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Channels dont swapped

Проблемы и их решения Asterisk как такового

Модератор: april22

Channels dont swapped

Сообщение thdonatello » 22 окт 2018, 20:30

Приветствую вас, коллеги.
Подскажите, куда копать с таким плавающим глюком?
Оператор в коллцентре общаясь с клиентом хочет "уточнить у специалиста". Для этого нажимает на сипфоне HOLD, переключается на вторую линию, набирает специалисту. Проходит дозвон, специалист отвечает на звонок, прекрасно слышит оператора, но оператор не слышит специалиста. Лог данного звонка (вякие MixMonitor посокращал для приятности глазу):
Код: выделить все
[C-00050a65] netsock2.c: Using SIP RTP CoS mark 5
[C-00050a65] pbx.c: Executing [4203@Office_IN001:1] Macro("SIP/4970-000cac6e", "recording,4970,4203") in new stack
[C-00050a65] pbx.c: Executing [s@macro-recording:1] GotoIf("SIP/4970-000cac6e", "1?mp3:no") in new stack
[C-00050a65] pbx_builtins.c: Goto (macro-recording,s,3)
[C-00050a65] pbx.c: Executing [s@macro-recording:3] Set("SIP/4970-000cac6e", "fname=16_27-4970-4203") in new stack
[C-00050a65] pbx.c: Executing [s@macro-recording:4] Set("SIP/4970-000cac6e", ...") in new stack
[C-00050a65] pbx.c: Executing [s@macro-recording:5] Set("SIP/4970-000cac6e", "CDR(filename)=16_27-4970-4203.mp3") in new stack
[C-00050a65] pbx.c: Executing [s@macro-recording:6] Set("SIP/4970-000cac6e", "CDR(realdst)=4203") in new stack
[C-00050a65] pbx.c: Executing [s@macro-recording:7] Set("SIP/4970-000cac6e", "CDR(remoteip)=192.168.2.79") in new stack
[C-00050a65] pbx.c: Executing [s@macro-recording:8] MixMonitor("SIP/4970-000cac6e", ...") in new stack
[C-00050a65] pbx.c: Executing [s@macro-recording:9] Goto("SIP/4970-000cac6e", "no") in new stack
[C-00050a65] pbx_builtins.c: Goto (macro-recording,s,15)
[C-00050a65] pbx.c: Executing [s@macro-recording:15] Verbose("SIP/4970-000cac6e", "Exit record") in new stack
[C-00050a65] app_verbose.c: Exit record
[C-00050a65] pbx.c: Executing [4203@Office_IN001:2] Dial("SIP/4970-000cac6e", "SIP/4203,,t") in new stack
[C-00050a65] netsock2.c: Using SIP RTP CoS mark 5
[C-00050a65] app_dial.c: Called SIP/4203
[C-00050a65] app_mixmonitor.c: Begin MixMonitor Recording SIP/4970-000cac6e
[C-00050a65] app_dial.c: SIP/4203-000cac6f is ringing
[C-00050a65] app_dial.c: SIP/4203-000cac6f answered SIP/4970-000cac6e
[C-00050a65] bridge_channel.c: Channel SIP/4203-000cac6f joined 'simple_bridge' basic-bridge <becd6e14-895a-4a4a-ba81-4f3d6dd2bd08>
[C-00050a65] bridge_channel.c: Channel SIP/4970-000cac6e joined 'simple_bridge' basic-bridge <becd6e14-895a-4a4a-ba81-4f3d6dd2bd08>
[C-00050a65] bridge_channel.c: Channel SIP/4970-000cac6e left 'simple_bridge' basic-bridge <becd6e14-895a-4a4a-ba81-4f3d6dd2bd08>
[C-00050a65] pbx.c: Spawn extension (Office_IN001, 4203, 2) exited non-zero on 'SIP/4970-000cac6e'
[C-00050a65] bridge_channel.c: Channel SIP/4203-000cac6f left 'simple_bridge' basic-bridge <becd6e14-895a-4a4a-ba81-4f3d6dd2bd08>
[C-00050a65] app_mixmonitor.c: MixMonitor close filestream (mixed)
[C-00050a65] app_mixmonitor.c: Executing [...]
[C-00050a65] app_mixmonitor.c: End MixMonitor Recording SIP/4970-000cac6e

