Прошу помощи. Избитая тема, вроде все облазил. Не могу понять в чем дело. Пытаюсь настроить fxs. Набираю на аппарате номер, никакой реакции, сигнал ответа станции не прекращается. В чем дело?


имеется
1 астерикс ip 192.168.0.50
1 lynksys 3102 ip 192.168.0.1
все в локальной сети
к fxs подключен телефон (тоновый набор)

настройка linksys 3102
Оставил заводские настройки +
на вкладке line 1 поменял
user id = spa3102
proxy = 192.168.0.50
dial plan = (S0<:100>)


файл конфигурации extensions.conf Код:
[phones]
exten => 100,1,Verbose(1/Echo test application)
exten => 100,n,Echo()
exten => 100,n,Hangup()
exten => 200,1,Answer()
exten => 200,n,Playback(dir-intro-oper)
exten => 200,n,Hangup()




файл конфигурации sip.conf Код:
[general]
[spa3102]
context=phones
type=friend
host=dynamic





вот выкладка debug Код:
NOTICE
[May 14 09:13:13] NOTICE[2985] chan_skinny.c: Unable to load config skinny.conf, Skinny disabled
[May 14 10:29:57] NOTICE[3040] cdr.c: CDR simple logging enabled.
[May 14 10:29:57] NOTICE[3040] loader.c: 144 modules will be loaded.
[May 14 10:29:58] NOTICE[3040] res_smdi.c: Unable to load config smdi.conf: SMDI disabled
[May 14 10:29:58] NOTICE[3040] chan_mgcp.c: Unable to load config mgcp.conf, MGCP disabled
[May 14 10:29:58] NOTICE[3040] chan_agent.c: No agent configuration found -- agent support disabled
[May 14 10:29:59] NOTICE[3040] pbx_ael.c: Starting AEL load process.
[May 14 10:29:59] NOTICE[3040] pbx_ael.c: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.
[May 14 10:29:59] NOTICE[3040] pbx_ael.c: File /etc/asterisk/extensions.ael not found; AEL declining load
[May 14 10:29:59] NOTICE[3040] chan_skinny.c: Unable to load config skinny.conf, Skinny disabled
DEBUG
[May 13 09:56:37] DEBUG[2965] devicestate.c: Changing state for SIP/spa3102 - state 1 (Not in use)
[May 13 09:56:37] DEBUG[2970] app_queue.c: Device 'SIP/spa3102' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 0: REGISTER sip:192.168.0.50 SIP/2.0 (33)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 1: Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-fc7a478d (55)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 2: From: spa3102 <sip:spa3102@192.168.0.50>;tag=7176c522e93f6439o1 (63)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 3: To: spa3102 <sip:spa3102@192.168.0.50> (38)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 4: Call-ID: b55d3808-bc661a3a@127.0.0.1 (36)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 5: CSeq: 22095 REGISTER (20)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 6: Max-Forwards: 70 (16)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 7: Contact: spa3102 <sip:spa3102@127.0.0.1:5061>;expires=3600 (58)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 8: User-Agent: Linksys/SPA3102-3.3.6(GW) (37)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 9: Content-Length: 0 (17)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 10: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 11: Supported: x-sipura (19)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Header 12: (0)
[May 13 09:56:37] DEBUG[2972] chan_sip.c: = Found Their Call ID: b55d3808-bc661a3a@127.0.0.1 Their Tag 7176c522e93f6439o1 Our tag: as3700b5a1
[May 13 09:56:37] DEBUG[2972] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER
[May 13 09:56:37] DEBUG[2972] chan_sip.c: Ignoring SIP message because of retransmit (REGISTER Seqno 22095, ours 22095)
[May 13 09:56:37] DEBUG[2972] devicestate.c: Notification of state change to be queued on device/channel SIP/spa3102
[May 13 09:56:37] DEBUG[2965] devicestate.c: No provider found, checking channel drivers for SIP - spa3102
[May 13 09:56:37] DEBUG[2965] chan_sip.c: Checking device state for peer spa3102
[May 13 09:56:37] DEBUG[2965] devicestate.c: Changing state for SIP/spa3102 - state 1 (Not in use)
[May 13 09:56:37] DEBUG[2970] app_queue.c: Device 'SIP/spa3102' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 0: REGISTER sip:192.168.0.50 SIP/2.0 (33)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 1: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-e7f0c08b (55)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 2: From: spa3102 <sip:spa3102@192.168.0.50>;tag=6e1eabea45576c11o0 (63)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 3: To: spa3102 <sip:spa3102@192.168.0.50> (38)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 4: Call-ID: e276418a-2f9bcea0@127.0.0.1 (36)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 5: CSeq: 48497 REGISTER (20)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 6: Max-Forwards: 70 (16)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 7: Contact: spa3102 <sip:spa3102@127.0.0.1:5060>;expires=3600 (58)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 8: User-Agent: Linksys/SPA3102-3.3.6(GW) (37)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 9: Content-Length: 0 (17)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 10: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 11: Supported: x-sipura (19)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 12: (0)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: = No match Their Call ID: b55d3808-bc661a3a@127.0.0.1 Their Tag 7176c522e93f6439o1 Our tag: as3700b5a1
[May 13 09:56:41] DEBUG[2972] chan_sip.c: = Found Their Call ID: e276418a-2f9bcea0@127.0.0.1 Their Tag 6e1eabea45576c11o0 Our tag: as7181636a
[May 13 09:56:41] DEBUG[2972] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Ignoring SIP message because of retransmit (REGISTER Seqno 48497, ours 48497)
[May 13 09:56:41] DEBUG[2972] devicestate.c: Notification of state change to be queued on device/channel SIP/spa3102
