Модератор: april22
Периодически падают SIP подключения Asterisk
Здравствуйте, на сервере установлен дистрибутив Sangoma Linux с FreePBX 14.0.13.4 периодически отваливаются SIP подключения у менеджеров, перезапуск сервиса астериска не помогает. Все начинает работать только после перезагрузки сервера.
IP адресов менеджеров нету в черном списке iptables.
Связь у менеджеров идет через webcrt ( библиотека sipml5 на фронте )
Из-за чего может падать подключение?
Вот часть логов астериска на момент сбоя ( менеджеры пожаловались на то что отвалилась связь в промежутке времени с 10:20 до 10:25 )
[Показать] Спойлер:
[2019-07-25 09:37:02] VERBOSE[15391] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.5.13’
[2019-07-25 09:40:07] VERBOSE[16476] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:49216’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 09:40:07] ERROR[16477] iostream.c: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2019-07-25 09:40:07] ERROR[16477] iostream.c: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2019-07-25 09:40:07] VERBOSE[16476] chan_sip.c: Registered SIP ‘6000003’ at **.**.***.***:49216
[2019-07-25 09:40:07] NOTICE[16476] chan_sip.c: Peer ‘6000003’ is now Reachable. (38ms / 2000ms)
[2019-07-25 09:40:23] VERBOSE[16515] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:31035’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 09:40:25] VERBOSE[16515] chan_sip.c: Registered SIP ‘6000003’ at **.**.***.***:31035
[2019-07-25 09:40:40] VERBOSE[10051] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:49216’ closed
[2019-07-25 10:12:40] VERBOSE[16081] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.5.13’
[2019-07-25 10:20:50] VERBOSE[25061][C-00000058] res_rtp_asterisk.c: DTLS ECDH initialized (automatic), faster PFS enabled
[2019-07-25 10:20:50] VERBOSE[25061][C-00000058] netsock2.c: Using SIP RTP TOS bits 184
[2019-07-25 10:20:50] VERBOSE[25061][C-00000058] netsock2.c: Using SIP RTP CoS mark 5
[2019-07-25 10:21:02] VERBOSE[21979] res_http_websocket.c: WebSocket connection from ‘**.***.**.***:31408’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:21:29] NOTICE[10051] chan_sip.c: Peer ‘999000001’ is now UNREACHABLE! Last qualify: 17
[2019-07-25 10:21:37] NOTICE[10051] chan_sip.c: Peer ‘6000003’ is now UNREACHABLE! Last qualify: 23
[2019-07-25 10:21:39] ERROR[10051] iostream.c: SSL_shutdown() failed: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2019-07-25 10:21:39] VERBOSE[10051] res_http_websocket.c: WebSocket connection from ‘**.***.**.***:28599’ forcefully closed due to fatal write error
[2019-07-25 10:21:39] WARNING[10051] chan_sip.c: sip_xmit of 0x7f7bf40439e0 (len 634) to **.***.**.***:28599 returned -1: Transport endpoint is not connected
[2019-07-25 10:21:46] VERBOSE[22077] res_http_websocket.c: WebSocket connection from ‘**.***.**.***:44722’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:21:53] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:22:07] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:22:21] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:22:35] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:22:49] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:23:03] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:23:17] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:23:31] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:23:45] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
[2019-07-25 10:24:16] VERBOSE[22459] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:48039’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:24:50] VERBOSE[22465] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:49800’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:25:01] VERBOSE[22468] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:49812’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:25:39] VERBOSE[22608] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:48701’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:25:43] VERBOSE[22609] res_http_websocket.c: WebSocket connection from ‘**.***.**.**:11725’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:26:51] VERBOSE[9595] asterisk.c: Remote UNIX connection
