Добрый день.
Такая проблема. Поставил FP (https://wiki.freepbx.org/pages/viewpage.action?pageId=67306922)
Отключил и iptables и firewalld
Но внутренние номера не регистрируются.
tcpdump при 5060 или 5160
06:09:53.891694 IP web.local.57633 > ****.****.ru.5160: UDP, length 572
Но в консоли и логе астера ничего нет.
В логе сип-клиента
Via: SIP/2.0/UDP 172.27.232.213:57633;rport;branch=z9hG4bKPj88f5d8ca754e4596896ad27201b91a24
Route: <sip:****:5160;lr>
Max-Forwards: 70
From: «101» <sip:101@***>;tag=8b5f427897ef416fb3b0767a50723f13
To: «101» <sip:101@***>
Call-ID: 8cbac29058054629aa603a7aaeb53890
CSeq: 41181 REGISTER
User-Agent: MicroSIP/3.15.1
Contact: «101» <sip:101@172.27.232.213:57633;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
—end msg—
09:10:21.525 tsx03467664 Timeout timer event
09:10:21.525 tsx03467664 .State changed from Calling to Terminated, event=TIMER
09:10:21.525 pjsua_acc.c …SIP registration failed, status=408 (Request Timeout)
09:10:21.525 pjsua_acc.c …Scheduling re-registration retry for acc 0 in 291 seconds..
09:10:21.525 tsx03467664 Timeout timer event
09:10:21.525 tsx03467664 .State changed from Terminated to Destroyed, event=TIMER
09:10:21.525 tdta034EBF20 ..Destroying txdata Request msg REGISTER/cseq=41181 (tdta034EBF20)
09:10:21.525 tsx03467664 Transaction destroyed!
*** — Заменил айпишник0
Подскажите в какую сторону копать.
Спасибо!
Problem
Registration fails with one of the following error codes; 818, 687, or 761
Solutions:
This is a generic timeout message:
- Check network settings and make sure the PBX has access to the internet.
- Try pinging a known good domain (i.e. google.com) from the command line.
- If you get «unknown host google.com» then you have a DNS issue. Verify your DNS settings and that you don’t have anything blocking DNS resolution.
- If you get «Network is unreachable» or no reply at all then you have a network issue. Double check the gateway, subnet, & firewall settings.
- If you get «unknown host google.com» then you have a DNS issue. Verify your DNS settings and that you don’t have anything blocking DNS resolution.
- If you are running behind a SonicWall firewall try the steps in the SonicWall Basic Troubleshooting article.
- If you are running an OpenVZ instance (such as with certain Hosted PBXs) review this wiki OpenVZ with FreePBX System article.
- Make sure this server hasn’t been previously registered.
- Login to https://portal.sangoma.com/ and Navigate to Deployments —> List All, and click the edit Icon for the deployment in question. If there is Zend ID listed you will need to reset the Zend ID and re-register the deployment. This should only happen if there have been hardware changes to the server. See this WIKI article for Resetting the Zend ID and Re-registering the deployment. How to Move a Deployment ID to a new PBX
- If none of those resolve the error contact Schmooze Support.
This typically means you are trying to create an account that is already registered.
- If that’s the case make sure you are selecting Yes for «Do you have a Portal account?»
- If that’s not the case try logging into https://portal.sangoma.com/ and/or use the password recovery option to re-gain access to your account.
- If neither of those options work contact the sales department to verify your account @ +1 (920) 886-8130
This usually happens when registering to an existing account.
- Make sure you’ve typed the account email address correctly
- Make sure this server hasn’t already been registered to this account.
- If neither of those options work contact the sales department to verify your account @ +1 (920) 886-8130
Related articles
SonicWall Basic Troubleshooting | OpenVZ with FreePBX System | How to Move a Deployment ID to a new PBX
Error rendering macro ‘contentbylabel’
parameters should not be empty
Здравствуйте.
Проблема такая же, как в первом сообщении.