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 0: SIP/2.0 100 Trying (18)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 1: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-e7f0c08b;received=192.168.0.1 (76)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 2: From: spa3102 <sip:spa3102@192.168.0.50>;tag=6e1eabea45576c11o0 (63)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 3: To: spa3102 <sip:spa3102@192.168.0.50> (38)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 4: Call-ID: e276418a-2f9bcea0@127.0.0.1 (36)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 5: CSeq: 48497 REGISTER (20)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 8: Supported: replaces (19)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 9: Content-Length: 0 (17)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 10: (0)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: = No match Their Call ID: b55d3808-bc661a3a@127.0.0.1 Their Tag 7176c522e93f6439o1 Our tag: as3700b5a1
[May 13 09:56:41] DEBUG[2972] chan_sip.c: = Found Their Call ID: e276418a-2f9bcea0@127.0.0.1 Their Tag 6e1eabea45576c11o0 Our tag: as7181636a
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Ignoring out of order response 48497 (expecting 101)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: SIP message could not be handled, bad request: e276418a-2f9bcea0@127.0.0.1
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 0: SIP/2.0 200 OK (14)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 1: Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-e7f0c08b;received=192.168.0.1 (76)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 2: From: spa3102 <sip:spa3102@192.168.0.50>;tag=6e1eabea45576c11o0 (63)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 3: To: spa3102 <sip:spa3102@192.168.0.50>;tag=as7181636a (53)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 4: Call-ID: e276418a-2f9bcea0@127.0.0.1 (36)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 5: CSeq: 48497 REGISTER (20)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 6: User-Agent: Asterisk PBX (24)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 7: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 8: Supported: replaces (19)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 9: Expires: 3600 (13)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 10: Contact: <sip:spa3102@127.0.0.1:5060>;expires=3600 (50)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 11: Date: Thu, 13 May 2010 05:56:41 GMT (35)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 12: Content-Length: 0 (17)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 13: (0)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: = No match Their Call ID: b55d3808-bc661a3a@127.0.0.1 Their Tag 7176c522e93f6439o1 Our tag: as3700b5a1
[May 13 09:56:41] DEBUG[2972] chan_sip.c: = Found Their Call ID: e276418a-2f9bcea0@127.0.0.1 Their Tag 6e1eabea45576c11o0 Our tag: as7181636a
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Ignoring out of order response 48497 (expecting 101)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: SIP message could not be handled, bad request: e276418a-2f9bcea0@127.0.0.1
[May 13 09:56:41] DEBUG[2965] devicestate.c: No provider found, checking channel drivers for SIP - spa3102
[May 13 09:56:41] DEBUG[2965] chan_sip.c: Checking device state for peer spa3102
[May 13 09:56:41] DEBUG[2965] devicestate.c: Changing state for SIP/spa3102 - state 1 (Not in use)
[May 13 09:56:41] DEBUG[2970] app_queue.c: Device 'SIP/spa3102' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 0: REGISTER sip:192.168.0.50 SIP/2.0 (33)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 1: Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-fc7a478d (55)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 2: From: spa3102 <sip:spa3102@192.168.0.50>;tag=7176c522e93f6439o1 (63)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 3: To: spa3102 <sip:spa3102@192.168.0.50> (38)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 4: Call-ID: b55d3808-bc661a3a@127.0.0.1 (36)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 5: CSeq: 22095 REGISTER (20)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 6: Max-Forwards: 70 (16)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 7: Contact: spa3102 <sip:spa3102@127.0.0.1:5061>;expires=3600 (58)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 8: User-Agent: Linksys/SPA3102-3.3.6(GW) (37)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 9: Content-Length: 0 (17)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 10: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 11: Supported: x-sipura (19)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Header 12: (0)
[May 13 09:56:41] DEBUG[2972] chan_sip.c: = Found Their Call ID: b55d3808-bc661a3a@127.0.0.1 Their Tag 7176c522e93f6439o1 Our tag: as3700b5a1
[May 13 09:56:41] DEBUG[2972] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER
[May 13 09:56:41] DEBUG[2972] chan_sip.c: Ignoring SIP message because of retransmit (REGISTER Seqno 22095, ours 22095)
[May 13 09:56:41] DEBUG[2972] devicestate.c: Notification of state change to be queued on device/channel SIP/spa3102
[May 13 09:56:41] DEBUG[2965] devicestate.c: No provider found, checking channel drivers for SIP - spa3102
[May 13 09:56:41] DEBUG[2965] chan_sip.c: Checking device state for peer spa3102
[May 13 09:56:41] DEBUG[2965] devicestate.c: Changing state for SIP/spa3102 - state 1 (Not in use)
[May 13 09:56:41] DEBUG[2970] app_queue.c: Device 'SIP/spa3102' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[May 13 09:56:43] DEBUG[2975] asterisk.c: Asterisk ending (0).
VERBOSE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:spa3102@127.0.0.1:5060>;expires=3600
Date: Fri, 14 May 2010 06:36:38 GMT
Content-Length: 0


