Пропала слышимость вызываемого абонента.

**darkday** » 08 авг 2017, 19:29

Добрый день. Понимаю что тема баян, схему по проверке слышимости помню, хочу уточнить некоторые моменты:

1. Настройки станций не менял. Оборудование не менял. Все станции имеют белые внешние IP, никакого NAT
2. Заведены два провайдера VOIP, проблема только при вызове абонентов одного из них.
3. Рабочая схема:
VOIP Провайдер (IP 193.200.32.23) <-->(Белый IP) ELASTIX (2.4 -Asterisk 1.8.2) (IP 192.168.1.116) <--> VOIP Шлюз за которым телефоны( IP 192.168.1.121)

Добавил для теста новую станцию (на которую вынес один из проблемных номеров) и новый шлюз:
VOIP Провайдер (IP 193.200.32.23) <-->(Белый IP2) FreePBX(Asterisk 13.17.0 ) (IP 192.168.1.117) <--> VOIP Шлюз за которым телефоны( IP 192.168.1.201)

Проверил прохождение rtp:

[Показать] Спойлер:

Станция 1:
192.168.1.121 - адрес VOIP шлюза
193.200.32.23 - адрес сервера оператора
Sent RTP packet to 192.168.1.121:9069 (type 08, seq 002406, ts 1024452960, len 000160)
Got RTP packet from 192.168.1.121:9069 (type 08, seq 020872, ts 2858056728, len 000160)
Sent RTP packet to 193.200.32.23:12798 (type 08, seq 034037, ts 2858056728, len 000160)
Got RTP packet from 193.200.32.23:12798 (type 08, seq 009104, ts 1024453120, len 000160)
Sent RTP packet to 192.168.1.121:9069 (type 08, seq 002407, ts 1024453120, len 000160)
Got RTP packet from 192.168.1.121:9069 (type 08, seq 020873, ts 2858056888, len 000160)
Sent RTP packet to 193.200.32.23:12798 (type 08, seq 034038, ts 2858056888, len 000160)
Got RTP packet from 193.200.32.23:12798 (type 08, seq 009105, ts 1024453280, len 000160)

Станция 2:
192.168.1.201 - адрес VOIP шлюза
193.200.32.23 - адрес сервера оператора

Sent RTP packet to 192.168.1.201:16436 (type 00, seq 032013, ts 3624758000, len 000160)
Got RTP packet from 192.168.1.201:16436 (type 00, seq 014258, ts 400065497, len 000160)
Sent RTP packet to 193.200.32.23:12972 (type 00, seq 000508, ts 400065496, len 000160)
Got RTP packet from 193.200.32.23:12972 (type 00, seq 011874, ts 3624758160, len 000160)
Sent RTP packet to 192.168.1.201:16436 (type 00, seq 032014, ts 3624758160, len 000160)
Got RTP packet from 192.168.1.201:16436 (type 00, seq 014259, ts 400065657, len 000160)
Sent RTP packet to 193.200.32.23:12972 (type 00, seq 000509, ts 400065656, len 000160)
Got RTP packet from 193.200.32.23:12972 (type 00, seq 011875, ts 3624758320, len 000160)

1.Правильно ли я понимаю, что здесь у нас и одна и вторая станция нормально обмениваются rtp-пакетами в обе стороны (Сервер провайдера-шлюз VOIP-FXS)?
Если нет, то где почитать про логику работы.
Если да, то куда копать дальше? Согласование кодеков?
2. Тип пакетов отличается (00) и (08). Я так понимаю разные кодеки (U) и (A). Это может быть истоник проблемы?
Спасибо.

**april22** » 08 авг 2017, 23:31

Лучше увидеть инвайт

**ded** » 09 авг 2017, 00:25

ТС, у вас должно быть два плеча в таком разговоре, и две цепочки RTP
1) 193.200.32.23) <-->(Белый IP)
2) (IP 192.168.1.116) <--> ( IP 192.168.1.121)

ну и в другом случае так же. Ни для каких участников этих соединений нет НАТа (nat=no) и никаких externip= & localnet=

Всё что не так - дефект.

darkday писал(а):2. Тип пакетов отличается (00) и (08). Я так понимаю разные кодеки (U) и (A). Это может быть истоник проблемы?

