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Asterisk и Retransmission timeout

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

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Asterisk и Retransmission timeout

Сообщение Vlaed » 28 ноя 2017, 14:51

Добрый день.

Asterisk 11.17.1, NAT - нет, телефоны Cisco 7940 (прошивка 8-12-00), Panasonic KH-HDV130

В логах периодически проявляется:
[2017-11-28 09:47:50] WARNING[15965] chan_sip.c: Retransmission timeout reached on transmission 00156387-a8640009-3d6615d4-7e7e81be@192.168.202.60 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response

Анализ логов показал что ошибка проявляется только при стечении двух обстоятельств:
- звонок идёт с телефона Cisco 7940 (с Panasonic подобного не наблюдается);
- вызываемый абонент не ответил на вызов.

Вызовы
[Показать] Спойлер: Вызов с Cisco 7940 - абонент занят
|Time | 192.168.202.60 |
| | | 192.168.202.5 |
|9.263813 | INVITE SDP (g711A g7 |SIP INVITE From: "5016" <sip:5016@192.168.202.5 To:<sip:5015@192.168.202.5 Call-ID:00156387-a8640009-3d6615d4-7e7e81be@192.168.202.60 CSeq:101
| |(50727) ------------------> (5060) |
|9.264796 | 401 Unauthorized |SIP Status 401 Unauthorized
| |(5060) <------------------ (5060) |
|9.352630 | ACK | |SIP ACK From: "5016" <sip:5016@192.168.202.5 To:<sip:5015@192.168.202.5 CSeq:101
| |(50741) ------------------> (5060) |
|9.421417 | INVITE SDP (g711A g7 |SIP INVITE From: "5016" <sip:5016@192.168.202.5 To:<sip:5015@192.168.202.5 Call-ID:00156387-a8640009-3d6615d4-7e7e81be@192.168.202.60 CSeq:102
| |(50727) ------------------> (5060) |
|9.422980 | 100 Trying| |SIP Status 100 Trying
| |(5060) <------------------ (5060) |
|9.424471 | 183 Session Progress |SIP Status 183 Session Progress
| |(5060) <------------------ (5060) |
|9.620135 | RTP (g711A) |RTP, 1302 packets. Duration: 26.021s SSRC: 0x64A88763
| |(19546) ------------------> (12056) |
|9.620498 | RTP (g711A) |RTP, 1302 packets. Duration: 26.021s SSRC: 0x19D81FFB
| |(19546) <------------------ (12056) |
|35.712133| CANCEL | |SIP Request CANCEL CSeq:102
| |(50727) ------------------> (5060) |
|35.712849| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|35.712955| 200 OK | |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|35.799938| ACK | |SIP ACK From: "5016" <sip:5016@192.168.202.5 To:<sip:5015@192.168.202.5 CSeq:102
| |(50727) ------------------> (5060) |
|36.212735| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|37.213944| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|39.212727| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|43.212725| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|47.212734| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|51.212668| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|55.212872| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|59.212704| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|63.212344| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|67.212701| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |


[Показать] Спойлер: Вызов с Panasonic 130 - абонент занят
|Time | 192.168.202.49 |
| | | 192.168.202.5 |
|54.987125| INVITE SDP (g722 g71 |SIP INVITE From: <sip:5017@192.168.202.5 To:<sip:5015@192.168.202.5 Call-ID:6a021dac-8a2ba9213eda7c3dac860080f0808080@KX-HDV130RU CSeq:1
| |(5060) ------------------> (5060) |
|54.988068| 401 Unauthorized |SIP Status 401 Unauthorized
| |(5060) <------------------ (5060) |
|55.076819| ACK | |SIP ACK From: <sip:5017@192.168.202.5 To:<sip:5015@192.168.202.5 CSeq:1
| |(5060) ------------------> (5060) |
|55.139893| INVITE SDP (g722 g71 |SIP INVITE From: <sip:5017@192.168.202.5 To:<sip:5015@192.168.202.5 Call-ID:6a021dac-8a2ba9213eda7c3dac860080f0808080@KX-HDV130RU CSeq:2
| |(5060) ------------------> (5060) |
|55.141522| 100 Trying| |SIP Status 100 Trying
| |(5060) <------------------ (5060) |
|55.143033| 183 Session Progress |SIP Status 183 Session Progress
| |(5060) <------------------ (5060) |
|55.421132| RTP (g711A) |RTP, 1032 packets. Duration: 20.616s SSRC: 0xAB836137
| |(16262) ------------------> (16344) |
|55.421494| RTP (g711A) |RTP, 1032 packets. Duration: 20.616s SSRC: 0x71E1A1DA
| |(16262) <------------------ (16344) |
|76.203477| CANCEL | |SIP Request CANCEL CSeq:2
| |(5060) ------------------> (5060) |
|76.204075| 487 Request Terminat |SIP Status 487 Request Terminated
| |(5060) <------------------ (5060) |
|76.204173| 200 OK | |SIP Status 200 OK
| |(5060) <------------------ (5060) |
|76.241798| ACK | |SIP ACK From: <sip:5017@192.168.202.5 To:<sip:5015@192.168.202.5 CSeq:2
| |(5060) ------------------> (5060) |