Выполнял по инструкции
https://wiki.miko.ru/astpanel:ats:freepbx_distro
АТС FreePBX 14.0.13.12
Версия модуля 2.11.3.47
Asterisk 13.22.0
http show status
HTTP Server Status:
Prefix: /asterisk
Server: Asterisk/13.22.0
Server Enabled and Bound to 0.0.0.0:8088
HTTPS Server Enabled and Bound to [::]:8089
Enabled URI’s:
/asterisk/httpstatus => Asterisk HTTP General Status
/asterisk/amanager => HTML Manager Event Interface w/Digest authentication
/asterisk/arawman => Raw HTTP Manager Event Interface w/Digest authentication
/asterisk/manager => HTML Manager Event Interface
/asterisk/rawman => Raw HTTP Manager Event Interface
/asterisk/static/… => Asterisk HTTP Static Delivery
/asterisk/amxml => XML Manager Event Interface w/Digest authentication
/asterisk/mxml => XML Manager Event Interface
/asterisk/ws => Asterisk HTTP WebSocket
Enabled Redirects:
None.
manager show settings
Global Settings:
—————-
Manager (AMI): Yes
Web Manager (AMI/HTTP): Yes
TCP Bindaddress: 0.0.0.0:5038
HTTP Timeout (seconds): 60
TLS Enable: No
TLS Bindaddress: Disabled
TLS Certfile: asterisk.pem
TLS Privatekey:
TLS Cipher:
Allow multiple login: Yes
Display connects: No
Timestamp events: No
Channel vars:
Debug: No
manager show user 1cami
username: 1cami
secret: <Set>
ACL: yes
read perm: call,user,cdr
write perm: call,reporting,originate
displayconnects: no
allowmultiplelogin: yes
Variables:
Пробовал прописывать deny=0.0.0.0/0.0.0.0permit=0.0.0.0/0.0.0.0 и в конфиге и через веб интерфейс, пробовал permit указывать айпи адрес с которого подключался, но ошибка не уходит.
Подскажите в чем может быть проблема?
Заранее благодарю.
Изменено: Васильев Виктор — 04.12.2019 14:57:21
Здравствуйте!
У меня почти такое проблема. Но, клиент как будто ни чего не отправляет, так как * ни чего не сообщает и в логах пусто.
Клиент: X-Lite 3.0
ОС: Ubuntu 10.04
Версия *: Asterisk 1.6.2.9
Asterisk стоит на 192.168.5.60, клиент на 192.168.5.25
Firewall’ы отключены.
Только сильно не пинайте, вчера начал изучать. Вчера все нормально работало и SIP и IAX2. Затем установил FreePBX 2.7. После этого начались все эти проблемы. Затем снес и FreePBX и *, поставил заново *. IAX заработал, но SIP не хочет.
sip.conf:
[general]
[1000]
type=friend
context=phones
host=dynamic
[1001]
type=friend
context=phones
host=dynamic
extensions.conf:
[globals]
[general]
autofallthrough=yes
[default]
exten => s,1,Verbose(1,Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming]
exten => s,1,Answer()
exten => s,n,Echo()
[internal]
exten => 500,1,Verbose(1,Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
exten => 1000,1,Verbose(1,Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()
exten => 1001,1,Verbose(1,Extension 1001)
exten => 1001,n,Dial(SIP/1001,30)
exten => 1001,n,Hangup()
exten => 1002,1,Verbose(1,Extension 1002)
exten => 1002,n,Dial(IAX2/zoiper,30)
exten => 1002,n,Hangup()
exten => 1003,1,Verbose(1,Extension 1003)
exten => 1003,n,Dial(IAX2/ulug,30)
exten => 1003,n,Hangup()
[phones]
include => internal
sip show peers:
Name/username Host Dyn Nat ACL Port Status