[2019-07-25 10:26:51] VERBOSE[22804] asterisk.c: Remote UNIX connection disconnected
[2019-07-25 10:27:02] VERBOSE[9595] asterisk.c: Remote UNIX connection
[2019-07-25 10:27:08] VERBOSE[22860] asterisk.c: Remote UNIX connection disconnected
[2019-07-25 10:27:13] VERBOSE[20929] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘192.168.5.13’
[2019-07-25 10:27:28] VERBOSE[9595] asterisk.c: Remote UNIX connection
[2019-07-25 10:27:28] VERBOSE[23006] asterisk.c: Remote UNIX connection disconnected
[2019-07-25 10:27:36] VERBOSE[9595] asterisk.c: Remote UNIX connection
[2019-07-25 10:27:43] VERBOSE[23022] res_http_websocket.c: WebSocket connection from ‘**.***.**.**:65344’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:29:05] VERBOSE[23228] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:50186’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:30:27] VERBOSE[23450] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:50711’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:31:21] VERBOSE[23591] res_http_websocket.c: WebSocket connection from ‘**.***.**.**:6911’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:31:59] VERBOSE[23621] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:51323’ for protocol ‘sip’ accepted using version ’13’
[2019-07-25 10:32:09] VERBOSE[23694] res_http_websocket.c: WebSocket connection from ‘**.**.***.***:51405’ for protocol ‘sip’ accepted using version ’13’
- XOLLER
- Сообщений: 2
- Зарегистрирован: 25 июл 2019, 16:09
Re: Периодически падают SIP подключения Asterisk
zzuz » 26 июл 2019, 17:28
[2019-07-25 09:40:07] ERROR[16477] iostream.c: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error
[2019-07-25 10:21:53] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
Если сеть не работает , то и перезапуск сервера не поможет.
Линия24 — Системы Массового Телефонного Обслуживания
-
zzuz - Сообщений: 1658
- Зарегистрирован: 21 сен 2010, 13:33
-
- Сайт
- ICQ
Re: Периодически падают SIP подключения Asterisk
XOLLER » 26 июл 2019, 17:36
так в том и дело, что сеть работает я подключаюсь спокойно по SSH и перезагружаю сервер. После перезагрузки все летает и работает на ура
- XOLLER
- Сообщений: 2
- Зарегистрирован: 25 июл 2019, 16:09
Re: Периодически падают SIP подключения Asterisk
ded » 26 июл 2019, 17:39
Здравствуйте, XOLLER!
Я периодически жалуюсь всем подряд на
1) боли в области поясницы (правда!)
2) невнимательность участников форума, которые приходят с проблемами, логами, вопросами, ответы на которые обычно лежат в этих самых логах (как у вас сейчас)
3) инертность пользователей, заказчиков, которым надо-надо прямо вчера ещё, а потом пропадают в туман и тишину.
Посоветуйте по этим трём проблемам что-то?
- ded
- Сообщений: 15466
- Зарегистрирован: 26 авг 2010, 19:00
Re: Периодически падают SIP подключения Asterisk
virus_net » 27 июл 2019, 11:33
ded писал(а):Я периодически жалуюсь всем подряд на
и я тоже жалуюсь, а мне помогут ?
Особенно на пользователей/заказчиков, которые смотрят в лог, где разработчик старался, выводил ошибку, а пользователь/заказчик в розовых очках видимо.
XOLLER писал(а):[2019-07-25 10:21:53] ERROR[10051] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
какое из 3х выделенных слов вам перевести ?
То что вы подключаетесь к серверу по SSH это конечно здорово, но не показатель того, что с сетью все в порядке. Вам же лог черным по белому пишет о том, что есть проблема.
Поэтому снимите розовые очки, смотрите в логи и не только астериска, но и самой системы, а так же лог свича и статистику порта куда воткнут сервер.
Открою вам ещё одно заморское слово — ТРАБЛШУТ (troubleshoot).
мой SIP URI sip:virus_net@asterisk.ru
bitname.ru — Домены .bit (namecoin) .emc .coin .lib .bazar (emercoin)
ENUMER — звони бесплатно и напрямую.