Нет. Про транскодинг слышали? Он автоматически делается на SIP.

**virus_net** » 09 авг 2017, 08:09

april22 писал(а):Лучше увидеть инвайт

+1
а совсем хорошо сделать дамп обоих плечей, о которых выше написал ded, затем сделать анализ дампа в wireshark

**darkday** » 09 авг 2017, 13:00

Добрый день. Сейчас не в офисе, проверяю пока "экспериментальную" вторую станцию.
Звонил с мобильного на нее, под спойлером лог sip debug:
externalip и localnet не выставлены
nat=never
До дампа пока не добрался.

[Показать] Спойлер:

<--- SIP read from UDP:193.200.32.23:5060 --->
INVITE sip:7320732@62.64.х.х:5160 SIP/2.0
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK2f01e25d;rport
Max-Forwards: 70
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>
Contact: <sip:067909090@193.200.32.23:5060>
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 102 INVITE
User-Agent: Telemost VoIP Gate
Date: Wed, 09 Aug 2017 08:32:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245

v=0
o=root 691373001 691373001 IN IP4 193.200.32.23
s=Asterisk PBX certified/11.6-cert13
c=IN IP4 193.200.32.23
t=0 0
m=audio 18198 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 193.200.32.23:5060 (no NAT)
Sending to 193.200.32.23:5060 (no NAT)
Using INVITE request as basis request - 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
Found peer '7320732' for '067909090' from 193.200.32.23:5060

<--- Reliably Transmitting (no NAT) to 193.200.32.23:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK2f01e25d;received=193.200.32.23;rport=5060
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>;tag=as6556ec5f
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 102 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2dcadebd"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:193.200.32.23:5060 --->
ACK sip:7320732@62.64.х.х:5160 SIP/2.0
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK2f01e25d;rport
Max-Forwards: 70
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>;tag=as6556ec5f
Contact: <sip:067909090@193.200.32.23:5060>
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 102 ACK
User-Agent: Telemost VoIP Gate
Content-Length: 0

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

<--- SIP read from UDP:193.200.32.23:5060 --->
INVITE sip:7320732@62.64.х.х:5160 SIP/2.0
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK067bd17d;rport
Max-Forwards: 70
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>
Contact: <sip:067909090@193.200.32.23:5060>
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 103 INVITE
User-Agent: Telemost VoIP Gate
Authorization: Digest username="7320732", realm="asterisk", algorithm=MD5, uri="sip:7320732@62.64.х.х:5160", nonce="2dcadebd", response="96d21fab935ddc5d26e17d775883e974"
Date: Wed, 09 Aug 2017 08:32:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245

v=0
o=root 691373001 691373002 IN IP4 193.200.32.23
s=Asterisk PBX certified/11.6-cert13
c=IN IP4 193.200.32.23
t=0 0
m=audio 18198 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Sending to 193.200.32.23:5060 (no NAT)
Using INVITE request as basis request - 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
Found peer '7320732' for '067909090' from 193.200.32.23:5060
[2017-08-09 11:36:58] ERROR[2788][C-0000000d]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("apollo.ventaltd.com.ua", "(null)", ...): Name or service not known
[2017-08-09 11:36:58] WARNING[2788][C-0000000d]: acl.c:800 resolve_first: Unable to lookup 'apollo.ventaltd.com.ua'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 193.200.32.23:18198
Looking for 7320732 in from-trunk-sip-7320732 (domain 62.64.х.х)
sip_route_dump: route/path hop: <sip:067909090@193.200.32.23:5060>

<--- Transmitting (no NAT) to 193.200.32.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK067bd17d;received=193.200.32.23;rport=5060
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 103 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7320732@62.64.х.х:5160>
Content-Length: 0

<------------>

<--- SIP read from UDP:193.200.32.23:5060 --->
INVITE sip:7320732@62.64.х.х:5160 SIP/2.0
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK067bd17d;rport
Max-Forwards: 70
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>
Contact: <sip:067909090@193.200.32.23:5060>
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 103 INVITE
User-Agent: Telemost VoIP Gate
Authorization: Digest username="7320732", realm="asterisk", algorithm=MD5, uri="sip:7320732@62.64.х.х:5160", nonce="2dcadebd", response="96d21fab935ddc5d26e17d775883e974"
Date: Wed, 09 Aug 2017 08:32:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245

v=0
o=root 691373001 691373002 IN IP4 193.200.32.23
s=Asterisk PBX certified/11.6-cert13
c=IN IP4 193.200.32.23
t=0 0
m=audio 18198 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 11 lines) ---
Ignoring this INVITE request