Буквально через пару секунд оператор снова звонит тому же специалисту, и все ок. Лог:
Код: выделить все
[C-00050a67] netsock2.c: Using SIP RTP CoS mark 5
[C-00050a67] pbx.c: Executing [4203@Office_IN001:1] Macro("SIP/4970-000cac78", "recording,4970,4203") in new stack
[C-00050a67] pbx.c: Executing [s@macro-recording:1] GotoIf("SIP/4970-000cac78", "1?mp3:no") in new stack
[C-00050a67] pbx_builtins.c: Goto (macro-recording,s,3)
[C-00050a67] pbx.c: Executing [s@macro-recording:3] Set("SIP/4970-000cac78", "fname=16_27-4970-4203") in new stack
[C-00050a67] pbx.c: Executing [s@macro-recording:4] Set(...) in new stack
[C-00050a67] pbx.c: Executing [s@macro-recording:5] Set("SIP/4970-000cac78", "CDR(filename)=16_27-4970-4203.mp3") in new stack
[C-00050a67] pbx.c: Executing [s@macro-recording:6] Set("SIP/4970-000cac78", "CDR(realdst)=4203") in new stack
[C-00050a67] pbx.c: Executing [s@macro-recording:7] Set("SIP/4970-000cac78", "CDR(remoteip)=192.168.2.79") in new stack
[C-00050a67] pbx.c: Executing [s@macro-recording:8] MixMonitor(...) in new stack
[C-00050a67] app_mixmonitor.c: Begin MixMonitor Recording SIP/4970-000cac78
[C-00050a67] pbx.c: Executing [s@macro-recording:9] Goto("SIP/4970-000cac78", "no") in new stack
[C-00050a67] pbx_builtins.c: Goto (macro-recording,s,15)
[C-00050a67] pbx.c: Executing [s@macro-recording:15] Verbose("SIP/4970-000cac78", "Exit record") in new stack
[C-00050a67] app_verbose.c: Exit record
[C-00050a67] pbx.c: Executing [4203@Office_IN001:2] Dial("SIP/4970-000cac78", "SIP/4203,,t") in new stack
[C-00050a67] netsock2.c: Using SIP RTP CoS mark 5
[C-00050a67] app_dial.c: Called SIP/4203
[C-00050a67] app_dial.c: SIP/4203-000cac79 is ringing
[C-00050a67] app_dial.c: SIP/4203-000cac79 answered SIP/4970-000cac78
[C-00050a67] bridge_channel.c: Channel SIP/4203-000cac79 joined 'simple_bridge' basic-bridge <8e549ff2-4e95-407d-8c5d-9f0c80d14a86>
[C-00050a67] bridge_channel.c: Channel SIP/4970-000cac78 joined 'simple_bridge' basic-bridge <8e549ff2-4e95-407d-8c5d-9f0c80d14a86>
[C-00050a67] res_musiconhold.c: Started music on hold, class 'default', on channel 'SIP/4203-000cac79'
][C-00050a67] bridge_channel.c: Channel SIP/4203-000cac79 left 'simple_bridge' basic-bridge <8e549ff2-4e95-407d-8c5d-9f0c80d14a86>
][C-00050a67] bridge_channel.c: Channel SIP/4970-000cac5d left 'simple_bridge' basic-bridge <de2dcc9e-affb-4b7c-bb37-56da5cac9767>
][C-00050a67] bridge_channel.c: Channel SIP/4203-000cac79 swapped with SIP/4970-000cac5d into 'simple_bridge' basic-bridge <de2dcc9e-affb-4b7c-bb37-56da5cac9767>
[C-00050a67] bridge_channel.c: Channel SIP/4970-000cac78 left 'simple_bridge' basic-bridge <8e549ff2-4e95-407d-8c5d-9f0c80d14a86>
[C-00050a67] pbx.c: Spawn extension (Office_IN001, 4203, 2) exited non-zero on 'SIP/4970-000cac78'
[C-00050a67] app_mixmonitor.c: MixMonitor close filestream (mixed)
[C-00050a67] app_mixmonitor.c: Executing [...]
[C-00050a67] res_musiconhold.c: Stopped music on hold on SIP/4203-000cac79
[C-00050a67] app_mixmonitor.c: End MixMonitor Recording SIP/4970-000cac78
[C-00050a67] bridge_channel.c: Channel SIP/4203-000cac79 left 'simple_bridge' basic-bridge <de2dcc9e-affb-4b7c-bb37-56da5cac9767>