- virus_net
- Сообщений: 2337
- Зарегистрирован: 05 июн 2013, 08:12
- Откуда: Москва
Re: Периодически падают SIP подключения Asterisk
luker » 31 июл 2019, 10:24
В первую очередь следует проверить , все сетевое оборудование. У меня был случай , когда после грозы вышел из строя свитч d-link не помню какой модели и начал в сеть выдавать непонятные фреймы , но не всегда это делал , а как его переклинит, от этого вся сеть страдала , я 3 дня ковырял эту плавающую неисправность отключая один свитч за другим , в конце все таки выловил благо wireshark примерно показал , куда копать,а zabbixа не было настроенного. Для начала попробуйте командой ping пробить самые дальние от себя устройства или например телефон менеджера , который отваливается на предмет потери пакетов. Есть еще програмка Ping-Terminal я ей пользуюсь. Потом нужно будет ковырять дальше.
- luker
- Сообщений: 14
- Зарегистрирован: 24 июл 2019, 17:45
Re: Периодически падают SIP подключения Asterisk
Zavr2008 » 06 авг 2019, 19:24
res_http_websocket.c: WebSocket connection from ‘**.***.**.***:44722’ for protocol ‘sip’ accepted using version ’13’
Для WebRTC нужно многое. Например понимание что это такое и как оно работает…
Российские шлюзы E1 Alvis-GW. Voip-Модернизация УПАТС, FreePBX, CRM. Продолжаем работать, импортозамещаем!
-
Zavr2008 - Сообщений: 2016
- Зарегистрирован: 27 янв 2011, 01:35
-
- Сайт
Вернуться в Конфигурация и настройка Asterisk
Кто сейчас на форуме
Сейчас этот форум просматривают: нет зарегистрированных пользователей и гости: 9
I have a setup of Asterisk PBX running on a Raspberry Pi 1 (IP 192.168.2.2) and it runs a few local IP phones just fine. I recently purchased a Grandstream HT813 gateway (IP 192.168.2.3) to connect both an analog phone to IP and IP to the PSTN.
The analog phone works perfectly and dialing in from the PSTN works perfectly. Dialing out was much more problematic. First of all, I have the following in my sip.conf: (I didn’t include the other phones, just the FXO port)
[general]
context=default
sipdebug=yes
bindaddr=0.0.0.0
[FXO]
type=peer
context=inbound
host=192.168.2.3
insecure=port
dtmfmode=rfc2833
canreinvite=no
secret=secret_was_here
I redacted the secret above for obvious reasons. When I dail out to the PSTN via a IP phone (IP 192.168.2.5), I get the following error, where [myphonenumber]
is my cell phone number that I was calling to test:
== Setting global variable 'SIPDOMAIN' to '192.168.2.2'
-- Executing [[myphonenumber]@local:1] Goto("PJSIP/104-00000000", "dialout,[myphonenumber],1") in new stack
-- Goto (dialout,[myphonenumber],1)
-- Executing [[myphonenumber]@dialout:1] Dial("PJSIP/104-00000000", "SIP/FXO/[myphonenumber], 30") in new stack
== Using SIP RTP CoS mark 5
Audio is at 13370
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.3:5060:
INVITE sip:[myphonenumber]@192.168.2.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK71b94203
Max-Forwards: 70
From: <sip:104@192.168.2.2>;tag=as162b0b68
To: <sip:[myphonenumber]@192.168.2.3>
Contact: <sip:104@192.168.2.2:5060>
Call-ID: 3dc727711219fa6e4171966d519ac8ee@192.168.2.2:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1
Date: Thu, 15 Aug 2019 19:11:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 1202030057 1202030057 IN IP4 192.168.2.2
s=Asterisk PBX 16.2.1~dfsg-1
c=IN IP4 192.168.2.2
t=0 0
m=audio 13370 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 0-16
a=maxptime:150
a=sendrecv
---
[Aug 15 20:11:17] ERROR[23679][C-00000001]: chan_sip.c:4321 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
So I set up a switch port to mirror the output of my Asterisk server Pi. It sends no data to the gateway when this error occurs, so the problem is in Asterisk. I have seen many posts saying that this is a permission or firewall error, of course. First of all, I disabled the firewall on the Asterisk server during testing. (It’s a small network) I also ran nmap -v -sU -p 5060 192.168.2.3
, which confirmed that my gateway is working just fine. If I run that command as the Asterisk user, I get a permission error, of course. I see nothing else in the syslog. Any ideas?