[Показать] Спойлер: sip set debug peer 5016
<--- Reliably Transmitting (no NAT) to 192.168.202.60:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK320fe637;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as40580d84
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 101 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="VoIPServer", nonce="4eea9542"
Content-Length: 0


<------------>
<--- Transmitting (no NAT) to 192.168.202.60:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:5015@192.168.202.5:5060>
Content-Length: 0


<------------>
<--- Transmitting (no NAT) to 192.168.202.60:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:5015@192.168.202.5:5060>
Content-Type: application/sdp
Content-Length: 249

v=0
o=root 1013886 1013886 IN IP4 192.168.202.5
s=VoIPServer
c=IN IP4 192.168.202.5
t=0 0
m=audio 13244 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

<------------>
<--- Reliably Transmitting (no NAT) to 192.168.202.60:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


<------------>
[2017-11-28 13:03:53] VERBOSE[15965][C-000023f9] chan_sip.c:
<--- Transmitting (no NAT) to 192.168.202.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 CANCEL
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


<------------>
[2017-11-28 13:03:53] VERBOSE[23570][C-000023f9] pbx.c: == Spawn extension (phones, 5015, 10) exited non-zero on 'SIP/5016-000057df'
[2017-11-28 13:03:53] VERBOSE[15965] chan_sip.c: Retransmitting #1 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:03:54] VERBOSE[15965] chan_sip.c: Retransmitting #2 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:03:56] VERBOSE[15965] chan_sip.c: Retransmitting #3 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:00] VERBOSE[15965] chan_sip.c: Retransmitting #4 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:04] VERBOSE[15965] chan_sip.c: Retransmitting #5 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:08] VERBOSE[15965] chan_sip.c: Retransmitting #6 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:12] VERBOSE[15965] chan_sip.c: Retransmitting #7 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:16] VERBOSE[15965] chan_sip.c: Retransmitting #8 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:20] VERBOSE[15965] chan_sip.c: Retransmitting #9 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:24] VERBOSE[15965] chan_sip.c: Retransmitting #10 (no NAT) to 192.168.202.60:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.202.60:5060;branch=z9hG4bK32b84430;received=192.168.202.60
From: "5016" <sip:5016@192.168.202.5>;tag=00156387a8640023183e148a-62ad90c8
To: <sip:5015@192.168.202.5>;tag=as7179b2d8
Call-ID: 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60
CSeq: 102 INVITE
Server: VoIPServer
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


---
[2017-11-28 13:04:25] WARNING[15965] chan_sip.c: Retransmission timeout reached on transmission 00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32001ms with no response
[2017-11-28 13:04:25] VERBOSE[15965] chan_sip.c: Really destroying SIP dialog '00156387-a864000d-27a7640f-1c8f2b55@192.168.202.60' Method: CANCEL



Настройка телефона Cisco 7940
[Показать] Спойлер: SIPDefault.cnf
# SIP Default Generic Configuration File
#erase protflash
#reset
#Disable debug: tty mon 0

# Image Version
image_version: P0S3-8-12-00

# Proxy Server
proxy1_address: 192.168.202.5 ; Can be dotted IP or FQDN
proxy1_port: 5060

# Proxy Registration (0-disable (default), 1-enable)
proxy_register: 1

# Phone Registration Expiration [1-3932100 sec] (Default - 3600)
timer_register_expires: 3600

# Codec for media stream (g711ulaw (default), g711alaw, g729a)
preferred_codec: g711alaw

#Настраиваем приоритезацию на телефоне
# TOS bits in media stream [0-5] (Default - 5)
#(Optional) Type of Service (ToS) возможные значения:
# 0 (IP_ROUTINE)
# 1 (IP_PRIORITY)
# 2 (IP_IMMEDIATE)
# 3 (IP_FLASH)
# 4 (IP_OVERIDE)
# 5 (IP_CRITIC)
#tos_media: 5

