Регистрация не проходит

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 =))