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Не регистрируется SIP клиент на Asterisk

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Avatara of zzuz
Откуда: SPb
Сообщений: 1307

Re: Не регистрируется SIP клиент на Asterisk

Peer '204' is trying to register, but not configured as host=dynamic
http://линия24.рф - Астериск и прочие бубны!
2009-11-28 15:52

Откуда: Almaty, Kazakhstan
Сообщений: 64

Re: Не регистрируется SIP клиент на Asterisk

zzuz:

Peer '204' is trying to register, but not configured as host=dynamic
Блин даже не знаю как благодарить, в этом деле на самом деле нужен глаз да глаз, все телефон зарегестрировался, теперь буду разбираться дальше!
Спасибо ВСЕМ!!!

добавил в sip.conf к extension'у host=dynamic раньше нормально было, видимо после обновлений это стало обязательным полем)

тем закрыта
2009-11-28 16:01

Сообщений: 5

Re: Не регистрируется SIP клиент на Asterisk

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.
2010-07-12 16:47

Avatara of zlat
Сообщений: 215

Re: Не регистрируется SIP клиент на Asterisk

где поле secret в настройке екстеншенов?
2010-07-12 17:08

Сообщений: 5

Re: Не регистрируется SIP клиент на Asterisk

где поле secret в настройке екстеншенов?
вчера и без него работал. и кстати в екстеншенов тоже бывает secret? а я думал что только в настройках sip'а нужно указать. В книге описано как необязательным, по этому его не прописал.
2010-07-12 17:20

Avatara of zlat
Сообщений: 215

Re: Не регистрируется SIP клиент на Asterisk

екстеншены - это и есть ваши внутренние телефоны
[1000]
type=friend
context=phones
host=dynamic
secret=???
2010-07-12 17:51

Сообщений: 5

Re: Не регистрируется SIP клиент на Asterisk

попробовал, не помогло. Мне кажется сообщение REGISTER не доходить до сервера. * ни чего не показывает. А это не связано с FreePBX? хотя его удалил, может какие-то настройки остались которые влияет напрямую на астериск.
2010-07-12 18:15

Avatara of zlat
Сообщений: 215

Re: Не регистрируется SIP клиент на Asterisk

reload в консоли делаете?
включайте дебаг того ip, на котором софтфон, и смотрите
заодно core set verboce 8, а то мало ли
2010-07-12 18:21

Сообщений: 6521

Re: Не регистрируется SIP клиент на Asterisk

zlat, рискуем получить километр логов. Оно нужно?
2010-07-12 18:23

Avatara of zlat
Сообщений: 215

Re: Не регистрируется SIP клиент на Asterisk

чтобы понять, действительно ли на * ничего не приходит от софтфона, действительно ли выключен фаервол, я думаю -да, дебаг сюда не нужен - это тоже да
2010-07-12 18:35

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