<--- Transmitting (no NAT) to 193.200.32.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK067bd17d;received=193.200.32.23;rport=5060
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 103 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7320732@62.64.х.х:5160>
Content-Length: 0

<------------>
-- Executing [7320732@from-trunk-sip-7320732:1] Set("SIP/7320732-00000019", "GROUP()=OUT_1") in new stack
-- Executing [7320732@from-trunk-sip-7320732:2] Goto("SIP/7320732-00000019", "from-trunk,7320732,1") in new stack
-- Goto (from-trunk,7320732,1)
-- Executing [7320732@from-trunk:1] Set("SIP/7320732-00000019", "__DIRECTION=INBOUND") in new stack
-- Executing [7320732@from-trunk:2] Gosub("SIP/7320732-00000019", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/7320732-00000019", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/7320732-00000019", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/7320732-00000019", "") in new stack
-- Executing [7320732@from-trunk:3] Set("SIP/7320732-00000019", "__FROM_DID=7320732") in new stack
-- Executing [7320732@from-trunk:4] Set("SIP/7320732-00000019", "CDR(did)=7320732") in new stack
-- Executing [7320732@from-trunk:5] ExecIf("SIP/7320732-00000019", "0 ?Set(CALLERID(name)=067909090)") in new stack
-- Executing [7320732@from-trunk:6] Set("SIP/7320732-00000019", "__MOHCLASS=") in new stack
-- Executing [7320732@from-trunk:7] Set("SIP/7320732-00000019", "__REVERSAL_REJECT=FALSE") in new stack
-- Executing [7320732@from-trunk:8] GotoIf("SIP/7320732-00000019", "1?post-reverse-charge") in new stack
-- Goto (from-trunk,7320732,10)
-- Executing [7320732@from-trunk:10] NoOp("SIP/7320732-00000019", "") in new stack
-- Executing [7320732@from-trunk:11] Set("SIP/7320732-00000019", "__CALLINGNAMEPRES_SV=allowed_not_screened") in new stack
-- Executing [7320732@from-trunk:12] Set("SIP/7320732-00000019", "__CALLINGNUMPRES_SV=allowed_not_screened") in new stack
-- Executing [7320732@from-trunk:13] Set("SIP/7320732-00000019", "CALLERID(name-pres)=allowed_not_screened") in new stack
-- Executing [7320732@from-trunk:14] Set("SIP/7320732-00000019", "CALLERID(num-pres)=allowed_not_screened") in new stack
-- Executing [7320732@from-trunk:15] NoOp("SIP/7320732-00000019", "CallerID Entry Point") in new stack
-- Executing [7320732@from-trunk:16] Set("SIP/7320732-00000019", "__CRM_DIRECTION=INBOUND") in new stack
-- Executing [7320732@from-trunk:17] Set("SIP/7320732-00000019", "__CRM_SOURCE=067909090") in new stack
-- Executing [7320732@from-trunk:18] Set("SIP/7320732-00000019", "__CRM_LINKEDID=1502267818.25") in new stack
-- Executing [7320732@from-trunk:19] ExecIf("SIP/7320732-00000019", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [7320732@from-trunk:20] Goto("SIP/7320732-00000019", "from-did-direct,767,1") in new stack
-- Goto (from-did-direct,767,1)
-- Executing [767@from-did-direct:1] GotoIf("SIP/7320732-00000019", "1?ext-local,767,1:followme-check,767,1") in new stack
-- Goto (ext-local,767,1)
-- Executing [767@ext-local:1] Set("SIP/7320732-00000019", "__RINGTIMER=15") in new stack
-- Executing [767@ext-local:2] Macro("SIP/7320732-00000019", "exten-vm,novm,767,0,0,0") in new stack
-- Executing [s@macro-exten-vm:1] Macro("SIP/7320732-00000019", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/7320732-00000019", "TOUCH_MONITOR=1502267818.25") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/7320732-00000019", "AMPUSER=067909090") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/7320732-00000019", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/7320732-00000019", "1?Set(REALCALLERIDNUM=067909090)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/7320732-00000019", "AMPUSER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/7320732-00000019", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/7320732-00000019", "AMPUSERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/7320732-00000019", "1?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/7320732-00000019", "0?continue") in new stack
-- Executing [s@macro-user-callerid:16] ExecIf("SIP/7320732-00000019", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
-- Executing [s@macro-user-callerid:17] Set("SIP/7320732-00000019", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:18] GotoIf("SIP/7320732-00000019", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("SIP/7320732-00000019", "CALLERID(number)=067909090") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/7320732-00000019", "CALLERID(name)=067909090") in new stack