<------------->
[May 14 10:36:38] VERBOSE[3054] logger.c: --- (13 headers 0 lines) ---
[May 14 10:36:42] VERBOSE[3054] logger.c:
<--- SIP read from 192.168.0.1:5060 --->
REGISTER sip:192.168.0.50 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-851d3107
From: spa3102 <sip:spa3102@192.168.0.50>;tag=595f6c7da095ecb0o0
To: spa3102 <sip:spa3102@192.168.0.50>
Call-ID: e276418a-2f9bcea0@127.0.0.1
CSeq: 49912 REGISTER
Max-Forwards: 70
Contact: spa3102 <sip:spa3102@127.0.0.1:5060>;expires=3600
User-Agent: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


<------------->
[May 14 10:36:42] VERBOSE[3054] logger.c: --- (12 headers 0 lines) ---
[May 14 10:36:42] VERBOSE[3054] logger.c: Using latest REGISTER request as basis request
[May 14 10:36:42] VERBOSE[3054] logger.c: Sending to 127.0.0.1 : 5060 (no NAT)
[May 14 10:36:42] VERBOSE[3054] logger.c:
<--- Transmitting (no NAT) to 127.0.0.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-851d3107;received=192.168.0.1
From: spa3102 <sip:spa3102@192.168.0.50>;tag=595f6c7da095ecb0o0
To: spa3102 <sip:spa3102@192.168.0.50>
Call-ID: e276418a-2f9bcea0@127.0.0.1
CSeq: 49912 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[May 14 10:36:42] VERBOSE[3054] logger.c:
<--- Transmitting (no NAT) to 127.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-851d3107;received=192.168.0.1
From: spa3102 <sip:spa3102@192.168.0.50>;tag=595f6c7da095ecb0o0
To: spa3102 <sip:spa3102@192.168.0.50>;tag=as603325e5
Call-ID: e276418a-2f9bcea0@127.0.0.1
CSeq: 49912 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:spa3102@127.0.0.1:5060>;expires=3600
Date: Fri, 14 May 2010 06:36:42 GMT
Content-Length: 0


<------------>
[May 14 10:36:42] VERBOSE[3054] logger.c: Scheduling destruction of SIP dialog 'e276418a-2f9bcea0@127.0.0.1' in 32000 ms (Method: REGISTER)
[May 14 10:36:42] VERBOSE[3054] logger.c:
<--- SIP read from 127.0.0.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-851d3107;received=192.168.0.1
From: spa3102 <sip:spa3102@192.168.0.50>;tag=595f6c7da095ecb0o0
To: spa3102 <sip:spa3102@192.168.0.50>
Call-ID: e276418a-2f9bcea0@127.0.0.1
CSeq: 49912 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------->
[May 14 10:36:42] VERBOSE[3054] logger.c: --- (10 headers 0 lines) ---
[May 14 10:36:42] VERBOSE[3054] logger.c:
<--- SIP read from 127.0.0.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-851d3107;received=192.168.0.1
From: spa3102 <sip:spa3102@192.168.0.50>;tag=595f6c7da095ecb0o0
To: spa3102 <sip:spa3102@192.168.0.50>;tag=as603325e5
Call-ID: e276418a-2f9bcea0@127.0.0.1
CSeq: 49912 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:spa3102@127.0.0.1:5060>;expires=3600
Date: Fri, 14 May 2010 06:36:42 GMT
Content-Length: 0


<------------->
[May 14 10:36:42] VERBOSE[3054] logger.c: --- (13 headers 0 lines) ---
[May 14 10:36:47] VERBOSE[3192] logger.c: -- Remote UNIX connection disconnected