Вижу разницу только в
Код: выделить все
Channel ... swapped with...

Но почему так происходит? Проблемы с сетью? Глюк плавающий, как видно, некоторые телефоны через VPN, но стабильный.
thdonatello
 
Сообщений: 4
Зарегистрирован: 22 окт 2018, 20:11

Re: Channels dont swapped

Сообщение Zavr2008 » 23 окт 2018, 12:00

Телепаты заряжены узнавать настройки пира как и sip debug.
Дружно тараканы в голове понадевали шапочки из фольги?
Asterisk-совместимые Российские SIP/E1 шлюзы Alvis. Для форумчан скидки ! В цены входит настройка связи с Asterisk! Помогаем в настройке TDM АТС: TDA/TDE/LDK и др.
Аватар пользователя
Zavr2008
 
Сообщений: 1194
Зарегистрирован: 27 янв 2011, 01:35

Re: Channels dont swapped

Сообщение thdonatello » 23 окт 2018, 20:05

Ну зачем так сразу, можно же было просто попросить...
Конфиг (шаблон, в самом пире только логин,пароль,ящик и колИД):
Код: выделить все
[udef](!)
type = friend
context = Office_IN001
deny = 0.0.0.0/0.0.0.0
permit = 192.168.0.0/16
host = dynamic
directmedia = no
nat=force_rport,comedia
qualify = yes
call-limit = 2
disallow = all
allow = alaw,ulaw
notifyringing = yes
notifyhold = yes
limitonpeers = yes
allowsubscribe = yes
subscribecontext = MY_BLF

Дебаг (verbose c "sip set debug peer 4970" - номер оператора):
Код: выделить все
[2018-10-23 15:13:06] VERBOSE[47422] chan_sip.c:
<--- SIP read from UDP:192.168.2.79:60649 --->
INVITE sip:4343@sip.contoso.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-252a5b2ee021aa21-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4970@192.168.2.79:60649;rinstance=90d0b42b934f202b>
To: <sip:4343@sip.contoso.org:5060>
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Content-Length: 403

v=0
o=3cxVCE 402224790 81415275 IN IP4 192.168.2.79
s=3cxVCE Audio Call
c=IN IP4 192.168.2.79
t=0 0
m=audio 40046 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40004 RTP/AVP 34
c=IN IP4 192.168.2.79
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
[2018-10-23 15:13:06] VERBOSE[47422] chan_sip.c: --- (13 headers 18 lines) ---
[2018-10-23 15:13:06] VERBOSE[47422] chan_sip.c: Sending to 192.168.2.79:60649 (no NAT)
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Sending to 192.168.2.79:60649 (no NAT)
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Using INVITE request as basis request - MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found peer '4970' for '4970' from 192.168.2.79:60649
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.2.79:60649 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-252a5b2ee021aa21-1---d8754z-;received=192.168.2.79;rport=60649
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
To: <sip:4343@sip.contoso.org:5060>;tag=as45e85122
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 1 INVITE
Server: VoIPserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="VoIPserver", nonce="52fdf547"
Content-Length: 0


<------------>
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Scheduling destruction of SIP dialog 'MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.' in 6656 ms (Method: INVITE)
[2018-10-23 15:13:06] VERBOSE[47422] chan_sip.c:
<--- SIP read from UDP:192.168.2.79:60649 --->
ACK sip:4343@sip.contoso.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-252a5b2ee021aa21-1---d8754z-;rport
Max-Forwards: 70
To: <sip:4343@sip.contoso.org:5060>;tag=as45e85122
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 1 ACK
Content-Length: 0

