sip show registry пустой, конфиг прописан правильно проверял уже 100 раз, не могу разобратся в чем дело... Firewall выключен! у кого проблема бы ла такая помогите! такое ощущение что порт 5060 зарезан, но у меня выделенка DSL с внешним IP даже подключал без routera все равно линк не поднимается, провайдер Sipgate, но это не важно другие тоже не поднимаются.((

1) reload пробовал делать?
2) покажи sip.conf

da kagdyi raz delau reload.
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preferenc
allow=g729
allow=gsm
allow=slinear
context = error
qualify=no
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear


register => name:passwd@sipgate.de/name



[sipgate-out]
type=friend
insecure=invite
nat=yes
username=name
fromuser=name
fromdomain=sipgate.de
secret=passwd
host=sipgate.de
qualify=yes
canreinvite=no
dtmfmode=rfc2833
context = from-sipgate
;outboundproxy=proxy.provider.domain
;port=80

sip set debug peer sipgate-out
в момент регистрации о чем говорит?

Reliably Transmitting (NAT) to 217.10.79.9:5060:
OPTIONS sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK78694ec6;rport
From: "asterisk" <sip:asterisk@192.168.0.102>;tag=as51ca1171
To: <sip:sipgate.de>
Contact: <sip:asterisk@192.168.0.102>
Call-ID: 11b79f1213f876b9594f7a1315fa5742@192.168.0.102
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 24 Aug 2009 15:40:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
sipip-ubuntu*CLI>
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 200 OK
Record-Route: <sip:217.10.79.9;lr=on;ftag=as51ca1171>
Via: SIP/2.0/UDP 192.168.0.102:5060;received=87.177.242.137;branch=z9hG4bK78694ec6;rport=61576
From: "asterisk" <sip:asterisk@192.168.0.102>;tag=as51ca1171
To: <sip:sipgate.de>;tag=fe1721141f05bd30d4b50c70da3ae228.0d80
Call-ID: 11b79f1213f876b9594f7a1315fa5742@192.168.0.102
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '11b79f1213f876b9594f7a1315fa5742@192.168.0.102' Method: OPTIONS

и это сообщение вываливается с перидичностью в 1 минуту

это отсылается "пинг" и возвращается ответ.
Нужны сообщения после REGISTER. Сделай sip reload и они должны появиться

ubuntu*CLI> reload
[Aug 24 17:58:35] NOTICE[4103]: cdr.c:1373 do_reload: CDR simple logging enabled.
[Aug 24 17:58:35] NOTICE[4103]: res_odbc.c:578 reload: Adding ENV var: INFORMIXSERVER=my_special_database
[Aug 24 17:58:35] NOTICE[4103]: res_odbc.c:578 reload: Adding ENV var: INFORMIXDIR=/opt/informix
[Aug 24 17:58:35] NOTICE[4103]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us'
[Aug 24 17:58:35] NOTICE[4103]: app_playback.c:457 reload: Reloading say.conf
[Aug 24 17:58:35] WARNING[4103]: chan_zap.c:12594 process_zap: Ignoring switchtype
[Aug 24 17:58:35] WARNING[4103]: chan_zap.c:12594 process_zap: Ignoring signalling
[Aug 24 17:58:35] WARNING[4103]: chan_zap.c:12594 process_zap: Ignoring rxwink
[Aug 24 17:58:35] NOTICE[4103]: pbx_ael.c:4131 pbx_load_module: Starting AEL load process.
[Aug 24 17:58:35] NOTICE[4103]: pbx_ael.c:4138 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.
[Aug 24 17:58:35] NOTICE[4103]: pbx_ael.c:4146 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Aug 24 17:58:35] NOTICE[4103]: pbx_ael.c:4149 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
[Aug 24 17:58:35] NOTICE[4103]: pbx_ael.c:4151 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Aug 24 17:58:35] NOTICE[4103]: pbx_ael.c:4154 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Aug 24 17:58:35] NOTICE[4103]: pbx_ael.c:4157 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
Reliably Transmitting (NAT) to 217.10.79.9:5060:
OPTIONS sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK67db266e;rport
From: "asterisk" <sip:asterisk@192.168.0.102>;tag=as666a8f53
To: <sip:sipgate.de>
Contact: <sip:asterisk@192.168.0.102>
Call-ID: 23fe686865e844eb61fed7874a5d7fc1@192.168.0.102
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 24 Aug 2009 15:58:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
sipip-ubuntu*CLI>
<--- SIP read from 217.10.79.9:5060 --->
SIP/2.0 200 OK
Record-Route: <sip:217.10.79.9;lr=on;ftag=as666a8f53>
Via: SIP/2.0/UDP 192.168.0.102:5060;received=87.177.242.137;branch=z9hG4bK67db266e;rport=61576
From: "asterisk" <sip:asterisk@192.168.0.102>;tag=as666a8f53
To: <sip:sipgate.de>;tag=fe1721141f05bd30d4b50c70da3ae228.e600
Call-ID: 23fe686865e844eb61fed7874a5d7fc1@192.168.0.102
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '23fe686865e844eb61fed7874a5d7fc1@192.168.0.102' Method: OPTIONS

вот пожалуйста

что за "билеберда"?

Если прописано это (в general):

register => name:passwd@sipgate.de/name

то CLI> sip show registry - не может быть пустым ...

Приведите вывод этой (sip show registry) команды сюда.

И делайте 'sip reload' вместо полного 'reload' ...