#Inband -послылаем сигналы по голосовуму тракту(тут пикаем, там распознает)
#Out of band - посылаем сигналы по отдельному каналу, например SIP пакетом
# Inband DTMF Settings (0-disable, 1-enable (default))
dtmf_inband: 1

# Out of band DTMF Settings (
# none - отключено,
# avt - генерирует Out of band по требованию и вырубает inband (default),
# avt_always - всегда генерировать Out of band, отключает inband! )
dtmf_outofband: avt

#Настраиваем уровень сигналов("ПИКов")
#(1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up)
dtmf_db_level: 3
# DTMF AVT Payload (Тип полезной нагрузки в AVT пакете: 96-127)
dtmf_avt_payload: 101 ; при привышении сбрасывается в дефолт: 101

# SIP Таймеры
timer_t1: 500 ; наименьшее значение таймера пересылки сип сообщения по умолчанию 500 msec
timer_t2: 4000 ; наибольшее значение таймера пересылки сип сообщения по умолчанию 4000 msec
sip_retx: 10 ; Максимальное количество попыток пересылки сип сообщения,кроме инвайта, 10 раз
sip_invite_retx: 6 ; Максимальное количество попыток пересылки сип инвайта, 6 раз
timer_invite_expires: 180 ; время истечения срока SIP Invite(в хидере), по умолч 180 sec

# Dialplan template (.xml файл на тфтп сервере откуда грузим диалплан)
dial_template: dialplan


# Папка на TFTP сервере где хранятся настройки
# tftp_cfg_dir: "" ; Например: ./sip_phone/

# Time Server (There are multiple values and configurations refer to Admin Guide for Specifics)
sntp_server: "192.168.202.1" ; SNTP Server IP Address
sntp_mode: unicast ; unicast, multicast, anycast, or directedbroadcast (default)
time_zone: BT ; Time Zone Phone is in
dst_auto_adjust: 0 ; Enable(1-Default)/Disable(0) DST automatic adjustment
time_format_24hr: 1 ; Enable(1 - 24Hr Default)/Disable(0 - 12Hr)
date_format: "D/M/Y"
# dst_offset: 01/00 ; Offset from Phone's time when DST is in effect
# dst_start_month: March ; Month in which DST starts
# dst_start_day: "" ; Day of month in which DST starts
# dst_start_day_of_week: Sunday ; Day of week in which DST starts
# dst_start_week_of_month: 8 ; Week of month in which DST starts
# dst_start_time: 2 ; Time of day in which DST starts
# dst_stop_month: Oct ; Month in which DST stops
# dst_stop_day: "" ; Day of month in which DST stops
# dst_stop_day_of_week: Sunday ; Day of week in which DST stops
# dst_stop_week_of_month: 8 ; Week of month in which DST stops 8=last week of month
# dst_stop_time: 3 ; Time of day in which DST stops

#Не беспокоить (0-выкл, 1-вкл, 2-выкл и вкл нельзя, 3-вкл и выкл нельзя)
dnd_control: 0 ; по умолчанию выкл

# Caller ID Blocking Включает режим блокировки определения номера
# (0-выкл, 1-вкл, 2-выкл и вкл нельзя, 3-вкл и выкл нельзя)
callerid_blocking: 0 ;по умолчанию 0, всегда отсылать свой номер

# не принимать вызовы от скрытых номеров
# Anonymous Call Blocking
# (0-выкл, 1-вкл, 2-выкл и вкл нельзя, 3-вкл и выкл нельзя)
anonymous_call_block: 0 ; по умолчанию 0 (принимать все вызовы)

#Значение, по которому можно сравнивать значения в syncinfo.xml
# перед выполнением удаленной перезагрузки. До 32 символов
sync: 1 ; По умолчанию 1

# Backup Proxy Support
proxy_backup: "" ; Dotted IP of Backup Proxy
proxy_backup_port: 5060 ; Backup Proxy port (default is 5060)

# Emergency Proxy Support
proxy_emergency: "" ; Dotted IP of Emergency Proxy
proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060)

# Outbound Proxy Support
outbound_proxy: "" ; restricted to dotted IP or DNS A record only
outbound_proxy_port: 5060 ; default is 5060