<------------->
[2018-10-23 15:13:06] VERBOSE[47422] chan_sip.c: --- (8 headers 0 lines) ---
[2018-10-23 15:13:06] VERBOSE[47422] chan_sip.c:
<--- SIP read from UDP:192.168.2.79:60649 --->
INVITE sip:4343@sip.contoso.org:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-441b9c369808e81b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4970@192.168.2.79:60649;rinstance=90d0b42b934f202b>
To: <sip:4343@sip.contoso.org:5060>
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="4970",realm="VoIPserver",nonce="52fdf547",uri="sip:4343@sip.contoso.org:5060",response="20cfe8c2c367a7060197dd90e6785ee9",algorithm=MD5
Content-Length: 403

v=0
o=3cxVCE 402224790 81415275 IN IP4 192.168.2.79
s=3cxVCE Audio Call
c=IN IP4 192.168.2.79
t=0 0
m=audio 40046 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 40004 RTP/AVP 34
c=IN IP4 192.168.2.79
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
a=sendrecv
<------------->
[2018-10-23 15:13:06] VERBOSE[47422] chan_sip.c: --- (14 headers 18 lines) ---
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Sending to 192.168.2.79:60649 (NAT)
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Using INVITE request as basis request - MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found peer '4970' for '4970' from 192.168.2.79:60649
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] netsock2.c: Using SIP RTP CoS mark 5
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found RTP audio format 0
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found RTP audio format 8
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found RTP audio format 3
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found RTP audio format 101
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found audio description format PCMU for ID 0
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found audio description format PCMA for ID 8
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found audio description format GSM for ID 3
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found audio description format telephone-event for ID 101
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found RTP video format 34
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Found video description format H263 for ID 34
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|gsm|alaw)/video=(h263)/text=(nothing), combined - (alaw|ulaw)
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Peer audio RTP is at port 192.168.2.79:40046
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c: Looking for 4343 in Office_IN001 (domain sip.contoso.org)
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] sip/route.c: sip_route_dump: route/path hop: <sip:4970@192.168.2.79:60649;rinstance=90d0b42b934f202b>
[2018-10-23 15:13:06] VERBOSE[47422][C-000515b4] chan_sip.c:
<--- Transmitting (NAT) to 192.168.2.79:60649 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-441b9c369808e81b-1---d8754z-;received=192.168.2.79;rport=60649
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
To: <sip:4343@sip.contoso.org:5060>
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 2 INVITE
Server: VoIPserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4343@192.168.0.25:5060>
Content-Length: 0


<------------>
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [4343@Office_IN001:1] Macro("SIP/4970-000cce4f", "recording,4970,4343") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:1] GotoIf("SIP/4970-000cce4f", "1?mp3:no") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx_builtins.c: Goto (macro-recording,s,3)
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:3] Set("SIP/4970-000cce4f", "fname=15_13_S%-4970-4343") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:4] Set("SIP/4970-000cce4f", "monopt=nice -n 19 /usr/bin/lame -b 32  --silent "/srv/records/2018/10/23/15_13_S%-4970-4343.wav"  "/srv/records/2018/10/23/15_13_S%-4970-4343.mp3" && rm -f "/srv/records/2018/10/23/15_13_S%-4970-4343.wav" && chmod o+r "/srv/records/2018/10/23/15_13_S%-4970-4343.mp3"") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:5] Set("SIP/4970-000cce4f", "CDR(filename)=15_13_S%-4970-4343.mp3") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:6] Set("SIP/4970-000cce4f", "CDR(realdst)=4343") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:7] Set("SIP/4970-000cce4f", "CDR(remoteip)=192.168.2.79") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:8] MixMonitor("SIP/4970-000cce4f", "/srv/records/2018/10/23/15_13_S%-4970-4343.wav,b,nice -n 19 /usr/bin/lame -b 32  --silent "/srv/records/2018/10/23/15_13_S%-4970-4343.wav"  "/srv/records/2018/10/23/15_13_S%-4970-4343.mp3" && rm -f "/srv/records/2018/10/23/15_13_S%-4970-4343.wav" && chmod o+r "/srv/records/2018/10/23/15_13_S%-4970-4343.mp3"") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:9] Goto("SIP/4970-000cce4f", "no") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx_builtins.c: Goto (macro-recording,s,15)
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [s@macro-recording:15] Verbose("SIP/4970-000cce4f", "Exit record") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] app_verbose.c: Exit record
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] pbx.c: Executing [4343@Office_IN001:2] Dial("SIP/4970-000cce4f", "SIP/4343,,t") in new stack
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] netsock2.c: Using SIP RTP CoS mark 5
[2018-10-23 15:13:06] VERBOSE[29318][C-000515b4] app_mixmonitor.c: Begin MixMonitor Recording SIP/4970-000cce4f
[2018-10-23 15:13:06] VERBOSE[29317][C-000515b4] app_dial.c: Called SIP/4343
[2018-10-23 15:13:08] VERBOSE[29317][C-000515b4] app_dial.c: SIP/4343-000cce50 is ringing
[2018-10-23 15:13:08] VERBOSE[29317][C-000515b4] chan_sip.c:
<--- Transmitting (NAT) to 192.168.2.79:60649 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-441b9c369808e81b-1---d8754z-;received=192.168.2.79;rport=60649
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
To: <sip:4343@sip.contoso.org:5060>;tag=as077cc445
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 2 INVITE
Server: VoIPserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4343@192.168.0.25:5060>
Content-Length: 0