-- Executing [s@macro-user-callerid:31] GotoIf("SIP/7320732-00000019", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/7320732-00000019", "CDR(cnam)=067909090") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/7320732-00000019", "CDR(cnum)=067909090") in new stack
-- Executing [s@macro-user-callerid:34] Set("SIP/7320732-00000019", "CHANNEL(language)=ru") in new stack
-- Executing [s@macro-exten-vm:2] Set("SIP/7320732-00000019", "RingGroupMethod=none") in new stack
-- Executing [s@macro-exten-vm:3] Set("SIP/7320732-00000019", "__EXTTOCALL=767") in new stack
-- Executing [s@macro-exten-vm:4] Set("SIP/7320732-00000019", "__PICKUPMARK=767") in new stack
-- Executing [s@macro-exten-vm:5] Set("SIP/7320732-00000019", "RT=") in new stack
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [s@macro-exten-vm:6] ExecIf("SIP/7320732-00000019", "0?Macro(vm,novm,DIRECTDIAL,)") in new stack
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [s@macro-exten-vm:7] ExecIf("SIP/7320732-00000019", "0?MacroExit()") in new stack
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [s@macro-exten-vm:8] ExecIf("SIP/7320732-00000019", "0?Gosub(ext-intercom,*80767,1())") in new stack
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [s@macro-exten-vm:9] ExecIf("SIP/7320732-00000019", "0?MacroExit()") in new stack
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2017-08-09 11:36:58] WARNING[744][C-0000000d]: pbx_functions.c:460 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/767/dial'?
-- Executing [s@macro-exten-vm:10] ExecIf("SIP/7320732-00000019", "0?ChanSpy(SIP/767,q)") in new stack
[2017-08-09 11:36:58] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
[2017-08-09 11:36:58] WARNING[744][C-0000000d]: pbx_functions.c:460 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/767/dial'?
[2017-08-09 11:36:59] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [s@macro-exten-vm:11] ExecIf("SIP/7320732-00000019", "0?MacroExit()") in new stack
[2017-08-09 11:36:59] ERROR[744][C-0000000d]: res_pjsip_header_funcs.c:461 func_read_header: This function requires a PJSIP channel.
-- Executing [s@macro-exten-vm:12] Gosub("SIP/7320732-00000019", "sub-record-check,s,1(exten,767,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/7320732-00000019", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/7320732-00000019", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/7320732-00000019", "NOW=1502267819") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/7320732-00000019", "__DAY=09") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/7320732-00000019", "__MONTH=08") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/7320732-00000019", "__YEAR=2017") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/7320732-00000019", "__TIMESTR=20170809-113659") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/7320732-00000019", "__FROMEXTEN=067909090") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/7320732-00000019", "__MON_FMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/7320732-00000019", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/7320732-00000019", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/7320732-00000019", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/7320732-00000019", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/7320732-00000019", "5?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/7320732-00000019", "1?sub-record-check,exten,1") in new stack
-- Goto (sub-record-check,exten,1)
-- Executing [exten@sub-record-check:1] NoOp("SIP/7320732-00000019", "Exten Recording Check between 067909090 and 767") in new stack
-- Executing [exten@sub-record-check:2] Set("SIP/7320732-00000019", "CALLTYPE=external") in new stack
-- Executing [exten@sub-record-check:3] ExecIf("SIP/7320732-00000019", "0?Set(CALLTYPE=)") in new stack
-- Executing [exten@sub-record-check:4] Set("SIP/7320732-00000019", "CALLEE=dontcare") in new stack
-- Executing [exten@sub-record-check:5] ExecIf("SIP/7320732-00000019", "0?Set(CALLEE=dontcare)") in new stack
-- Executing [exten@sub-record-check:6] GotoIf("SIP/7320732-00000019", "1?callee") in new stack
-- Goto (sub-record-check,exten,11)
-- Executing [exten@sub-record-check:11] Gosub("SIP/7320732-00000019", "recordcheck,1(dontcare,external,767)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/7320732-00000019", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/7320732-00000019", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("SIP/7320732-00000019", "") in new stack
