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Пропадание звука в звонках с астериск?

Asterisk IP PBX 4 сообщений -
#1
Ситуация следующая. Есть сервер с Asterisk 11.18.0. К нему подключен gsm шлюз Openvox. Также создано 3 внутренних абонента, к каждому из которых привязана симкарта на шлюзе. Ситуация следующая, менеджер звонит (либо менеджеру звонят), разговор начинается. Все друг друга хорошо слышат. Но в последнее время стал всё чаще пропадать звук. ТО есть разговор не прерывается, а просто абонент и менеджер не слышит друг друга. И это происходит не всегда. Скажите, как можно отследить, почему пропадает звук о обеих сторон? Может у вас есть какие-либо предположения, что это может быть?
Дебаг вызова, при котором пропал звук:
Код:


INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport
Max-Forwards: 70
From: "+375293204872" ;tag=as71d12b59
To:
Contact:
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 INVITE
User-Agent: VoxStack Wireless Gateway
Date: Wed, 25 Feb 1970 00:45:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Best-Codec: alaw
Content-Type: application/sdp
Content-Length: 434

v=0
o=root 615765964 615765964 IN IP4 192.168.0.150
s=VoxStack Wireless Gateway
c=IN IP4 192.168.0.150
t=0 0
m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (15 headers 19 lines) ---
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: Sending to 192.168.0.150:5060 (NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Sending to 192.168.0.150:5060 (NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;received=192.168.0.150;rport=5060
From: "+375293204872" ;tag=as71d12b59
To: ;tag=as378bb9e4
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1edea583"
Content-Length: 0



[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Scheduling destruction of SIP dialog '76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060' in 6400 ms (Method:$
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:

ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK233836d7;rport
Max-Forwards: 70
From: "+375293204872" ;tag=as71d12b59
To: ;tag=as378bb9e4
Contact:
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 102 ACK
User-Agent: VoxStack Wireless Gateway
Content-Length: 0

[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (10 headers 0 lines) ---
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:

INVITE sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;rport
Max-Forwards: 70
From: "+375293204872" ;tag=as71d12b59
To:
Contact:
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
User-Agent: VoxStack Wireless Gateway
Authorization: Digest username="openvox", realm="asterisk", algorithm=MD5, uri="sip:+375296022323@192.168.0.104:4524", nonce="1edea583", response="a9b79f406c3e3acd7088$
Date: Wed, 25 Feb 1970 00:45:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Best-Codec: alaw
Content-Type: application/sdp
Content-Length: 434

v=0
o=root 615765964 615765965 IN IP4 192.168.0.150
s=VoxStack Wireless Gateway
c=IN IP4 192.168.0.150
t=0 0
m=audio 18710 RTP/AVP 8 0 3 9 4 111 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (16 headers 19 lines) ---
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Sending to 192.168.0.150:5060 (no NAT)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Using INVITE request as basis request - 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found peer 'openvox' for '+375293204872' from 192.168.0.150:5060
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] netsock2.c: == Using SIP RTP CoS mark 5
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 8
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 0
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 3
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 9
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 4
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 111
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 18
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found RTP audio format 101
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format PCMA for ID 8
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format PCMU for ID 0
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format GSM for ID 3
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G722 for ID 9
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G723 for ID 4
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G726-32 for ID 111
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format G729 for ID 18
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Found audio description format telephone-event for ID 101
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|gsm|ulaw|alaw|g726|g729|g722)/video=(nothing)/text=(not$
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (tel$
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Peer audio RTP is at port 192.168.0.150:18710
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: Looking for +375296022323 in from-gsm (domain 192.168.0.104)
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c: list_route: hop:
[Aug 13 14:40:46] VERBOSE[31702][C-00000e7f] chan_sip.c:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060
From: "+375293204872" ;tag=as71d12b59
To:
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Length: 0

[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:1] GotoIfTime("SIP/openvox-0000192d", "00:00-23:59|sat-sun|*|*?workinghour$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:2] GotoIfTime("SIP/openvox-0000192d", "18:00-09:00|mon-fri|*|*?workinghour$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [+375296022323@from-gsm:3] Goto("SIP/openvox-0000192d", "otvet3,s,1") in new stack
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Goto (otvet3,s,1)
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:1] Set("SIP/openvox-0000192d", "fname=201508131440-+375293204872-s") in new stack
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:2] MixMonitor("SIP/openvox-0000192d", "/var/www/html/callrecords/201508131440-+375293204$
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] pbx.c: -- Executing [s@otvet3:3] BackGround("SIP/openvox-0000192d", "/var/lib/asterisk/moh/privetstvie/privetstvie") i$
[Aug 13 14:40:46] VERBOSE[20711][C-00000e7f] app_mixmonitor.c: == Begin MixMonitor Recording SIP/openvox-0000192d
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Audio is at 10704
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] chan_sip.c:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK49ade539;received=192.168.0.150;rport=5060
From: "+375293204872" ;tag=as71d12b59
To: ;tag=as5f9b50a2
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact:
Content-Type: application/sdp
Require: timer
Content-Length: 277

v=0
o=root 392634 392634 IN IP4 192.168.0.104
s=Asterisk PBX 11.18.0
c=IN IP4 192.168.0.104
t=0 0
m=audio 10704 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:

ACK sip:+375296022323@192.168.0.104:4524 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.150:5060;branch=z9hG4bK5bf0957e;rport
Max-Forwards: 70
From: "+375293204872" ;tag=as71d12b59
To: ;tag=as5f9b50a2
Contact:
Call-ID: 76c8f9af1a3a4cc960a1d9d23ed84964@192.168.0.150:5060
CSeq: 103 ACK
User-Agent: VoxStack Wireless Gateway
Content-Length: 0

[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c: --- (10 headers 0 lines) ---
[Aug 13 14:40:46] VERBOSE[20710][C-00000e7f] file.c: -- Playing '/var/lib/asterisk/moh/privetstvie/privetstvie.slin' (language 'ru')
[Aug 13 14:40:46] VERBOSE[31676] manager.c: == HTTP Manager 'amocrm' timed out from 127.0.0.1
[Aug 13 14:40:46] VERBOSE[31702] chan_sip.c:


Последний раз редактировалось: Vlady_32 (Чт Авг 13, 2015 13:05)
#2
1. rtp set debug on
2. sip set debug ....

и CLI вам в руки
_________________
P4 3.0 + 1Gb CentOS 5.8 Aster 1.8.16
Не люблю gui-сборки: натуральный продукт вкуснее.
И еще: я ПРОФИ так как НЕ ЛЕНЮСЬ читать литературу.
#3
Искать в наше время видимо очень сложно...

Записывайте с помощью tcpdump в pcap все RTP,
Осваивайте отличную программу Wireshark - RTP Analyzer - он Вам и проиграть сможет, покажет все проблемы с лоссами, джиттером, убеганием клоков итп.
_________________
http://mh.otx.ru SIP/E1 шлюзы Alvis, Добавь E1 к Asterisk, Проапгрейди Заказчику TDA/LDK!
NEW! Alvis-GW-2E1-L по 29500р, спеши! цены
#4
Поиграйтесь с параметром ptime на устройствах. Мне в случае с телефонами Yealink установка пар-ра ptime=20 помогло, т.к тоже были такого рода проблемы