UPDATE: I had another look at it and I traced the call through the C code. It looks like the issue is that the sipsock
variable (line 1101) is set to -1 when sip_prepare_socket
confirms that the connection requires a UDP socket (line 29503). This returns -1 throwing an error into __sip_xmit
. I will search my log files for any additional information. Also, my Asterisk version is v16.2.1.
Moderators: enjay, williamconley, Staydog, mflorell, MJCoate, mcargile, Kumba
- Reply with quote
Errors after Fresh install
I’m having these 3 issues after a fresh iciBox_v8_1.x86_64-8.1.2 install:
1) Asterisk is not starting on boot. If I start manually it works.
2)I keep getting this on CLI:
sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
3)No sound when logging in using WebRTC viciphone.
Here’s the CLI:
[Oct 31 10:42:44] == WebSocket connection from ‘12.226.124.15:56590’ for protocol ‘sip’ accepted using version ’13’
[Oct 31 10:42:45] ERROR[4442]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe
[Oct 31 10:42:45] == WebSocket connection from ‘12.226.124.15:56307’ forcefully closed due to fatal write error
[Oct 31 10:42:45] — Registered SIP ‘2525’ at 12.226.124.15:56590
I do not hear the initial «You are the only……»
- jfsfx
- Posts: 113
- Joined: Fri Sep 10, 2010 8:36 pm
- Reply with quote
Re: Errors after Fresh install
by jfsfx » Wed Oct 31, 2018 12:30 pm
Issue seems to be with the certificate (I’ve used vicibox-certbot). I did not get any errors while installation.
ERROR[4675]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe
Any ideas?
- jfsfx
- Posts: 113
- Joined: Fri Sep 10, 2010 8:36 pm
- Reply with quote
Re: Errors after Fresh install
by jfsfx » Wed Oct 31, 2018 12:46 pm
I get this immediately after login:
Agent Screen:
No one is in your session: 8600051
WebRTC log shows:
REGISTERED
2018-10-31 13:45:31 =>
displayName: 2525
uri:
2525@66.XXX
.X.146
authorizationUser: 2525
password: m5Y4FLgd543543weeWWW
wsServers:
wss://dialer.certf.com:8089/ws
CLI:
ERROR[4675]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe
- jfsfx
- Posts: 113
- Joined: Fri Sep 10, 2010 8:36 pm
- Reply with quote
Re: Errors after Fresh install
by thephaseusa » Wed Oct 31, 2018 3:08 pm
Did you enter the location of your certificates in /etc/apache2/vhosts.d/1111-default-ssl.conf
SSLCertificateFile /etc/certbot/live/FQDN/cert.pem
SSLCACertificateFile /etc/certbox/live/FQDN/fullchain.pem
SSLCertificateKeyFile /etc/certbox/live/FQDN/privkey.pem
- thephaseusa
- Posts: 345
- Joined: Tue May 16, 2017 2:23 pm
- Reply with quote
Re: Errors after Fresh install
by jfsfx » Wed Oct 31, 2018 3:33 pm
Now I’m getting this:
viciphone debug log:
2018-10-31 15:21:53 =>
displayName: 2525
uri:
2525@66.58.85.146
authorizationUser: 2525
password: m5Y4FLQOUfROo2weeWWW
wsServers:
wss://dialer.crft.com:8089/ws
2018-10-31 15:21:56 => Got Invite from <0000000000> «ACagcW15410136786666666666666666»
2018-10-31 15:21:56 => Auto-Answered Call
Call gets auto-answered on viciphone. But here’s the CLI:
[Oct 31 15:12:26] == WebSocket connection from ‘12.333.333.65:49729’ for protocol ‘sip’ accepted using version ’13’
[Oct 31 15:12:26] — Registered SIP ‘2525’ at 12.XXX.XXX.65:49729
[Oct 31 15:12:26] NOTICE[3405]: chan_sip.c:24639 handle_response_peerpoke: Peer ‘2525’ is now Reachable. (105ms / 2000ms)
[Oct 31 15:12:29] == Manager ‘sendcron’ logged on from 127.0.0.1
[Oct 31 15:12:29] == DTLS ECDH initialized (automatic), faster PFS enabled
[Oct 31 15:12:29] == Using SIP RTP CoS mark 5
[Oct 31 15:12:29] — Called 2525
[Oct 31 15:12:29] — SIP/2525-00000003 is ringing
asterisk is not detecting the answer.