-- Executing [exten@sub-record-check:12] Return("SIP/7320732-00000019", "") in new stack
-- Executing [s@macro-exten-vm:13] GotoIf("SIP/7320732-00000019", "1?macrodial") in new stack
-- Goto (macro-exten-vm,s,19)
-- Executing [s@macro-exten-vm:19] GosubIf("SIP/7320732-00000019", "0?clrheader,1()") in new stack
-- Executing [s@macro-exten-vm:20] Macro("SIP/7320732-00000019", "dial-one,,Ttr,767") in new stack
-- Executing [s@macro-dial-one:1] Set("SIP/7320732-00000019", "DEXTEN=767") in new stack
-- Executing [s@macro-dial-one:2] Set("SIP/7320732-00000019", "__CRM_SOURCE=067909090") in new stack
-- Executing [s@macro-dial-one:3] ExecIf("SIP/7320732-00000019", "0?Set(EXTTOCALL=767)") in new stack
-- Executing [s@macro-dial-one:4] Set("SIP/7320732-00000019", "DIALSTATUS_CW=") in new stack
-- Executing [s@macro-dial-one:5] GosubIf("SIP/7320732-00000019", "0?screen,1()") in new stack
-- Executing [s@macro-dial-one:6] GosubIf("SIP/7320732-00000019", "0?cf,1()") in new stack
-- Executing [s@macro-dial-one:7] GotoIf("SIP/7320732-00000019", "1?skip1") in new stack
-- Goto (macro-dial-one,s,10)
-- Executing [s@macro-dial-one:10] GotoIf("SIP/7320732-00000019", "0?nodial") in new stack
-- Executing [s@macro-dial-one:11] GotoIf("SIP/7320732-00000019", "0?continue") in new stack
-- Executing [s@macro-dial-one:12] Set("SIP/7320732-00000019", "EXTHASCW=ENABLED") in new stack
-- Executing [s@macro-dial-one:13] GotoIf("SIP/7320732-00000019", "0?next1:cwinusebusy") in new stack
-- Goto (macro-dial-one,s,25)
-- Executing [s@macro-dial-one:25] GotoIf("SIP/7320732-00000019", "0?next3:continue") in new stack
-- Goto (macro-dial-one,s,27)
-- Executing [s@macro-dial-one:27] GotoIf("SIP/7320732-00000019", "0?nodial") in new stack
-- Executing [s@macro-dial-one:28] GosubIf("SIP/7320732-00000019", "1?dstring,1():dlocal,1()") in new stack
-- Executing [dstring@macro-dial-one:1] Set("SIP/7320732-00000019", "DSTRING=") in new stack
-- Executing [dstring@macro-dial-one:2] Set("SIP/7320732-00000019", "DEVICES=767") in new stack
-- Executing [dstring@macro-dial-one:3] ExecIf("SIP/7320732-00000019", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:4] ExecIf("SIP/7320732-00000019", "0?Set(DEVICES=67)") in new stack
-- Executing [dstring@macro-dial-one:5] Set("SIP/7320732-00000019", "LOOPCNT=1") in new stack
-- Executing [dstring@macro-dial-one:6] Set("SIP/7320732-00000019", "ITER=1") in new stack
-- Executing [dstring@macro-dial-one:7] Set("SIP/7320732-00000019", "THISDIAL=SIP/767") in new stack
-- Executing [dstring@macro-dial-one:8] GosubIf("SIP/7320732-00000019", "1?zap2dahdi,1()") in new stack
-- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/7320732-00000019", "0?Return()") in new stack
-- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/7320732-00000019", "NEWDIAL=") in new stack
-- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/7320732-00000019", "LOOPCNT2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/7320732-00000019", "ITER2=1") in new stack
-- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/7320732-00000019", "THISPART2=SIP/767") in new stack
-- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/7320732-00000019", "0?Set(THISPART2=DAHDI/767)") in new stack
-- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/7320732-00000019", "NEWDIAL=SIP/767&") in new stack
-- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/7320732-00000019", "ITER2=2") in new stack
-- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/7320732-00000019", "0?begin2") in new stack
-- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/7320732-00000019", "THISDIAL=SIP/767") in new stack
-- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/7320732-00000019", "") in new stack
-- Executing [dstring@macro-dial-one:9] GotoIf("SIP/7320732-00000019", "1?docheck") in new stack
-- Goto (macro-dial-one,dstring,14)
-- Executing [dstring@macro-dial-one:14] GotoIf("SIP/7320732-00000019", "0?skipset") in new stack
-- Executing [dstring@macro-dial-one:15] Set("SIP/7320732-00000019", "DSTRING=SIP/767&") in new stack
-- Executing [dstring@macro-dial-one:16] Set("SIP/7320732-00000019", "ITER=2") in new stack
-- Executing [dstring@macro-dial-one:17] GotoIf("SIP/7320732-00000019", "0?begin") in new stack
-- Executing [dstring@macro-dial-one:18] ExecIf("SIP/7320732-00000019", "0?Return()") in new stack