1000 (Unspecified) D N 5060 Unmonitored
1001 (Unspecified) D N 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]
sip show settings:
lobal Settings:
—————-
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.9
SDP Session Name: Asterisk PBX 1.6.2.9
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Network QoS Settings:
—————————
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No
Network Settings:
—————————
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Internal IP: 127.0.0.1:5060
STUN server: 0.0.0.0:0
Global Signalling Settings:
—————————
Codecs: 0x8000e (gsm|ulaw|alaw|h263)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Default Settings:
——————
Allowed transports: UDP
Outbound transport: UDP
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
—-
dialplan show:
[ Context ‘ael-dundi-e164-canonical’ created by ‘pbx_ael’ ]
[ Context ‘ael-dundi-e164-customers’ created by ‘pbx_ael’ ]
[ Context ‘ael-dundi-e164-via-pstn’ created by ‘pbx_ael’ ]
[ Context ‘ael-dundi-e164-local’ created by ‘pbx_ael’ ]
Include => ‘ael-dundi-e164-canonical’ [pbx_ael]
Include => ‘ael-dundi-e164-customers’ [pbx_ael]
Include => ‘ael-dundi-e164-via-pstn’ [pbx_ael]
[ Context ‘ael-dundi-e164-switch’ created by ‘pbx_ael’ ]
Alt. Switch => ‘DUNDi/e164’ [pbx_ael]
[ Context ‘ael-dundi-e164-lookup’ created by ‘pbx_ael’ ]
Include => ‘ael-dundi-e164-local’ [pbx_ael]
Include => ‘ael-dundi-e164-switch’ [pbx_ael]
[ Context ‘ael-dundi-e164’ created by ‘pbx_ael’ ]
‘s’ => 1. MSet(LOCAL(exten)=${ARG1}) [pbx_ael]
2. Goto(${exten},1) [pbx_ael]
3. Return() [pbx_ael]
[ Context ‘ael-iaxtel700’ created by ‘pbx_ael’ ]
‘_91700XXXXXXX’ => 1. Dial(IAX2/${IAXINFO-AEL}@iaxtel.com/${EXTEN:1}@iaxtel) [pbx_ael]
[ Context ‘ael-iaxprovider’ created by ‘pbx_ael’ ]
[ Context ‘ael-trunkint’ created by ‘pbx_ael’ ]
‘_9011.’ => 1. Gosub(ael-dundi-e164,s,1(${EXTEN:4})) [pbx_ael]
2. Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}}) [pbx_ael]
Include => ‘ael-dundi-e164-lookup’ [pbx_ael]
[ Context ‘ael-trunkld’ created by ‘pbx_ael’ ]
‘_91NXXNXXXXXX’ => 1. Gosub(ael-dundi-e164,s,1(${EXTEN:1})) [pbx_ael]
2. Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}}) [pbx_ael]
Include => ‘ael-dundi-e164-lookup’ [pbx_ael]
[ Context ‘ael-trunklocal’ created by ‘pbx_ael’ ]
‘_9NXXXXXX’ => 1. Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}}) [pbx_ael]
[ Context ‘ael-trunktollfree’ created by ‘pbx_ael’ ]
‘_91800NXXXXXX’ => 1. Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}}) [pbx_ael]
‘_91866NXXXXXX’ => 1. Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}}) [pbx_ael]
‘_91877NXXXXXX’ => 1. Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}}) [pbx_ael]
‘_91888NXXXXXX’ => 1. Dial(${OUTBOUND-TRUNK}/${EXTEN:${OUTBOUND-TRUNKMSD}}) [pbx_ael]
[ Context ‘ael-international’ created by ‘pbx_ael’ ]
Include => ‘ael-longdistance’ [pbx_ael]
Include => ‘ael-trunkint’ [pbx_ael]
Ignore pattern => ‘9’ [pbx_ael]
[ Context ‘ael-longdistance’ created by ‘pbx_ael’ ]
Include => ‘ael-local’ [pbx_ael]
Include => ‘ael-trunkld’ [pbx_ael]
