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Регистрация не проходит

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Доброго времени суток. Народ я уже задолбался, кто знает помогите.. Это уже 3 тема про одну и туже проблему. Значит провайдер дал тестовый номер с регистрацией по адресу.. настройки такие

172.17.99.41 ip моей карточки
255.255.255.224 маска
172.17.99.33 шлюз
217.30.246.164 call server
a law
20 ms
5673998 номер телефон

настройки sip.conf

[5673998]
type=friend
host=217.30.246.164
context=office
disallow=all
allow=alaw
insecure=invite
canreinvite=no
dtmfmode=rfc2833

настройки extensions.conf

[general]
[office]
exten => 200,1, Dial(SIP/200,20,t)
exten => 201,1, Dial(SIP/201,20,t)
exten => 202,1, Dial(SIP/202,20,t)
exten => 250,1, Dial(SIP/250,20,t)
include => bee
include => from5673998

[bee]
exten => _0xxxxxxx,1,Dial(SIP/5673998/${EXTEN:1})
;exten => _0xxxxxxx,2,Playback(noasnswer)
exten => _0xxxxxxx,2,Hangup

[from5673998]
exten => s,1,Answer
exten => s,2,Dial(SIP/200,25,Ttr)
exten => s,3,Hangup

Пробую позвонить на номер 5676067 выдается такое

<------------>
Scheduling destruction of SIP dialog 'b_4ff90-960ef774e6d93ff_I@192.168.10.28' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.10.28:1024 --->
ACK sip:05676067@192.168.10.30 SIP/2.0
From: sip:250@192.168.10.30;tag=26e998114e6894544e6d97ee_F250192.168.10.28
To: sip:05676067@192.168.10.30;tag=as2fe1a911
Call-ID: b_4ff90-960ef774e6d93ff_I@192.168.10.28
CSeq: 11 ACK
Via: SIP/2.0/UDP 192.168.10.28;branch=z9hG4bKb_4ff90-2f57078b4e6d97f4_I250
Content-Length: 0
Max-Forwards: 70
User-Agent: Avaya one-X Deskphone
Supported: eventlist, 100rel, replaces


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.10.28:1024 --->
INVITE sip:05676067@192.168.10.30 SIP/2.0
From: sip:250@192.168.10.30;tag=26e998114e6894544e6d97ee_F250192.168.10.28
To: sip:05676067@192.168.10.30
Call-ID: b_4ff90-960ef774e6d93ff_I@192.168.10.28
CSeq: 12 INVITE
Via: SIP/2.0/UDP 192.168.10.28;branch=z9hG4bKc_4ff91-2c4d9e7b4e6d9404_I250
Content-Length: 355
Max-Forwards: 70
Contact: <sip:250@192.168.10.28;transport=udp>
Accept-Language: en
Allow: INVITE,CANCEL,BYE,ACK,SUBSCRIBE,NOTIFY,MESSAGE,INFO,PUBLISH,REFER,UPDATE,PRACK
Content-Type: application/sdp
User-Agent: Avaya one-X Deskphone
Supported: eventlist, 100rel, replaces
Authorization: Digest username="250",realm="asterisk",nonce="4b3bb7f2",uri="sip:05676067@192.168.10.30",response="562d3874cc8e0f9557663c6f09cd8b76"

v=0
o=sip:250@192.168.10.28 1 12 IN IP4 192.168.10.28
s=sip:250@192.168.10.28
c=IN IP4 192.168.10.28
b=CT:64
b=AS:64
b=TIAS:64000
t=0 0
m=audio 5004 RTP/AVP 0 8 18 110 120
b=TIAS:64000
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:110 G726-32/8000/1
a=rtpmap:120 telephone-event/8000/1

<------------->
--- (15 headers 16 lines) ---
Sending to 192.168.10.28 : 5060 (no NAT)
Using INVITE request as basis request - b_4ff90-960ef774e6d93ff_I@192.168.10.28
Found peer '250' for '250' from 192.168.10.28:1024
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 110
Found RTP audio format 120
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 110
Found audio description format telephone-event for ID 120
Capabilities: us - 0x8 (alaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.10.28:5004
Looking for 05676067 in office (domain 192.168.10.30)
list_route: hop: <sip:250@192.168.10.28;transport=udp>