-- Executing [dstring@macro-dial-one:19] Set("SIP/7320732-00000019", "DSTRING=SIP/767") in new stack
-- Executing [dstring@macro-dial-one:20] Return("SIP/7320732-00000019", "") in new stack
-- Executing [s@macro-dial-one:29] GotoIf("SIP/7320732-00000019", "0?nodial") in new stack
-- Executing [s@macro-dial-one:30] GotoIf("SIP/7320732-00000019", "0?skiptrace") in new stack
-- Executing [s@macro-dial-one:31] GosubIf("SIP/7320732-00000019", "1?ctset,1():ctclear,1()") in new stack
-- Executing [ctset@macro-dial-one:1] Set("SIP/7320732-00000019", "DB(CALLTRACE/767)=067909090") in new stack
-- Executing [ctset@macro-dial-one:2] Return("SIP/7320732-00000019", "") in new stack
-- Executing [s@macro-dial-one:32] Set("SIP/7320732-00000019", "D_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dial-one:33] NoOp("SIP/7320732-00000019", "Blind Transfer: , Attended Transfer: , User: , Alert Info: ") in new stack
-- Executing [s@macro-dial-one:34] ExecIf("SIP/7320732-00000019", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:35] ExecIf("SIP/7320732-00000019", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:36] ExecIf("SIP/7320732-00000019", "0?Set(ALERT_INFO=)") in new stack
-- Executing [s@macro-dial-one:37] ExecIf("SIP/7320732-00000019", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:38] ExecIf("SIP/7320732-00000019", "0?Set(ALERT_INFO=Normal;volume=)") in new stack
-- Executing [s@macro-dial-one:39] GosubIf("SIP/7320732-00000019", "0?func-set-sipheader,s,1(Alert-Info,)") in new stack
-- Executing [s@macro-dial-one:40] ExecIf("SIP/7320732-00000019", "0?Set(CHANNEL(musicclass)=)") in new stack
-- Executing [s@macro-dial-one:41] GosubIf("SIP/7320732-00000019", "0?qwait,1()") in new stack
-- Executing [s@macro-dial-one:42] Set("SIP/7320732-00000019", "__CWIGNORE=") in new stack
-- Executing [s@macro-dial-one:43] Set("SIP/7320732-00000019", "__KEEPCID=TRUE") in new stack
-- Executing [s@macro-dial-one:44] GotoIf("SIP/7320732-00000019", "0?usegoto,1") in new stack
-- Executing [s@macro-dial-one:45] GotoIf("SIP/7320732-00000019", "1?godial") in new stack
-- Goto (macro-dial-one,s,50)
-- Executing [s@macro-dial-one:50] Macro("SIP/7320732-00000019", "dialout-one-predial-hook,") in new stack
-- Executing [s@macro-dialout-one-predial-hook:1] MacroExit("SIP/7320732-00000019", "") in new stack
-- Executing [s@macro-dial-one:51] ExecIf("SIP/7320732-00000019", "1?Set(D_OPTIONS=trI)") in new stack
-- Executing [s@macro-dial-one:52] Dial("SIP/7320732-00000019", "SIP/767,,trIb(func-apply-sipheaders^s^1)") in new stack
[2017-08-09 11:36:59] ERROR[744][C-0000000d]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("apollo.ventaltd.com.ua", "(null)", ...): Name or service not known
[2017-08-09 11:36:59] WARNING[744][C-0000000d]: acl.c:800 resolve_first: Unable to lookup 'apollo.ventaltd.com.ua'
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- SIP/767-0000001a Internal Gosub(func-apply-sipheaders,s,1) start
-- Executing [s@func-apply-sipheaders:1] ExecIf("SIP/767-0000001a", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
-- Executing [s@func-apply-sipheaders:2] NoOp("SIP/767-0000001a", "Applying SIP Headers to channel") in new stack
-- Executing [s@func-apply-sipheaders:3] Set("SIP/767-0000001a", "SIPHEADERKEYS=") in new stack
-- Executing [s@func-apply-sipheaders:4] While("SIP/767-0000001a", "0") in new stack
-- Jumping to priority 8
-- Executing [s@func-apply-sipheaders:9] Return("SIP/767-0000001a", "") in new stack
== Spawn extension (from-internal, 767, 1) exited non-zero on 'SIP/767-0000001a'
-- SIP/767-0000001a Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
Audio is at 16642
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.201:5060:
INVITE sip:767@192.168.1.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.117:5160;branch=z9hG4bK321aa49c
Max-Forwards: 70
From: "067909090" <sip:067909090@192.168.1.117:5160>;tag=as1840ba23
To: <sip:767@192.168.1.201:5060>
Contact: <sip:067909090@192.168.1.117:5160>
Call-ID: 63226cc6213902771545eaa624544442@192.168.1.117:5160
CSeq: 102 INVITE
User-Agent: FPBX-13.0.192.16(13.17.0)
Date: Wed, 09 Aug 2017 08:36:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "067909090" <sip:067909090@192.168.1.117>
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 451705511 451705511 IN IP4 192.168.1.117
s=Asterisk PBX 13.17.0
c=IN IP4 192.168.1.117
t=0 0
m=audio 16642 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
-- Called SIP/767