<------------>
[2018-10-23 15:13:20] VERBOSE[29317][C-000515b4] app_dial.c: SIP/4343-000cce50 answered SIP/4970-000cce4f
[2018-10-23 15:13:20] VERBOSE[29317][C-000515b4] chan_sip.c: Audio is at 12768
[2018-10-23 15:13:20] VERBOSE[29317][C-000515b4] chan_sip.c: Adding codec alaw to SDP
[2018-10-23 15:13:20] VERBOSE[29317][C-000515b4] chan_sip.c: Adding codec ulaw to SDP
[2018-10-23 15:13:20] VERBOSE[29317][C-000515b4] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2018-10-23 15:13:20] VERBOSE[29317][C-000515b4] chan_sip.c:
<--- Reliably Transmitting (NAT) to 192.168.2.79:60649 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-441b9c369808e81b-1---d8754z-;received=192.168.2.79;rport=60649
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
To: <sip:4343@sip.contoso.org:5060>;tag=as077cc445
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 2 INVITE
Server: VoIPserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4343@192.168.0.25:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1068416234 1068416234 IN IP4 192.168.0.25
s=VoIPserver
c=IN IP4 192.168.0.25
t=0 0
m=audio 12768 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 34

<------------>
[2018-10-23 15:13:20] VERBOSE[29392][C-000515b4] bridge_channel.c: Channel SIP/4343-000cce50 joined 'simple_bridge' basic-bridge <4ad35ff1-1de0-43a3-84dc-8e4e57f659e0>
[2018-10-23 15:13:20] VERBOSE[29317][C-000515b4] bridge_channel.c: Channel SIP/4970-000cce4f joined 'simple_bridge' basic-bridge <4ad35ff1-1de0-43a3-84dc-8e4e57f659e0>
[2018-10-23 15:13:20] VERBOSE[47422] chan_sip.c:
<--- SIP read from UDP:192.168.2.79:60649 --->
ACK sip:4343@192.168.0.25:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-66217c5907329d71-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4970@192.168.2.79:60649;rinstance=90d0b42b934f202b>
To: <sip:4343@sip.contoso.org:5060>;tag=as077cc445
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 2 ACK
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="4970",realm="VoIPserver",nonce="52fdf547",uri="sip:4343@sip.contoso.org:5060",response="20cfe8c2c367a7060197dd90e6785ee9",algorithm=MD5
Content-Length: 0

<------------->
[2018-10-23 15:13:20] VERBOSE[47422] chan_sip.c: --- (11 headers 0 lines) ---
[2018-10-23 15:13:20] VERBOSE[47422] chan_sip.c:
<--- SIP read from UDP:192.168.2.79:60649 --->


