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NGN U-tel и Asterisk 1.6

Asterisk IP PBX 14 сообщений 02.12.2009 20:54 - 29.12.2010 04:28
#1 02.12.2009 20:54

NGN U-tel и Asterisk 1.6

Господа, подскажете, кто нибудь скрешивал * и NGN от U-tel?
У нас проблема - не регистрируется * и соответственно не принимает и не отправляет звонки.
#2 02.12.2009 22:13
Теоретически все, что поддерживает sip (rfc3261) должно скрещиваться.
Практически - бывает, что возникает проблемы. Проблемы интерконнекта люди обычно либо решают сами, либо постят на тематических форумах дампы сигнализации, сопровождаемые дебагом их телефонных станций и комментариями по топологии сети
#3 03.12.2009 06:35
да
Code:

register => YYYYYY:*****@chel.media.usi.ru/XXXXXX

[usi.ru]
type=friend
host=chel.media.usi.ru
username=YYYYY
fromuser=YYYYY
fromdomain=chel.media.usi.ru
outboundproxy=62.148.237.132
secret=*****
context=inbuh
insecure = invite
dtmfmode = rfc2833


Думаю с этим должно завестись....
ТОлько учтите что если будите очень часто производить подключения то уси может заблокировать ваш ак. примерно на пол часа
#4 08.12.2009 14:36
В тех поддержке NGN сослались на неверную строку Invite при исходящем звонке!
Типа по умолчанию * ставит туда IP, а должен hostname!
Какисправить?
#5 08.12.2009 16:02
externhost ?
_________________
Ubuntu-Server 6.06 /Gentoo /Asterisk 1.4.21.1/app_fax(t38) /spandsp-0.0.5pre4
Digium TDM400/Polycom IP301 SP /Snom 360/ Seiros ТИ24/Linksys 9X/ Kirk 600IPv3

Last edited by Cache on Tue Dec 08, 2009 19:22
#6 08.12.2009 17:00
может: fromdomain=chel.media.usi.ru или host=chel.media.usi.ru???
У меня работает при тех настройках, что я вам выложил Rolling Eyes

А входящие идут???
может в нагрузку еще нат?
#7 13.12.2009 10:24
Проблема была в блокировке акаунта со стороны провайдера.
Таймаут со стороны УСИ - 180 с со стороны * - по-моему - 120 с.

Added after 1 hours 23 minutes:

Кстати вопрос!
Как правильно * указать таймаут для регистрации?
#8 14.12.2009 09:25
Господа, поднимаю тему!
Как изменить таймаут сесии регистрации на провайдере?
#9 15.01.2010 13:07

Еще одна проблема с u-tel

Добрый день, возникла такая проблема. Не получается совершить звонок, техподдержка ругается на то, что
Contact: должно быть
Contact: .
Code:
Audio is at 90.157.12.137 port 56492
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
INVITE sip:2047399@ektngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 90.157.12.137:5060;branch=z9hG4bK5c2f32b2;rport
Max-Forwards: 70
From: "2047399" ;tag=as0f9fb954
To:
Contact:
Call-ID: 66df81ca35b187011b614d3e251f7e90@ektngn.usi.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Remote-Party-ID: "2047399" ;privacy=off;screen=no
Date: Fri, 15 Jan 2010 11:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 322

v=0
o=root 2027202349 2027202349 IN IP4 192.168.0.46
s=Platinum PBX
c=IN IP4 192.168.0.46
t=0 0
m=audio 47986 RTP/AVP 8 18 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
ubuntu*CLI>

SIP/2.0 100 Trying
From: "2047399";tag=as0f9fb954
To:
Call-ID: 66df81ca35b187011b614d3e251f7e90@ektngn.usi.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 90.157.12.137:5060;rport=5060;branch=z9hG4bK5c2f32b2
Content-Length: 0



--- (7 headers 0 lines) ---
ubuntu*CLI>

SIP/2.0 500 Server Internal Error
From: "2047399";tag=as0f9fb954
To: ;tag=2093856481
Call-ID: 66df81ca35b187011b614d3e251f7e90@ektngn.usi.ru
CSeq: 102 INVITE
Via: SIP/2.0/UDP 90.157.12.137:5060;rport=5060;branch=z9hG4bK5c2f32b2
contact:
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0



--- (9 headers 0 lines) ---
Transmitting (no NAT) to 62.148.237.152:5060:
ACK sip:2047399@ektngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 90.157.12.137:5060;branch=z9hG4bK5c2f32b2;rport
Max-Forwards: 70
From: "2047399" ;tag=as0f9fb954
To: ;tag=2093856481
Contact:
Call-ID: 66df81ca35b187011b614d3e251f7e90@ektngn.usi.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX
Remote-Party-ID: "2047399" ;privacy=off;screen=no
Content-Length: 0