<--- Transmitting (no NAT) to 193.200.32.23:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK067bd17d;received=193.200.32.23;rport=5060
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>;tag=as7abf054d
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 103 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7320732@62.64.х.х:5160>
Content-Length: 0

<------------>
-- Connected line update to SIP/7320732-00000019 prevented.

<--- SIP read from UDP:192.168.1.201:5060 --->
SIP/2.0 100 Trying
To: <sip:767@192.168.1.201:5060>
From: "067909090" <sip:067909090@192.168.1.117:5160>;tag=as1840ba23
Call-ID: 63226cc6213902771545eaa624544442@192.168.1.117:5160
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.117:5160;branch=z9hG4bK321aa49c
Server: Linksys/SPA8000-6.1.3
Allow-Events: talk, hold, conference
Content-Length: 0

sip_route_dump: route/path hop: <sip:767@192.168.1.201:5060>
-- Connected line update to SIP/7320732-00000019 prevented.
-- SIP/767-0000001a is ringing

<--- Transmitting (no NAT) to 193.200.32.23:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK067bd17d;received=193.200.32.23;rport=5060
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>;tag=as7abf054d
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 103 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7320732@62.64.х.х:5160>
Content-Length: 0

<------------>

<--- SIP read from UDP:192.168.1.201:5060 --->
SIP/2.0 200 OK
To: <sip:767@192.168.1.201:5060>;tag=25b6d3472c85c81ai0
From: "067909090" <sip:067909090@192.168.1.117:5160>;tag=as1840ba23
Call-ID: 63226cc6213902771545eaa624544442@192.168.1.117:5160
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.117:5160;branch=z9hG4bK321aa49c
Contact: 767 <sip:767@192.168.1.201:5060>
Server: Linksys/SPA8000-6.1.3
Remote-Party-ID: 767 <sip:767@192.168.1.117:5160>;screen=yes;party=called
Allow-Events: talk, hold, conference
Content-Length: 257
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 6957110 6957110 IN IP4 192.168.1.201
s=-
c=IN IP4 192.168.1.201
t=0 0
m=audio 16478 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.201:16478
sip_route_dump: route/path hop: <sip:767@192.168.1.201:5060>
set_destination: Parsing <sip:767@192.168.1.201:5060> for address/port to send to
set_destination: set destination to 192.168.1.201:5060
Transmitting (no NAT) to 192.168.1.201:5060:
ACK sip:767@192.168.1.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.117:5160;branch=z9hG4bK6e1b1f43
Max-Forwards: 70
From: "067909090" <sip:067909090@192.168.1.117:5160>;tag=as1840ba23
To: <sip:767@192.168.1.201:5060>;tag=25b6d3472c85c81ai0
Contact: <sip:067909090@192.168.1.117:5160>
Call-ID: 63226cc6213902771545eaa624544442@192.168.1.117:5160
CSeq: 102 ACK
User-Agent: FPBX-13.0.192.16(13.17.0)
Content-Length: 0

---
-- Connected line update to SIP/7320732-00000019 prevented.
-- SIP/767-0000001a answered SIP/7320732-00000019