- jfsfx
- Posts: 113
- Joined: Fri Sep 10, 2010 8:36 pm
- Reply with quote
Re: Errors after Fresh install
by jfsfx » Wed Oct 31, 2018 7:46 pm
[Oct 31 20:43:08] == WebSocket connection from ‘12.226.XX.65:57378’ for protocol ‘sip’ accepted using version ’13’
[Oct 31 20:43:08] — Registered SIP ‘2525’ at 12.226.XX.65:57378
[Oct 31 20:43:08] > Saved useragent «VICIphone 1.0-rc1» for peer 2525
[Oct 31 20:43:08] NOTICE[22371]: chan_sip.c:24639 handle_response_peerpoke: Peer ‘2525’ is now Reachable. (100ms / 2000ms)
[Oct 31 20:43:11] == Manager ‘sendcron’ logged on from 127.0.0.1
[Oct 31 20:43:11] == DTLS ECDH initialized (automatic), faster PFS enabled
[Oct 31 20:43:11] == Using SIP RTP CoS mark 5
[Oct 31 20:43:11] — Called 2525
[Oct 31 20:43:11] — SIP/2525-00000000 is ringing
WebRTC log:
2018-10-31 20:43:44 =>
displayName: 2525
uri:
2525@66.XXX
.4.146
authorizationUser: 2525
password: m5Y4FLQOUfRygyg666gg
wsServers:
wss://dialer.clrt.com:8089/ws
2018-10-31 20:43:47 => Got Invite from <0000000000> «ACagcW15410329896666666666666666»
2018-10-31 20:43:52 => Answered Call
- jfsfx
- Posts: 113
- Joined: Fri Sep 10, 2010 8:36 pm
- Reply with quote
Re: Errors after Fresh install
by Kumba » Fri Nov 02, 2018 1:03 pm
if vicibox-certbot ran without any real issues then it should have set the correct SSL certificates in /etc/asterisk/http.conf and /etc/apache2/vhosts.d/1111-default-ssl.conf.
The only thing you really need to do is edit the ‘WebRTC’ template to point to the certificates as well. So I would double check that and see if that’s your issue. You can find it under Admin —> Templates. It looks like you already got the WSS set correctly since it’s registering.
Other then that I’m not sure. Some of the issues sound network related. All 3 issues can be attributed to network issues but that doesn’t mean it’s their cause. There is an init script log at /var/log/vicidial.log that should show you what the init script was seeing during start up. Item 2 and 3 almost 100% describe a NAT traversal issue.
- Kumba
- Posts: 922
- Joined: Tue Oct 16, 2007 11:44 pm
- Location: Florida
- Reply with quote
Re: Errors after Fresh install
by dspaan » Mon Dec 03, 2018 3:35 pm
I’m also getting this error on the asterisk console while testing vicibox 8.1.2:
Serious Network Trouble; __sip_xmit returns error for pkt data
It happens after a while when you are logged in as agent with webphone, not immediately.
It seems to be WebRTC related:
https://community.freepbx.org/t/chan-si … a/39534/16
I also found this:
In chan_sip, a “__sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error” error typically means that there’s a misconfiguration on the IP addressing on your system. Asterisk is trying to send a SIP message, and passes the network packet to the operating system, and the operating system says “Whoa there buddy, that’s not right”.
However i have checked all the obvious places but my IP is correct. Also ran the update IP script.