Никак не могу найти как это исправить...
transargo0 звонит с 192,168,0,46 помощью пира на ektngn.usi.ru
[usi.ru]
Code:

[transargo0]
type=friend
username=transargo0
nat=yes
secret=321123
callerid=""
host=dynamic
disallow=all
allow=alaw,gsm
context=3850000_out
dtmfmode=auto;
[usi.ru]
type=friend
host=ektngn.usi.ru
username=xxx
fromuser=xxx
fromdomain=ektngn.usi.ru
outboundproxy=62.148.237.152
secret=yyy
context=default
contact=transargo00@ektngn.usi.ru;???
insecure = invite
dtmfmode = rfc2833
nat=no
canreinvite=nonat
reinvite=no

Входящие звонки проходят, регистрация срабатывает без проблем.
#10 16.01.2010 23:52
Quote:

техподдержка ругается на то, что:
Contact:
должно быть
Contact: .


С какого разу она ругается? Поле Contact - личное дело шлюза или терминала. Там можно указать "tel:номер бабушки" правда, при этом, бабушке придется отвечать на все blind transfer Smile
А тех поддержка не просила посмотреть на лампочки? Smile
_________________
ys
http://voip.rus.net/
#11 17.01.2010 15:12
ругается при исходящих... То, что asterisk выдает логичную штуку понятно... Регистрация проходит нормально, так скажем регистрирует то, что надо...
:Contact:
Code:

REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:ektngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 90.157.12.137:5060;branch=z9hG4bK049657bb;rport
Max-Forwards: 70
From: ;tag=as38802d20
To:
Call-ID: 532e392c1fe3d5ef7c6ea236297c9bc6@127.0.1.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact:
Content-Length: 0


---
ubuntu*CLI>

SIP/2.0 100 Trying
From: ;tag=as38802d20
To:
Call-ID: 532e392c1fe3d5ef7c6ea236297c9bc6@127.0.1.1
CSeq: 102 REGISTER
Via: SIP/2.0/UDP 90.157.12.137:5060;rport=5060;branch=z9hG4bK049657bb
Content-Length: 0



--- (7 headers 0 lines) ---
ubuntu*CLI>

SIP/2.0 407 Proxy Authentication Required
From: ;tag=as38802d20
To: ;tag=642077299
Call-ID: 532e392c1fe3d5ef7c6ea236297c9bc6@127.0.1.1
CSeq: 102 REGISTER
Via: SIP/2.0/UDP 90.157.12.137:5060;rport=5060;branch=z9hG4bK049657bb
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
proxy-authenticate: Digest realm="Realm",nonce="MTI2MzczNzA2NTg1NmQwMjA1YjY1MzM4MTY0N2UzOGQ3MzczOTE5YmIwMjM4",stale=false,algorithm=MD5,qop="auth,auth-int"
Content-Length: 0



--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name ektngn.usi.ru
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:ektngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 90.157.12.137:5060;branch=z9hG4bK7a97a1c4;rport
Max-Forwards: 70
From: ;tag=as6289d295
To:
Call-ID: 532e392c1fe3d5ef7c6ea236297c9bc6@127.0.1.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="transargo00", realm="Realm", algorithm=MD5, uri="sip:ektngn.usi.ru", nonce="MTI2MzczNzA2NTg1NmQwMjA1YjY1MzM4MTY0N2UzOGQ3MzczOTE5YmIwMjM4", response="0e10813d39929792658bd1f6455d1d18", qop=auth, cnonce="0d9f35a0", nc=00000001
Expires: 120
Contact:
Content-Length: 0


---
ubuntu*CLI>

SIP/2.0 100 Trying
From: ;tag=as6289d295
To:
Call-ID: 532e392c1fe3d5ef7c6ea236297c9bc6@127.0.1.1
CSeq: 103 REGISTER
Via: SIP/2.0/UDP 90.157.12.137:5060;rport=5060;branch=z9hG4bK7a97a1c4
Content-Length: 0



--- (7 headers 0 lines) ---
ubuntu*CLI>

SIP/2.0 200 Registration Successful
From: "transargo00 transargo00";tag=as6289d295
To: ;tag=1333885784
Call-ID: 532e392c1fe3d5ef7c6ea236297c9bc6@127.0.1.1
CSeq: 103 REGISTER
Via: SIP/2.0/UDP 90.157.12.137:5060;rport=5060;branch=z9hG4bK7a97a1c4
contact: ;expires=112
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0