<------------->
[2018-10-23 15:13:26] VERBOSE[47422] chan_sip.c:
<--- SIP read from UDP:192.168.2.79:60649 --->
BYE sip:4343@192.168.0.25:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-f03ebe3fe72fc83b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4970@192.168.2.79:60649;rinstance=90d0b42b934f202b>
To: <sip:4343@sip.contoso.org:5060>;tag=as077cc445
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 3 BYE
User-Agent: 3CXPhone 6.0.26523.0
Authorization: Digest username="4970",realm="VoIPserver",nonce="52fdf547",uri="sip:4343@192.168.0.25:5060",response="811094f24304be253fc49c621d8bc51e",algorithm=MD5
Content-Length: 0

<------------->
[2018-10-23 15:13:26] VERBOSE[47422] chan_sip.c: --- (11 headers 0 lines) ---
[2018-10-23 15:13:26] VERBOSE[47422][C-000515b4] chan_sip.c: Sending to 192.168.2.79:60649 (NAT)
[2018-10-23 15:13:26] VERBOSE[29317][C-000515b4] bridge_channel.c: Channel SIP/4970-000cce4f left 'simple_bridge' basic-bridge <4ad35ff1-1de0-43a3-84dc-8e4e57f659e0>
[2018-10-23 15:13:26] VERBOSE[29392][C-000515b4] bridge_channel.c: Channel SIP/4343-000cce50 left 'simple_bridge' basic-bridge <4ad35ff1-1de0-43a3-84dc-8e4e57f659e0>
[2018-10-23 15:13:26] VERBOSE[47422][C-000515b4] chan_sip.c: Scheduling destruction of SIP dialog 'MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.' in 6656 ms (Method: BYE)
[2018-10-23 15:13:26] VERBOSE[47422][C-000515b4] chan_sip.c:
<--- Transmitting (NAT) to 192.168.2.79:60649 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-f03ebe3fe72fc83b-1---d8754z-;received=192.168.2.79;rport=60649
From: "username"<sip:4970@sip.contoso.org:5060>;tag=5c6e2966
To: <sip:4343@sip.contoso.org:5060>;tag=as077cc445
Call-ID: MzQ0ODAzOTcyMzc5ZmU4YWNmYTMyNWYxY2VkMmJmNjk.
CSeq: 3 BYE
Server: VoIPserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2018-10-23 15:13:26] VERBOSE[29317][C-000515b4] pbx.c: Spawn extension (Office_IN001, 4343, 2) exited non-zero on 'SIP/4970-000cce4f'
[2018-10-23 15:13:26] VERBOSE[29318][C-000515b4] app_mixmonitor.c: MixMonitor close filestream (mixed)
[2018-10-23 15:13:26] VERBOSE[29318][C-000515b4] app_mixmonitor.c: Executing [nice -n 19 /usr/bin/lame -b 32  --silent "/srv/records/2018/10/23/15_13_S%-4970-4343.wav"  "/srv/records/2018/10/23/15_13_S%-4970-4343.mp3" && rm -f "/srv/records/2018/10/23/15_13_S%-4970-4343.wav" && chmod o+r "/srv/records/2018/10/23/15_13_S%-4970-4343.mp3"]
[2018-10-23 15:13:26] VERBOSE[29318][C-000515b4] app_mixmonitor.c: End MixMonitor Recording SIP/4970-000cce4f


Посмотрел такой же лог "здорового" звонка, в нем после строк:
Код: выделить все
bridge_channel.c: Channel SIP/4343-000cce50 joined 'simple_bridge' basic-bridge
bridge_channel.c: Channel SIP/4970-000cce4f joined 'simple_bridge' basic-bridge

Имеется:
Код: выделить все
VERBOSE[47422] chan_sip.c: Retransmitting #1 (NAT) to 192.168.2.79:60649:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.79:60649;branch=z9hG4bK-d8754z-82595f02ee631149-1---d8754z-;received=192.168.2.79;rport=60649
From: "username"<sip:4970@sip.contoso.org:5060>;tag=e556c645
To: <sip:4343@sip.contoso.org:5060>;tag=as2cef4e95
Call-ID: ZTViZDUyZWIxOWY4MWUwNjNlMDU3ZTU5MzIzNDdiY2Q.
CSeq: 2 INVITE
Server: VoIPserver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:4343@192.168.0.25:5060>
Content-Type: application/sdp
Content-Length: 288
v=0
o=root 1228107379 1228107379 IN IP4 192.168.0.25
s=VoIPserver
c=IN IP4 192.168.0.25
t=0 0
m=audio 18668 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 34
---