Another find on the FreePBX forum:
So since I posted the OP, I found out what the issue was. (or at least eliminated it by changing a setting.) On old versions of FreePBX, in Advanced SIP settings, in order to enable TCP connections, you had to manually type:
tcpenable =yes
In new versions, there is now a button to enable TCP. Having both settings enabled seems to cause these issues. Most likely, the config file for sip.conf gets messed up having these similar/identical settings. Removing the old entry eliminates the errors.
This is a Vicibox 8.1.2 server with SVN 5053.
The cert was installed with vicibox-certbot.
I checked /etc/asterisk/http.conf and /etc/apache2/vhosts.d/1111-default-ssl.conf and both have the correct paths.
The webphone is working normally there are no functional problems as far as i can see.
I also see this error:
ERROR[4675]: tcptls.c:447 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Broken pipe
Regards, Dennis
Vicibox 9.0.1
Version: 2.14b0.5
SVN Version: 3199
DB Schema Version: 1588
Build: 200310-1801
- dspaan
- Posts: 1365
- Joined: Fri Aug 21, 2009 1:40 pm
- Location: The Netherlands
- Reply with quote
Re: Errors after Fresh install
by carpenox » Wed Dec 09, 2020 7:54 pm
dennis,
you ever get the broken pipe error situated?
Nox
Alma Linux 8.6 | Version: 2.14-865a | BUILD: 220831-0850 | SVN Version: 3636 | DB Schema Version: 1666 | Asterisk 16.17.0-vici
www.CyburDial.net -:- 725-22-CYBUR -:- My Blog: http://vicidial.blog -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
- carpenox
- Posts: 1923
- Joined: Wed Apr 08, 2020 2:02 am
- Location: Coral Springs, FL
-
- Website
- YIM
- Reply with quote
Re: Errors after Fresh install
by dspaan » Thu Dec 10, 2020 3:32 am
Not actively, but i don’t think i see them anymore or i have become blind for them
I do keep seeing ‘Serious Network Trouble’ and ‘SRTCP unprotect failed because of authentication failure’
Regards, Dennis
Vicibox 9.0.1
Version: 2.14b0.5
SVN Version: 3199
DB Schema Version: 1588
Build: 200310-1801
- dspaan
- Posts: 1365
- Joined: Fri Aug 21, 2009 1:40 pm
- Location: The Netherlands
- Reply with quote
Re: Errors after Fresh install
by carpenox » Thu Dec 10, 2020 8:52 am
that just started recently right? does it effect the audio on some agents calls only?
Alma Linux 8.6 | Version: 2.14-865a | BUILD: 220831-0850 | SVN Version: 3636 | DB Schema Version: 1666 | Asterisk 16.17.0-vici
www.CyburDial.net -:- 725-22-CYBUR -:- My Blog: http://vicidial.blog -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
- carpenox
- Posts: 1923
- Joined: Wed Apr 08, 2020 2:02 am
- Location: Coral Springs, FL
-
- Website
- YIM
- Reply with quote
Re: Errors after Fresh install
by dspaan » Thu Dec 10, 2020 9:03 am
I just saw the Underlying BIO error too today. We have no audio problems resulting from any of them. Also the vicidial group said all these warnings can be safely ignored.
Regards, Dennis
Vicibox 9.0.1
Version: 2.14b0.5
SVN Version: 3199
DB Schema Version: 1588
Build: 200310-1801
- dspaan
- Posts: 1365
- Joined: Fri Aug 21, 2009 1:40 pm
- Location: The Netherlands
- Reply with quote
Re: Errors after Fresh install
by carpenox » Thu Dec 10, 2020 10:06 am
sounds good
Alma Linux 8.6 | Version: 2.14-865a | BUILD: 220831-0850 | SVN Version: 3636 | DB Schema Version: 1666 | Asterisk 16.17.0-vici
www.CyburDial.net -:- 725-22-CYBUR -:- My Blog: http://vicidial.blog -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
- carpenox
- Posts: 1923
- Joined: Wed Apr 08, 2020 2:02 am
- Location: Coral Springs, FL
-
- Website
- YIM
Return to ViciBox Server Install and Demo
Who is online
Users browsing this forum: No registered users and 11 guests