Ignore pattern => ‘9’ [pbx_ael]
[ Context ‘ael-local’ created by ‘pbx_ael’ ]
Include => ‘ael-default’ [pbx_ael]
Include => ‘ael-trunklocal’ [pbx_ael]
Include => ‘ael-iaxtel700’ [pbx_ael]
Include => ‘ael-trunktollfree’ [pbx_ael]
Include => ‘ael-iaxprovider’ [pbx_ael]
Ignore pattern => ‘9’ [pbx_ael]
[ Context ‘ael-std-exten-ael’ created by ‘pbx_ael’ ]
‘a’ => 1. VoiceMailMain(${ext}) [pbx_ael]
2. Return() [pbx_ael]
‘s’ => 1. MSet(LOCAL(ext)=${ARG1}) [pbx_ael]
2. MSet(LOCAL(dev)=${ARG2}) [pbx_ael]
3. MSet(LOCAL(~~EXTEN~~)=${EXTEN}) [pbx_ael]
4. MSet(LOCAL(~~EXTEN~~)=${~~EXTEN~~}) [pbx_ael]
5. Dial(${dev}/${ext},20) [pbx_ael]
6. Goto(sw-1-${DIALSTATUS},10) [pbx_ael]
7. NoOp(Finish switch-ael-std-exten-ael-1) [pbx_ael]
8. Return() [pbx_ael]
‘sw-1-‘ => 10. Goto(sw-1-.,10) [pbx_ael]
‘sw-1-BUSY’ => 10. Voicemail(${ext},b) [pbx_ael]
11. Goto(s,7) [pbx_ael]
‘_sw-1-.’ => 10. Voicemail(${ext},u) [pbx_ael]
11. Goto(s,7) [pbx_ael]
[ Context ‘ael-demo’ created by ‘pbx_ael’ ]
‘#’ => 1. Playback(demo-thanks) [pbx_ael]
2. Hangup() [pbx_ael]
‘1000’ => 1. Goto(ael-default,s,1) [pbx_ael]
‘2’ => 1. Background(demo-moreinfo) [pbx_ael]
2. Goto(s,instructions) [pbx_ael]
‘3’ => 1. Set(LANGUAGE()=fr) [pbx_ael]
2. Goto(s,restart) [pbx_ael]
‘500’ => 1. Playback(demo-abouttotry) [pbx_ael]
2. Dial(IAX2/guest@misery.digium.com/s@default) [pbx_ael]
3. Playback(demo-nogo) [pbx_ael]
4. Goto(s,instructions) [pbx_ael]
‘600’ => 1. Playback(demo-echotest) [pbx_ael]
2. Echo() [pbx_ael]
3. Playback(demo-echodone) [pbx_ael]
4. Goto(s,instructions) [pbx_ael]
‘8500’ => 1. VoicemailMain() [pbx_ael]
2. Goto(s,instructions) [pbx_ael]
‘i’ => 1. Playback(invalid) [pbx_ael]
‘s’ => 1. Wait(1) [pbx_ael]
2. Answer() [pbx_ael]
3. Set(TIMEOUT(digit)=5) [pbx_ael]
4. Set(TIMEOUT(response)=10) [pbx_ael]
[restart] 5. Background(demo-congrats) [pbx_ael]
[instructions] 6. MSet(x=$[0]) [pbx_ael]
7. GotoIf($[ ${x} < 3]?8:12) [pbx_ael]
8. Background(demo-instruct) [pbx_ael]
9. WaitExten() [pbx_ael]
10. MSet(x=$[${x} + 1]) [pbx_ael]
11. Goto(7) [pbx_ael]
12. NoOp(Finish for-ael-demo-3) [pbx_ael]
‘t’ => 1. Goto(#,1) [pbx_ael]
‘_1234’ => 1. Gosub(ael-std-exten-ael,s,1(${EXTEN}, «IAX2»)) [pbx_ael]
[ Context ‘ael-default’ created by ‘pbx_ael’ ]
Include => ‘ael-demo’ [pbx_ael]
[ Context ‘parkedcalls’ created by ‘features’ ]
‘700’ => 1. Park() [features]
[ Context ‘app_dial_gosub_virtual_context’ created by ‘app_dial’ ]
‘s’ => 1. NoOp() [app_dial]
[ Context ‘app_queue_gosub_virtual_context’ created by ‘app_queue’ ]
‘s’ => 1. NoOp() [app_queue]
[ Context ‘phones’ created by ‘pbx_config’ ]
Include => ‘internal’ [pbx_config]
[ Context ‘internal’ created by ‘pbx_config’ ]
‘1000’ => 1. Verbose(1,Extension 1000) [pbx_config]
2. Dial(SIP/1000,30) [pbx_config]
3. Hangup() [pbx_config]
‘1001’ => 1. Verbose(1,Extension 1001) [pbx_config]
2. Dial(SIP/1001,30) [pbx_config]
3. Hangup() [pbx_config]
‘1002’ => 1. Verbose(1,Extension 1002) [pbx_config]
2. Dial(IAX2/zoiper,30) [pbx_config]
3. Hangup() [pbx_config]
‘1003’ => 1. Verbose(1,Extension 1003) [pbx_config]