Как я понимаю последовательность пакетов такова, что сначала linksys справшивает регистрацию в поле contact пишет spa3102@127.0.0.1 Потом астериск начинает отправлять ответ (обработка запроса) и пакет (успешная регистрация) на адрес 127.0.0.1 Откуда взялся этот 127.0.0.1 если ip астерикса 192.168.0.50 у linksys ip 192.168.0.1? когда поднимаю трубку телефона подключенного к fsx, никаких пакетов связянных с вызовом не вижу....блин..не понимаю...

sip show peers что показывает?

sip show peers
вывод
name/surname : 3102
host : (unspecified)
dyn
nat
acl
port : 5060
status : unmonitored

name/surname : spa3102/spa3102
host : 127.0.0.1
dyn : D
nat
acl
port : 5060
status : unmonitored


sip show registry
пусто

1) sip show peers имеет не такой вывод
2)

name/surname : spa3102/spa3102
host : 127.0.0.1

разве это не настораживает?
3) нет сил читать километровые дебаги! Надо стараться учиться из кучи мусора выделять главное.

user id = spa3102 сменить на номер, который будет иметь этот внутренний экстеншн. И в соответствии с этим номером (например - 1001_
sip.conf Код:

[1001]
type=friend
host=dynamic
secret= уж нужно ввести пароль, которым spa3102 должен регистрироваться, он ведь dynamic!
context=phones

Нет. Я переписал вручную. Выводит в правильной форме. Не знаю, как копировать в файл вывод sip show peers из cli

ded:

3) нет сил читать километровые дебаги! Надо стараться учиться из кучи мусора выделять главное.

Согласен. как я понимаю, странно следующее
Приход запрос на регистрацию от Линсиса к Астериксу
<--- SIP read from 192.168.0.1:5060 --->
REGISTER sip:192.168.0.50 SIP/2.0
В поле пакета
Contact: spa3102 <sip:1001@127.0.0.1:5060>;expires=3600
Линсис оставляет свой обратный адрес 127.0.0.1

Потом Астерикс начинает отправлять ответы на 127.0.0.1

<--- Transmitting (no NAT) to 127.0.0.1:5060 --->
SIP/2.0 100 Trying
и
<--- Transmitting (no NAT) to 127.0.0.1:5060 --->
SIP/2.0 401 Unauthorized

Мне непонятно, как поменять 127.0.0.1 на 192.168.0.1 (адрес Линксиса), может из-за этого он не может зарегистрироваться и позвонить

ded:

user id = spa3102 сменить на номер, который будет иметь этот внутренний экстеншн. И в соответствии с этим номером (например - 1001_
sip.conf Код:

[1001]
type=friend
host=dynamic
secret= уж нужно ввести пароль, которым spa3102 должен регистрироваться, он ведь dynamic!
context=phones

Поменял, вот новые настройки, но позвонить с телефона подключенного к fxs Линсиса не могу,
sip show peers показыват
name/surname : 1001/1001
host : (unspecified)
dyn : D
nat
acl
port : 0
status : unmonitored

файл 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()

[phones]
exten => 100,1,Answer()
exten => 100,n,Playback(tt-weasels)
exten => 100,n,Hangup()

файл sip.conf
[general]
[1001]
context=phones
type=friend
host=dynamic
secret=123
username=1001
nat=no

в Линксисе на вкладке line1 прописал
password 123
user id 1001
auth id 1001
proxy = 192.168.0.50
dial plan = (S0<:100>)

name/surname : 1001/1001
host : (unspecified) Линксис не зарегистрировался.
Вывод команд оформлять тагами [code [\code
Скобки специально не закрыты.

Если запрос на регистрацию от Линсиса к Астериксу
<--- SIP read from 192.168.0.1:5060 --->
REGISTER sip:192.168.0.50 SIP/2.0
В поле пакета
Contact: spa3102 <sip:1001@127.0.0.1:5060>;expires=3600
то отправлять ответ Астериск будет на 192.168.0.1 а не на 127.0.0.1

Вы уж разберитесь с этим? В итернете полно инструкций по настройке 3102.

Вот например одна из таких настроек
http://docs.google.com/View?docid=dmqx96v_3qk8phn


Я неделю разбираюсь, никак не получается, может дело в диалплане линксиса?

народ, помогите, с меня пиво на охотном ряду )

Может. И эта тема (дело в диалплане линксиса) тоже раскрывалась.

Вот хороший ресурс по диалплану Линксиса.
http://igorg.ru/2008/04/03/linksys_dialplan/

Я свой диалплан взял из руководства Линксиса

 (S0 <:100>)

хотлайн
Но, все равно, не работает...блин