Audio is at 18380
Adding codec ulaw to SDP

<--- Reliably Transmitting (no NAT) to 193.200.32.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.200.32.23:5060;branch=z9hG4bK067bd17d;received=193.200.32.23;rport=5060
From: "067909090" <sip:067909090@193.200.32.23>;tag=as54fb23cb
To: <sip:7320732@62.64.х.х:5160>;tag=as7abf054d
Call-ID: 7b437b9b3d1512bb11b16b325c7f4f0e@193.200.32.23:5060
CSeq: 103 INVITE
Server: FPBX-13.0.192.16(13.17.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:7320732@62.64.х.х:5160>
Content-Type: application/sdp
Require: timer
Content-Length: 196

v=0
o=root 1330463149 1330463149 IN IP4 62.64.х.х
s=Asterisk PBX 13.17.0
c=IN IP4 62.64.х.х
t=0 0
m=audio 18380 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv

**darkday** » 09 авг 2017, 17:08

Взял дампы с обоих плеч тестовой станции.
RTP трафик присутствует, кодек позволяет восстановить аудио. Т.е. такое ощущение, что на провайдера все уходит четко.

1. Плечо локальное
https://prnt.sc/g6c2vz

2. Плечо на провайдера

https://prnt.sc/g6c385

**darkday** » 09 авг 2017, 17:22

Сводка
http://prntscr.com/g6c9al

Меня смущает то, что:
1. Второй оператор с теми же настройками SIP чувтсвует себя прекрасно
2. Проблема не между станциями - дозвон на проблемного оператора с мобильного - также нет звука вызываемого.

Пробовал на различных каналах связи до провайдера, результат не меняется.
Может быть что-то у провайдера на сервере?
Я не хочу наезжать на провайдера, не проверив себя до конца, но на своей стороне не вижу в чем может быть загвоздка.
Если я не прав, то прошу подсказать, что еще можно проверить.

**april22** » 09 авг 2017, 17:29

если Вы будете . скрывать в картинках ваши не понятные изыскаия - то разбираться ВАМ .
Хотите что бы помогли - выкладывайте дампы полностью ...
думаю ваша серая сеть - это не страшно . да и IP провайдера - не секрет - не вы же один у него клиент.
я бы посмотрел - дампы , может что увидел - картинки да красивые :-)

**darkday** » 09 авг 2017, 17:49

april22 писал(а):если Вы будете . скрывать в картинках ваши не понятные изыскаия - то разбираться ВАМ .
Хотите что бы помогли - выкладывайте дампы полностью ...
думаю ваша серая сеть - это не страшно . да и IP провайдера - не секрет - не вы же один у него клиент.
я бы посмотрел - дампы , может что увидел - картинки да красивые

193.200.32.23 IP провайдера, я его не закрывал
62.64.Х.Х это мой белый внешний. Сеть у меня не серая. + скрыл номера телефонов.
Если это принципиально - могу выложить дамп, только в каком виде? raw? или экспорт из wiresharka?

**april22** » 09 авг 2017, 18:57

как сняли

хватит играть в шпионов ....
не видя всей картины - сложно что то сказать .

Пропала слышимость вызываемого абонента.

Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

Re: Пропала слышимость вызываемого абонента.

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