2. Dial(IAX2/ulug,30) [pbx_config]
3. Hangup() [pbx_config]
‘500’ => 1. Verbose(1,Echo test application) [pbx_config]
2. Echo() [pbx_config]
3. Hangup() [pbx_config]
[ Context ‘incoming’ created by ‘pbx_config’ ]
‘s’ => 1. Answer() [pbx_config]
2. Echo() [pbx_config]
[ Context ‘default’ created by ‘pbx_config’ ]
‘s’ => 1. Verbose(1,Unrouted call handler) [pbx_config]
2. Answer() [pbx_config]
3. Wait(1) [pbx_config]
4. Playback(tt-weasels) [pbx_config]
5. Hangup() [pbx_config]
-= 35 extensions (85 priorities) in 26 contexts. =-
Пожалуйста, помогите.
После окончания установки сборки Asterisk FreePBX система загрузилась все модули в работе но ни в какую не получается войти в меню Admin->System Admin и выполнить дополнительные настройки. Точнее говоря в этот пункт меню дает в себя войти, но продолжить конфигурацию можно только после активации, а она не проходит со следующей ошибкой «Activation Error! Unable to display activation page. Error returned was: Resolving timed out after 3001 milliseconds»:
Кому интересно ознакомится с пошаговой инструкцией по активации, если проблем с ее запуском не существует добро пожаловать на страницу WiKi FreePBX.
У нас же проблема есть, и сейчас я расскажу Вам как проанализировать происходящее и все исправить чтобы пройти процедуру активации штатно.
На большинстве форумов при возникновении подобной проблемы рекомендуют начать проверку с разрешения в брандмауэре и открыть внешние правила для порта 53 TCP и UDP, так как для DNS обычно используют UDP 53-го порта, но в некоторых случаях переключается на TCP.
Мы можем довольно быстро проверить работоспособность нашего DNS просто попробуем запустить ping до одного из ресурсов интернет:
Как видно мы пытались запустить ping до сайта ya.ru и проверка DNS удалась, так как DNS разрешил имя ya.ru и в ответ выдал нам ip-адрес закрепленный за доменным именем (87.250.250.242). Но так же в этом частном случае мы можем наблюдать, что передано 3 пакета данных из них получено 0, и 100% пакетов утеряно. Мы пробовали отправлять пакеты из внутренней локальной сети в сторону сети интернет, при том DNS работает, делаем вывод, что у нас проблема с шлюзом или же для нашей сети недоступен интернет через шлюз.
И так и есть, ведь мы настраивали Asterisk FreePBX для работы во внутренней сети и не предполагалось, что ему нужен будет выход в Интернет. Более того мы намеренно изолировали эту сеть, чтобы максимально ограничить возможность вмешательства из вне, с целью обезопасить себя от взлома, и дальнейшего похищения телефонного трафика. И так мы разобрались, меняем настройки шлюза или временно меняем настройки сети для получения доступа к сети Интернет! Следующим шагом регистрируем нашу систему и можем увести ее назад в изолированную сеть, если это столь критично и вы работаете в «параноидальном» режиме.
p.s. шутка в тему
Паранойя — это когда тебе кажется, что по твоей ноге ползет паук, и ты постоянно проверяешь, правда ли это. Но на самом деле это кот трется о твою ногу. И тут ты вспоминаешь, что у тебя нет кота. А уже отсюда вытекает шизофрения.