<--- Transmitting (no NAT) to 192.168.10.28:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.28;branch=z9hG4bKc_4ff91-2c4d9e7b4e6d9404_I250;received=192.168.10.28
From: sip:250@192.168.10.30;tag=26e998114e6894544e6d97ee_F250192.168.10.28
To: sip:05676067@192.168.10.30
Call-ID: b_4ff90-960ef774e6d93ff_I@192.168.10.28
CSeq: 12 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:05676067@192.168.10.30>
Content-Length: 0


<------------>
Audio is at 172.17.99.41 port 17050
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.30.246.164:5060:
INVITE sip:5676067@217.30.246.164 SIP/2.0
Via: SIP/2.0/UDP 172.17.99.41:5060;branch=z9hG4bK5b1713f6;rport
Max-Forwards: 70
From: "Adel" <sip:250@172.17.99.41>;tag=as2f3d5e9e
To: <sip:5676067@217.30.246.164>
Contact: <sip:250@172.17.99.41>
Call-ID: 72bd9e1a31b7aafe62b7fd5e70d1d73a@172.17.99.41
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 08 Sep 2011 09:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 245

v=0
o=root 175789196 175789196 IN IP4 172.17.99.41
s=Asterisk PBX 1.6.2.9-2+squeeze3
c=IN IP4 172.17.99.41
t=0 0
m=audio 17050 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:217.30.246.164:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.99.41:5060;branch=z9hG4bK5b1713f6;rport
From: "Adel" <sip:250@172.17.99.41>;tag=as2f3d5e9e
To: <sip:5676067@217.30.246.164>
Call-ID: 72bd9e1a31b7aafe62b7fd5e70d1d73a@172.17.99.41
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:217.30.246.164:5060 --->
SIP/2.0 403 Forbidden
Call-ID: 72bd9e1a31b7aafe62b7fd5e70d1d73a@172.17.99.41
CSeq: 102 INVITE
From: "Adel" <sip:250@172.17.99.41>;tag=as2f3d5e9e
To: <sip:5676067@217.30.246.164>;tag=y13nnag292
Via: SIP/2.0/UDP 172.17.99.41:5060;branch=z9hG4bK5b1713f6;rport=5060
Reason: Q.850;cause=21;text="Call rejected"
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 217.30.246.164:5060:
ACK sip:5676067@217.30.246.164 SIP/2.0
Via: SIP/2.0/UDP 172.17.99.41:5060;branch=z9hG4bK5b1713f6;rport
Max-Forwards: 70
From: "Adel" <sip:250@172.17.99.41>;tag=as2f3d5e9e
To: <sip:5676067@217.30.246.164>;tag=y13nnag292
Contact: <sip:250@172.17.99.41>
Call-ID: 72bd9e1a31b7aafe62b7fd5e70d1d73a@172.17.99.41
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
[Sep  8 13:09:25] WARNING[5416]: chan_sip.c:17987 handle_response_invite: Received response: "Forbidden" from '"Adel" <sip:250@172.17.99.41>;tag=as2f3d5e9e'
Scheduling destruction of SIP dialog 'b_4ff90-960ef774e6d93ff_I@192.168.10.28' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 192.168.10.28:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.10.28;branch=z9hG4bKc_4ff91-2c4d9e7b4e6d9404_I250;received=192.168.10.28
From: sip:250@192.168.10.30;tag=26e998114e6894544e6d97ee_F250192.168.10.28
To: sip:05676067@192.168.10.30;tag=as5fd1474e
Call-ID: b_4ff90-960ef774e6d93ff_I@192.168.10.28
CSeq: 12 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP:192.168.10.28:1024 --->
ACK sip:05676067@192.168.10.30 SIP/2.0
From: sip:250@192.168.10.30;tag=26e998114e6894544e6d97ee_F250192.168.10.28
To: sip:05676067@192.168.10.30;tag=as5fd1474e
Call-ID: b_4ff90-960ef774e6d93ff_I@192.168.10.28
CSeq: 12 ACK
Via: SIP/2.0/UDP 192.168.10.28;branch=z9hG4bKc_4ff91-2c4d9e7b4e6d9404_I250
Content-Length: 0
Max-Forwards: 70
User-Agent: Avaya one-X Deskphone
Supported: eventlist, 100rel, replaces


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '72bd9e1a31b7aafe62b7fd5e70d1d73a@172.17.99.41' Method: INVITE

<--- SIP read from UDP:217.30.246.164:5060 --->
OPTIONS sip:172.17.99.41:5060 SIP/2.0
Call-ID: f4u9d-nmbyx@217.30.246.164
CSeq: 198 OPTIONS
From: <sip:anonymous.invalid@217.30.246.164:5060>;tag=09ug41bnau
To: <sip:anonymous.invalid@172.17.99.41:5060>
Via: SIP/2.0/UDP 217.30.246.164:5060;branch=z9hG4bK-w9wac-emlpx
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Looking for s in office (domain 172.17.99.41)