# Настройка VAD(Voice Activity Detection) - определения голоса
enable_vad: 0 ; 0-выкл (по умолчанию), 1-вкл


# NAT/Firewall Traversal
nat_enable: 0 ; 0-выкл (по умолчанию), 1-Включено
nat_address: "" ; Внешний адрес под каким мы выходим (IP/DNS)
voip_control_port: 5060 ; UDP port использ для SIP messages (по умолчанию - 5060)
start_media_port: 10000 ; начало RTP диапазона для данных (по умолчанию - 16384)
end_media_port: 20000 ; конец RTP диапазона для данных (default - 32766)
nat_received_processing: 0 ; 0-выкл (по умолчанию), 1-Включено (нужно включать)


# При 3стороннем разговоре, когда кладём трубку, соеденить оставшиеся стороны
cnf_join_enable : 1 ; 0-откл, 1-вкл (по умолчанию)


# разрешить переадресацию, пока телефон еще звонит
semi_attended_transfer: 1 ; 0-откл, 1-вкл (по умолчанию)

# Telnet Level (разрешает или запрещает использование телнета на телефоне)
telnet_level: 0; 0-Откл (default), 1-Вкл, 2-привелигированный режим

#настройки прокси для доступа к сервисам в инете(если надо)
http_proxy_addr: "" ; Address of HTTP Proxy server
http_proxy_port: 80 ; Port of HTTP Proxy Server (80-default)

# Поддержка динамических DNS/TFTP серверов
dyn_dns_addr_1: "" ; restricted to dotted IP
dyn_dns_addr_2: "" ; restricted to dotted IP
dyn_tftp_addr: "" ; restricted to dotted IP

# Remote Party ID (штучка для CallerID )
remote_party_id: 0 ; 0-выкл (по умолчанию), 1-вкл

#Если линия занята, а нам звонят, то после завершения текущего разговора,
# цыска перезвонит
# Call Hold Ringback (0-выкл, 1-вкл, 2-выкл и вкл нельзя, 3-вкл и выкл нельзя)
call_hold_ringback: 0 ; По умолчанию 0

# Dialtone Stutter for MWI(подача сигнала при наличии сообщений)
stutter_msg_waiting: 0 ; 0-Откл (по умолчанию), 1-Вкл


# RTP Call Statistics (SIP BYE/200 OK message exchange)
call_stats: 0

#Трансфер когда вешают трубку
transfer_onhook_enabled: 1

#network_media_type— Тип согласования эзернет порта. Значения :
# Auto — Автосогласование (по умолчанию)
# Full100 — full-duplex, 100MB
# Half100 — half-duplex, 100MB
# Full10 — full-duplex, 10MB
# Half10 — half-duplex, 10MB.
network_media_type: "Auto"

#autocomplete—(Дополнительно) набор номера без необходимости нажимать кнопку Dial
# 0 (откл), 1 (вкл,). По умолчанию 1.
autocomplete: 1


Подскажите, в каком направлении дальше "копать"?
Vlaed
 
Сообщений: 2
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Re: Asterisk и Retransmission timeout

Сообщение awsswa » 28 ноя 2017, 16:00

стоит ?
directmedia = no
платный суппорт по мере возможностей
awsswa
 
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Откуда: Россия, Пермь skype: yarick_perm
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Re: Asterisk и Retransmission timeout

Сообщение Vlaed » 28 ноя 2017, 16:35

Да, стоит

[Показать] Спойлер: sip.conf
[general]
context=default
externaddr=XXX.XXX.XXX.XXX
localnet=192.168.0.0/16

allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
bindport=5060
srvlookup=no
Language = ru
alwaysauthreject = yes
directmedia=no
;
sendrpid=yes
rpid_update=yes
trustrpid=no
;
progressinband = no
;
useragent=VoIPServer
sdpsession=VoIPServer
realm=VoIPServer
;
session-timers = refuse
session-expires = 120
session-minse = 90
session-refresher = uac

[peer_office]
type=friend
host=dynamic
qualify=no
nat=no
call-limit=2
busylevel=1
transport=udp
dtmfmode=rfc2833
directmedia = no
disallow=all
allow=alaw

[5015](peer_office)
secret=Password
context=phones
cid_number=5015
defaultuser=Test-7970
callerid="Test-7970" <5015>

[5016](peer_office)
secret=Password
context=phones
cid_number=5016
defaultuser=Test-7940
callerid="Test-7940" <5016>


[5017](peer_office)
secret=Password
context=phones
cid_number=5017
defaultuser=Test-HDV130
callerid="Test-HDV130" <5017>
Vlaed
 
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