Подскажите, куда дальше копать?
Спасибо.
thdonatello
 
Сообщений: 4
Зарегистрирован: 22 окт 2018, 20:11

Re: Channels dont swapped

Сообщение BorisTheBlade » 25 окт 2018, 22:19

я бы посмотрел в сторону NAT - rtp set debug on- посмотреть трафик откуда\куда.
BorisTheBlade
 
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Re: Channels dont swapped

Сообщение murr » 26 окт 2018, 12:23

sip.contoso.org

Так задумано?
murr
 
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Re: Channels dont swapped

Сообщение Zavr2008 » 26 окт 2018, 12:45

murr правильно мыслит)
Настройки этого транка очень важны чтобы понять.
Дальше подойдем к тому, что ТС нужно не утаивать что за маршрутизатор, какой там SIP ALG, почему нет в топике настроек секции [general]
Только надо ли нам это если ему это самому не нужно? :)
Asterisk-совместимые Российские SIP/E1 шлюзы Alvis. Для форумчан скидки ! В цены входит настройка связи с Asterisk! Помогаем в настройке TDM АТС: TDA/TDE/LDK и др.
Аватар пользователя
Zavr2008
 
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Re: Channels dont swapped

Сообщение thdonatello » 31 окт 2018, 08:34

BorisTheBlade спасибо, коротко и по делу.
murr, по-моему логично не палить домен. username у Вас же не вызвало подозрений.
Да с Вас, господин Zavr, как молока с...
И да, мне не нужно, сам разобрался.
Форум звездочетов агрессивный какой-то, неприятно.
thdonatello
 
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Re: Channels dont swapped

Сообщение ded » 31 окт 2018, 11:42

Трое ответили, одному спасибо, двум другим - фу, какие вы неприятные!
Хотя вопрос был по очевидному - в дебаге фигурируют адреса -
Retransmitting #1 (NAT) to 192.168.2.79:60649: хотя поле Contact: <sip:4343@192.168.0.25:5060>

То есть вопрос murr я бы интерпретировал так: sip.contoso.org - во что резольвится? Где там НАТ вообще? Между 192.168.2.79 и 192.168.0.25? Оба за НАТом? Это же классика, наблюдается при включении RTP debug, и решается классическими методами.

И на основании этих замечаний берётся ведёрко с коричневой жидкостью и весь форум красится - делается глобальный вывод
thdonatello писал(а):Форум звездочетов агрессивный какой-то, неприятно.
ded
 
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Re: Channels dont swapped

Сообщение thdonatello » 31 окт 2018, 13:15

Здравствуйте, ded. Думаю, мое личное дело, кому высказывать благодарность, а кому нет.
Ну адреса и адреса, что с того? я вам больше скажу, NAT'а у меня даже в VPN нет, а 192.168.0.25 и 192.168.2.79 - это вообще один бродкаст домен (22).
К murr'у я действительно оказался не справедлив, простите murr. sip.contoso.org резолвится в звездочку.
И на основании этих замечаний берётся ведёрко с коричневой жидкостью и весь форум красится - делается глобальный вывод

Ну да, встречают по одежке... Первым ответом моим тараканам шапочки надели, а они, когда теряют связь со спутниками, ведут себя как сорвавшиеся собаки.
У всех прошу прощения за свою невоспитанность. Впредь буду тактичнее.
thdonatello
 
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Re: Channels dont swapped

Сообщение ded » 31 окт 2018, 13:22

Ну да, как то боком. Вы сами то не увидели парадокса
chan_sip.c: Retransmitting #1 (NAT)
при том что
thdonatello писал(а):NAT'а у меня даже в VPN нет,
???
ded
 
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