<--- Transmitting (no NAT) to 217.30.246.164:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.30.246.164:5060;branch=z9hG4bK-w9wac-emlpx;received=217.30.246.164
From: <sip:anonymous.invalid@217.30.246.164:5060>;tag=09ug41bnau
To: <sip:anonymous.invalid@172.17.99.41:5060>;tag=as6a1cf1bb
Call-ID: f4u9d-nmbyx@217.30.246.164
CSeq: 198 OPTIONS
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'f4u9d-nmbyx@217.30.246.164' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog 'sn3ij-cskxn@217.30.246.164' Method: OPTIONS

<--- SIP read from UDP:192.168.10.23:5060 --->

в /var/log/asterisk/message

[Sep  8 13:09:11] WARNING[6088] res_config_ldap.c: No directory user found, anonymous binding as default.
[Sep  8 13:09:11] ERROR[6088] res_config_ldap.c: No directory URL or host found.
[Sep  8 13:09:11] NOTICE[6088] res_config_ldap.c: Cannot reload LDAP RealTime driver.
[Sep  8 13:09:11] NOTICE[6088] pbx_ael.c: Starting AEL load process.
[Sep  8 13:09:11] NOTICE[6088] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Sep  8 13:09:11] NOTICE[6088] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Sep  8 13:09:11] NOTICE[6088] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Sep  8 13:09:11] NOTICE[6088] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Sep  8 13:09:11] NOTICE[6088] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
[Sep  8 13:09:11] NOTICE[6088] app_meetme.c: A reload of the SLA configuration has been requested and will be completed when the system is idle.
[Sep  8 13:09:11] WARNING[6088] pbx_config.c: ==!!== Unknown directive: static at line 42 -- IGNORING!!!
[Sep  8 13:09:11] WARNING[6088] pbx_config.c: ==!!== Unknown directive: writeprotect at line 47 -- IGNORING!!!
[Sep  8 13:09:11] WARNING[6088] pbx_config.c: ==!!== Unknown directive: clearglobalvars at line 109 -- IGNORING!!!
[Sep  8 13:09:11] NOTICE[6088] app_queue.c: queuerules.conf has not changed since it was last loaded. Not taking any action.
[Sep  8 13:09:11] NOTICE[6088] chan_skinny.c: Configuring skinny from skinny.conf
[Sep  8 13:09:25] WARNING[5416] chan_sip.c: Received response: "Forbidden" from '"Adel" <sip:250@172.17.99.41>;tag=as2f3d5e9e'
[Sep  8 13:09:50] NOTICE[5416] chan_sip.c: Correct auth, but based on stale nonce received from '"250"<sip:250@192.168.10.30;transport=UDP>;tag=as2ab69725'
[Sep  8 13:09:50] NOTICE[5416] chan_sip.c: Correct auth, but based on stale nonce received from '"202"<sip:202@192.168.10.30;transport=UDP>;tag=as283a6ec7'

Народ помогите кто чем может =))

удалить закрыть спам изменить тег редактировать

спросил 2011-09-08 12:53:02 +0400

hoaxer Gravatar hoaxer
121 12 3 11

2 Ответа

3

поставь в описании транка fromuser=5673998 и бут те счастье, а так он на твой внутренний номер матюкается "Какой-такой sip:250@172.17.99.41? Незнаю... Пошел на фиг!" :-)

ссылка удалить спам редактировать

ответил 2011-09-08 13:11:22 +0400

CheeZ Gravatar CheeZ
1055 6 6 24

обновил 2011-09-08 13:12:57 +0400

Comments

поставил... так же короткие гудки .. может у провайдера что то ??? hoaxer ( 2011-09-08 13:24:24 +0400 )редактировать
точно не знаю из за чего но процентов 99 что это из за fromuser=5673998... хотя до этого 1000 раз так пробовал... =))) спасибо !!! hoaxer ( 2011-09-08 13:30:05 +0400 )редактировать
0

Уряяяяяя получилось !!!!

ссылка удалить спам редактировать

ответил 2011-09-08 13:28:55 +0400

hoaxer Gravatar hoaxer
121 12 3 11

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Задан: 2011-09-08 12:53:02 +0400

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Обновлен: Sep 08 '11

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.