(sub-record-check,s,1(rg,601,always)) [pbx_config]
                    11. Set(RingGroupMethod=ringall)              [pbx_config]
     [DIALGRP]      12. Macro(dial,80,m(default)${REPLACE(DIAL_OPTIONS,r)},101-102) [pbx_config]
                    13. Gosub(sub-record-cancel,s,1())            [pbx_config]
                    14. Set(RingGroupMethod=)                     [pbx_config]
                    15. GotoIf($["foo${RRNODEST}" != "foo"]?nodest) [pbx_config]
                    16. Set(__NODEST=)                            [pbx_config]
                    17. Macro(blkvm-clr,)                         [pbx_config]
                    18. Goto(ext-group,601,1)                     [pbx_config]
     [nodest]       19. Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST}) [pbx_config]
  'h' =>            1. Macro(hangupcall,)                         [pbx_config]
  Include =>        'ext-group-custom'                            [pbx_config]

-= 3 extensions (38 priorities) in 1 context. =-

[Drill]

Судя по диалплану, звонок должен проходить без проблем

Еще:
dialplan show 134628@from-trunk

должно совпадать с
dialplan show 134628@from-pstn

Если звонок отбивается, попробуйте убрать любые два пира из группы, оставьте какой-нибудь один, например 102. И попробуйте позвонить. Повторите с другими номерами.

[Назарий Луговой]

spoiler

issabel*CLI> dialplan show 134628@from-trunk
[ Included context 'ext-did-0002' created by 'pbx_config' ]
  '134628' =>       1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                    3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                    4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                    5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                    6. Set(__MOHCLASS=default)                    [pbx_config]
                    7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                    8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
     [dest-ext]     9. Goto(from-did-direct,102,1)                [pbx_config]

[ Included context 'ext-did-catchall' created by 'pbx_config' ]
  '_.' =>           1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Noop(Received an unknown call with DID set to ${EXTEN}) [pbx_config]
                    3. Goto(s,a2)                                 [pbx_config]

-= 2 extensions (12 priorities) in 2 contexts. =-

issabel*CLI> dialplan show 134628@from-pstn
[ Included context 'ext-did-0002' created by 'pbx_config' ]
  '134628' =>       1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Gosub(app-blacklist-check,s,1())           [pbx_config]
                    3. Set(CDR(did)=${FROM_DID})                  [pbx_config]
                    4. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                    5. Set(CHANNEL(musicclass)=default)           [pbx_config]
                    6. Set(__MOHCLASS=default)                    [pbx_config]
                    7. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                    8. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
     [dest-ext]     9. Goto(from-did-direct,102,1)                [pbx_config]

[ Included context 'ext-did-catchall' created by 'pbx_config' ]
  '_.' =>           1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                    2. Noop(Received an unknown call with DID set to ${EXTEN}) [pbx_config]
                    3. Goto(s,a2)                                 [pbx_config]

-= 2 extensions (12 priorities) in 2 contexts. =-

Пробовал и с одним, и с разными номерами. Результат тот-же.
При том, если я указываю номер, вместо положить трубку, в последнем пункте "Назначение, если никто не дозвонился" то звонок сразу идет на него, как будто не отрабатывает Ring time

[Drill]

1. Назначение, если никто не дозвонился, верните обратно.
2. Измените настройки логирования как на картинке:


3. Запустите sngrep
4. Сделайте звонок.
5. В sngrep отметьте пробелом все инвайты и сохраните дамп в файл file.pcap (сохранить выделенное - F2)
6. Сбросьте мне сохраненный file.pcap и логфайл /var/log/asterisk/full

[Drill]

в телеграме есть или в вайбере?

[Назарий Луговой]

viber [phone]
full и invite
https://drive.google.com/file/d/1ZalSZnXzvlaajG7MZ...
invite был всего один

[Drill]

В логах:
Failed to execute '/var/lib/asterisk/agi-bin/dialparties.agi': Permission denied

Не отрабатывает agi-скрипт!!!

Проверьте права на каталоги /var/lib/asterisk/* Владелец и группа - asterisk:asterisk (установить )

И на скрипты в этом каталоге /var/lib/asterisk/agi-bin/.
Владелец и группа у скриптов- asterisk:asterisk, если root:root, то исполнение разрешено для всех (chmod 755)

[Drill]

Должно быть:

spoiler

ls -l /var/lib/asterisk/agi-bin
-rwxr-xr-- 1 asterisk asterisk  1485 Янв 22 10:16 checksound.agi
-rwxr-xr-- 1 asterisk asterisk 31868 Янв 22 10:16 dialparties.agi
-rwxr-xr-- 1 asterisk asterisk 12694 Янв 22 10:16 directory
-rwxr-xr-- 1 asterisk asterisk  4792 Янв 22 10:16 enumlookup.agi
-rwxr-xr-x 1 root     root       703 Сен  7 21:58 festival-script.pl
-rwxr-xr-x 1 root     root       784 Сен  7 21:58 festival-weather-script.pl
-rwxr-xr-- 1 asterisk asterisk   778 Янв 22 10:16 fixlocalprefix
-rwxr-xr-x 1 root     root      6366 Сен  7 21:58 hangup.agi
-rwxr-xr-x 1 root     root     28770 Сен  7 21:58 imap.agi
-rwxr-xr-x 1 root     root      7438 Сен  7 21:58 intervenir.agi
-rwxr-xr-- 1 asterisk asterisk  1263 Янв 22 10:16 list-item-remove.php
-rwxr-xr-x 1 root     root      3659 Сен  7 21:58 nv-weather.php
-rwxr-xr-- 1 asterisk asterisk  6438 Янв 22 10:16 page.agi
-rwxr-xr-- 1 asterisk asterisk  1927 Янв 22 10:16 parkfetch.agi
-rwxr-xr-- 1 asterisk asterisk  2049 Янв 22 10:16 parkuser.php
-rwxr-xr-- 1 asterisk asterisk  9835 Янв 22 10:16 pbdirectory
-rwxrwxr-x 1 asterisk asterisk 31158 Янв 22 10:06 phpagi-asmanager.php
-rwxrwxr-x 1 asterisk asterisk 65353 Янв 22 10:06 phpagi.php
-rwxr-xr-- 1 asterisk asterisk  7255 Янв 22 10:16 queue_devstate.agi
-rwxr-xr-- 1 asterisk asterisk 20498 Янв 22 10:16 sql.php
-rwxr-xr-- 1 asterisk asterisk 22959 Янв 22 10:16 user_login_out.agi
-rwxr-xr-x 1 root     root      3991 Сен  7 21:58 wakeconfirm.agi
-rwxr-xr-x 1 root     root     21372 Сен  7 21:58 wakeup.php
-rwxr-xr-x 1 root     root      1248 Сен  7 21:58 weather.agi

[Назарий Луговой]

Drill, Вы профи!
/var/lib/asterisk/ был asterisk:asterisk
а вот /var/lib/asterisk/agi-bin/ зеркальная противоположность Вашему логу

spoiler

ls -l /var/lib/asterisk/agi-bin
итого 340
-rwxr-xr--  1 root     root      1485 Янв 25 22:27 checksound.agi
-rwxr-xr--  1 root     root     31868 Янв 25 22:27 dialparties.agi
-rwxr-xr--  1 root     root     12694 Янв 25 22:27 directory
-rwxr-xr--  1 root     root      4792 Янв 25 22:27 enumlookup.agi
-rwxr-xr-x. 1 asterisk asterisk   703 Сен  7 21:58 festival-script.pl
-rwxr-xr-x. 1 asterisk asterisk   784 Сен  7 21:58 festival-weather-script.pl
-rwxr-xr--  1 root     root       778 Янв 25 22:27 fixlocalprefix
-rwxr-xr-x. 1 asterisk asterisk  6366 Сен  7 21:58 hangup.agi
-rwxr-xr-x. 1 asterisk asterisk 28770 Сен  7 21:58 imap.agi
-rwxr-xr-x. 1 asterisk asterisk  7438 Сен  7 21:58 intervenir.agi
-rwxr-xr--  1 root     root      1263 Янв 25 22:27 list-item-remove.php
-rwxr-xr-x. 1 asterisk asterisk  3659 Сен  7 21:58 nv-weather.php
-rwxr-xr--  1 root     root      6438 Янв 25 22:27 page.agi
-rwxr-xr--  1 root     root      1927 Янв 25 22:27 parkfetch.agi
-rwxr-xr--  1 root     root      2049 Янв 25 22:27 parkuser.php
-rwxr-xr--  1 root     root      9835 Янв 25 22:27 pbdirectory
-rwxr-xr-x. 1 asterisk asterisk 31158 Сен 20 18:03 phpagi-asmanager.php
-rwxr-xr-x. 1 asterisk asterisk 65353 Сен 20 18:03 phpagi.php
-rwxr-xr--  1 root     root      7255 Янв 25 22:27 queue_devstate.agi
-rwxr-xr--  1 root     root     20498 Янв 25 22:27 sql.php
-rwxr-xr--  1 root     root     22959 Янв 25 22:27 user_login_out.agi
-rwxr-xr-x. 1 asterisk asterisk  3991 Сен  7 21:58 wakeconfirm.agi
-rwxr-xr-x. 1 asterisk asterisk 21372 Сен  7 21:58 wakeup.php
-rwxr-xr-x. 1 asterisk asterisk  1248 Сен  7 21:58 weather.agi

я выставил всем root:root и chmod 755
После этого все заработало!

[Drill]

Хорошо. Я перенесу свой коммент в ответ, а вы отметьте как решение.

Answer 1

[Drill]

В логах:
Failed to execute '/var/lib/asterisk/agi-bin/dialparties.agi': Permission denied

Не отрабатывает agi-скрипт. Надо дать астериску права на исполнение скриптов в каталоге /var/lib/asterisk/agi-bin/

Comments

[Назарий Луговой]

Обнаружился нюанс, с группой номеров. Не стал создавать новую тему, пишу здесь.
Нюанс в функции "Назначение, если никто не ответил".
Я поставил номер с автоответчиком на нее, но к сожалению переадресация на этот номер происходит если лишь если все номера в группе отклонили входящий звонок. А вот как раз если никто не поднял трубку, то перенаправления на номер нет.