Но ошибка 500 выдается при исходящих, лог из которых был постом ранее. Самое интересное, что такая ошибка стала возникать после того, как они перезагрузили сервер свой... Изначально все работало прекрасно. УСИшкини, помогите![/b]
#12 04.02.2010 15:42
DIX:
Quote:
...Как я понял там был хак, который при перезагрузке слетел и заставил ektngn выдавать ошибку 500. Так что если не хочешь портить себе нервы выстави sendrpid=no.
#13 28.12.2010 13:12
У меня такая же проблема, с IP телефона конектится и звонится. А еластрикс не хочет звонить ни принимать.
Code:
voip*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
nugngn.usi.ru:5060 N 201901 105 Registered Sat, 25 Dec 2010 11:05:16
1 SIP registrations.

Как понял что работает.
В общем при звонке с улице получаю вот что:

Code:
CSeq: 5859 REGISTER
Contact:
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0



--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK3034867581766030529;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5859 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0



SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK3034867581766030529;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as5b0036ae
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5859 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="43851e45"
Content-Length: 0


Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)


REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2028716029890824009;rport
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5860 REGISTER
Contact:
Authorization: Digest username="007", realm="asterisk", nonce="43851e45", uri="sip:192.168.100.44:5060", response="f00f064f84229e843d47d7a0768678de", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2028716029890824009;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5860 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK38131ae1;rport
Max-Forwards: 70
From: "Unknown" ;tag=as5e73e59a
To:
Contact:
Call-ID: 1ba0fa5023c0b8a37043b2794ee4e1f2@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 27 Dec 2010 05:35:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2028716029890824009;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as5b0036ae
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5860 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: ;expires=60
Date: Mon, 27 Dec 2010 05:35:38 GMT
Content-Length: 0


Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK38131ae1;rport
From: "Unknown" ;tag=as5e73e59a
To:
Call-ID: 1ba0fa5023c0b8a37043b2794ee4e1f2@192.168.100.44
CSeq: 102 OPTIONS
Contact:
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0


--- (11 headers 0 lines) ---
Really destroying SIP dialog '1ba0fa5023c0b8a37043b2794ee4e1f2@192.168.100.44' Method: OPTIONS
Really destroying SIP dialog '59776e52d291f11b20e85e74777a17eddf018a@62.148.237.152' Method: ACK
Really destroying SIP dialog '321181390410115-2516193436649@192.168.100.68' Method: REGISTER


REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK30780121931304228956;rport
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5861 REGISTER
Contact:
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK30780121931304228956;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5861 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0




SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK30780121931304228956;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as42d175cb
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5861 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3be3adb3"
Content-Length: 0


Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)


REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2959992612496529687;rport
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5862 REGISTER
Contact:
Authorization: Digest username="007", realm="asterisk", nonce="3be3adb3", uri="sip:192.168.100.44:5060", response="7e57f8163fe43922cd90e04b82aa8194", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2959992612496529687;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5862 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK3c8be1d6;rport
Max-Forwards: 70
From: "Unknown" ;tag=as575d4fae
To:
Contact:
Call-ID: 0d1223673bb0aa7d78e8a7737c23797e@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 27 Dec 2010 05:36:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2959992612496529687;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as42d175cb
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5862 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: ;expires=60
Date: Mon, 27 Dec 2010 05:36:36 GMT
Content-Length: 0


Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK3c8be1d6;rport
From: "Unknown" ;tag=as575d4fae
To:
Call-ID: 0d1223673bb0aa7d78e8a7737c23797e@192.168.100.44
CSeq: 102 OPTIONS
Contact:
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0


--- (11 headers 0 lines) ---
Really destroying SIP dialog '0d1223673bb0aa7d78e8a7737c23797e@192.168.100.44' Method: OPTIONS
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:nugngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK751a985a;rport
Max-Forwards: 70
From: ;tag=as6db62f0b
To:
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 106 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Authorization: Digest username="201901", realm="Realm", algorithm=MD5, uri="sip:nugngn.usi.ru", nonce="MTI5MzQyMDc2NTg3MjAxNmMyN2Y1NGI2OWQ4N2E2NjBiMGUxMmI5Njc4OTQw", response="8a54c50dedc33416ba798d6d32e2601e", qop=auth, cnonce="178dc8f6", nc=00000004
Expires: 120
Contact:
Content-Length: 0

---


SIP/2.0 100 Trying
From: ;tag=as6db62f0b
To:
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 106 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK751a985a
Content-Length: 0


--- (7 headers 0 lines) ---


SIP/2.0 407 Proxy Authentication Required
From: ;tag=as6db62f0b
To: ;tag=863400534
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 106 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK751a985a
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
proxy-authenticate: Digest realm="Realm",nonce="MTI5MzQyMTA3NzYyNjIwMzk4NzM1Mjg3ZGIzNjRkNzI2Yjc5YWE4YzRkNjAy",stale=true,algorithm=MD5,qop="auth,auth-int"
Content-Length: 0


--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name nugngn.usi.ru
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:nugngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK15c63bf4;rport
Max-Forwards: 70
From: ;tag=as7fc9a353
To:
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 107 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Proxy-Authorization: Digest username="201901", realm="Realm", algorithm=MD5, uri="sip:nugngn.usi.ru", nonce="MTI5MzQyMTA3NzYyNjIwMzk4NzM1Mjg3ZGIzNjRkNzI2Yjc5YWE4YzRkNjAy", response="27f6f4f1bffb20c6478ff65d313e050b", qop=auth, cnonce="709a908f", nc=00000001
Expires: 120
Contact:
Content-Length: 0
---


SIP/2.0 100 Trying
From: ;tag=as7fc9a353
To:
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 107 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK15c63bf4
Content-Length: 0


--- (7 headers 0 lines) ---


SIP/2.0 200 Registration Successful
From: "201901 201901";tag=as7fc9a353
To: ;tag=266258229
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 107 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK15c63bf4
contact: ;expires=4,;expires=116
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0


--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '321181390410115-2516193436649@192.168.100.68' Method: REGISTER
Really destroying SIP dialog '356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1' Method: REGISTER


REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK1999646311544822368;rport
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5863 REGISTER
Contact:
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK1999646311544822368;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5863 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0




SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK1999646311544822368;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as2c004b17
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5863 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="73839620"
Content-Length: 0


Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)


REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK4286254401256713458;rport
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5864 REGISTER
Contact:
Authorization: Digest username="007", realm="asterisk", nonce="73839620", uri="sip:192.168.100.44:5060", response="806a1fc3ebaeb031db0453b9faeff5c5", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK4286254401256713458;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5864 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK5e4c21e3;rport
Max-Forwards: 70
From: "Unknown" ;tag=as64e5b805
To:
Contact:
Call-ID: 4d69381a4d3064d949b88dc4147cf772@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 27 Dec 2010 05:37:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK4286254401256713458;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as2c004b17
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5864 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: ;expires=60
Date: Mon, 27 Dec 2010 05:37:33 GMT
Content-Length: 0


Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK5e4c21e3;rport
From: "Unknown" ;tag=as64e5b805
To:
Call-ID: 4d69381a4d3064d949b88dc4147cf772@192.168.100.44
CSeq: 102 OPTIONS
Contact:
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0


--- (11 headers 0 lines) ---
Really destroying SIP dialog '4d69381a4d3064d949b88dc4147cf772@192.168.100.44' Method: OPTIONS


Если изнутри:

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;received=192.168.100.68;rport=5060
From: 007 ;tag=2409814001
To: "250823" ;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact:
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1056850601 1056850601 IN IP4 192.168.100.44
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.100.44
t=0 0
m=audio 17154 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Executing [s@macro-outisbusy:2] GotoIf("SIP/007-0000001d", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/007-0000001d", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/007-0000001d", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- Playing 'all-circuits-busy-now.gsm' (language 'en')
Really destroying SIP dialog '32feaa0462b38e4f720d9c7303589d33@192.168.100.44' Method: INVITE
-- Playing 'pls-try-call-later.gsm' (language 'en')


CANCEL sip:250823@192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;rport
From: 007 ;tag=2409814001
To: "250823"
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: Voip Phone 1.0
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)


SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;received=192.168.100.68;rport=5060
From: 007 ;tag=2409814001
To: "250823" ;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0




SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;received=192.168.100.68;rport=5060
From: 007 ;tag=2409814001
To: "250823" ;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 CANCEL
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/007-0000001d' in macro 'outisbusy'
== Spawn extension (from-internal, 250823, 5) exited non-zero on 'SIP/007-0000001d'
-- Executing [h@from-internal:1] Macro("SIP/007-0000001d", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/007-0000001d", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/007-0000001d", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/007-0000001d", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/007-0000001d", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/007-0000001d", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/007-0000001d", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/007-0000001d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/007-0000001d'


ACK sip:250823@192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;rport
From: 007 ;tag=2409814001
To: "250823" ;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0


Ну Это не весь кусок, писанины много, что в терминалке стока строк не отображает.

Added after 42 minutes:

в общем проблему решил методом тыка. работает.
#14 29.12.2010 04:28
вчера работало, сегодня уже нет, звонки проходят через раз, то есть то нету. также на входящий.

Added after 1 minutes:

вот что пишет на входящий звонок.
Code:
Really destroying SIP dialog '72d6d4ab35827331750e6752577ccf87@192.168.100.44' Method: OPTIONS
Really destroying SIP dialog '67672746522226-229372301815489@192.168.100.68' Method: REGISTER


REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK706127485295087026;rport
From: 007 ;tag=974921936
To: 007
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1969 REGISTER
Contact:
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0



--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK706127485295087026;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1969 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0





SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK706127485295087026;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as11634ef1
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1969 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07969da8"
Content-Length: 0



Scheduling destruction of SIP dialog '67672746522226-229372301815489@192.168.100.68' in 32000 ms (Method: REGISTER)


REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2215923361990026385;rport
From: 007 ;tag=974921936
To: 007
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1970 REGISTER
Contact:
Authorization: Digest username="007", realm="asterisk", nonce="07969da8", uri="sip:192.168.100.44:5060", response="95d97a35e526118592c4936d35823126", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0



--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2215923361990026385;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1970 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0



Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK385d8d46;rport
Max-Forwards: 70
From: "Unknown" ;tag=as7f140993
To:
Contact:
Call-ID: 011d5a6b785d3e2264052b681bf1ba47@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 29 Dec 2010 05:22:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2215923361990026385;received=192.168.100.68;rport=5060
From: 007 ;tag=974921936
To: 007 ;tag=as11634ef1
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1970 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: ;expires=60
Date: Wed, 29 Dec 2010 05:22:33 GMT
Content-Length: 0



Scheduling destruction of SIP dialog '67672746522226-229372301815489@192.168.100.68' in 32000 ms (Method: REGISTER)


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK385d8d46;rport
From: "Unknown" ;tag=as7f140993
To:
Call-ID: 011d5a6b785d3e2264052b681bf1ba47@192.168.100.44
CSeq: 102 OPTIONS
Contact:
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0



--- (11 headers 0 lines) ---
Really destroying SIP dialog '011d5a6b785d3e2264052b681bf1ba47@192.168.100.44' Method: OPTIONS


CANCEL sip:s@192.168.100.44:5060;maddr=188.19.10.72 SIP/2.0
From: ;tag=-45026-41d8732-4a2142f5-41d8732
To:
Call-ID: 6caf7203d227708c20e85e3c7b6aa7786353a3@62.148.237.152
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 62.148.237.152:5060;branch=z9hG4bK-23dbc6-8c127d8d-60dd155f
Max-Forwards: 70
Content-Length: 0



--- (8 headers 0 lines) ---


SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 62.148.237.152:5060;branch=z9hG4bK-23dbc6-8c127d8d-60dd155f;received=62.148.237.152
From: ;tag=-45026-41d8732-4a2142f5-41d8732
To: ;tag=as4c82ab73
Call-ID: 6caf7203d227708c20e85e3c7b6aa7786353a3@62.148.237.152
CSeq: 1 CANCEL
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0



REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:nugngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK672cbbab;rport
Max-Forwards: 70
From: ;tag=as00228927
To:
Call-ID: 32c26aad029c7b9b15e2cc1148899fae@127.0.0.1
CSeq: 109 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Authorization: Digest username="201901", realm="Realm", algorithm=MD5, uri="sip:nugngn.usi.ru", nonce="MTI5MzU5Mjg4MzQxN2Q2NWY4YTFhNWFhZDJjN2E2ZDUzMjY2MTk1OTg3YzE4", response="8a040867f9d2114fbbf519f0a3c8d507", qop=auth, cnonce="76666a32", nc=00000003
Expires: 120
Contact:
Content-Length: 0


---


SIP/2.0 100 Trying
From: ;tag=as00228927
To:
Call-ID: 32c26aad029c7b9b15e2cc1148899fae@127.0.0.1
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=55334;branch=z9hG4bK672cbbab
Content-Length: 0



--- (7 headers 0 lines) ---


SIP/2.0 200 Registration Successful
From: "201901 201901";tag=as00228927
To: ;tag=1010258005
Call-ID: 32c26aad029c7b9b15e2cc1148899fae@127.0.0.1
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=55334;branch=z9hG4bK672cbbab
contact: ;expires=111
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0



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Scheduling destruction of SIP dialog '32c26aad029c7b9b15e2cc1148899fae@127.0.0.1' in 32000 ms (Method: REGISTER)