провайдер не видит моего ack'а

приветствую всех! не бейте ногами, я здесь новенький

Столкнулся с проблемой, что провайдер не видит моего ACK'а
куда копать вроде бы вчера все еще работало :(

схема - астериск 1,8 + 4 voip шлюза Zyxel-2024 в одной локальной сети на одном интерфейсе, на другом локалка провайдера, где получаю по SIP'у peer-to-peer входящую связь

сейчас 50 человек сидят без городской связи, и мне икается :(

звонок с города идет на гор номер, выслушиваю приветствие, делаю донабор напр на 101 номер (факс на автомате), сеачала пару смазанных гудков, соединение состоялось, звука нет
в консоли

[Nov 8 14:20:06] NOTICE[3510]: chan_sip.c:19523 handle_response_peerpoke: Peer
'MT01' is now Lagged. (21041ms / 3000ms)
== Spawn extension (office, 141, 1) exited non-zero on 'SIP/136-000000c4'
[Nov 8 14:20:16] NOTICE[3510]: chan_sip.c:19523 handle_response_peerpoke: Peer
'MT01' is now Reachable. (20ms / 3000ms)

провайдер утверждает, что на мой инвайт он присылает OK 200 , а в ответ ничего нет.. хотя должен быть ACK

сейчас выложу кусок дебага

2010-11-08 14:23

neumax

Откуда: г.Щёлково МО

Сообщений: 3

Re: провайдер не видит моего ack'а

o=HuaweiSoftX3000 918067 918071 IN IP4 10.220.0.2
s=Sip Call
c=IN IP4 10.220.0.2
t=0 0
m=audio 38896 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
<------------->
--- (9 headers 10 lines) ---

<--- SIP read from UDP:10.220.0.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.220.9.5:5060;branch=z9hG4bK1783e083;rport=5060
Call-ID: 10f1e5be2eac3c716e0adb7d03f9dcb0@10.220.0.2
From: "4957480025"<sip:4957480025@10.220.0.2>;tag=as3e8c8b46
To: <sip:84855287992@10.220.0.2;user=phone>;tag=7565b2ba
CSeq: 105 INVITE
Contact: <sip:84855287992@10.220.0.2:5060;user=phone>
Content-Length: 222
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 918067 918071 IN IP4 10.220.0.2
s=Sip Call
c=IN IP4 10.220.0.2
t=0 0
m=audio 38896 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
<------------->
--- (9 headers 10 lines) ---

<--- SIP read from UDP:85.142.210.125:5060 --->
REGISTER sip:109.073.033.053 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.61:5060;branch=z9hG4bK364c9000a410565
From: <sip:565@109.073.033.053>;tag=364c9000a4
To: sip:565@109.073.033.053
Call-ID: 36bfa04c-3849-90b9-8000-0002a4fffff0@192.168.15.61
CSeq: 10565 REGISTER
Date: Thu, 30 Sep 2010 18:44:55 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="565", realm="asterisk", nonce="78b5710c", uri="s
ip:109.073.033.053", response="103ba8f37fbe0b720567361482952604", algorithm=MD5
Contact: <sip:565@192.168.15.61>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70

<------------->
--- (13 headers 0 lines) ---
Sending to 85.142.210.125:5060 (no NAT)

<--- Transmitting (NAT) to 85.142.210.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.61:5060;branch=z9hG4bK364c9000a410565;received=85.14
2.210.125;rport=5060
From: <sip:565@109.073.033.053>;tag=364c9000a4
To: sip:565@109.073.033.053
Call-ID: 36bfa04c-3849-90b9-8000-0002a4fffff0@192.168.15.61
CSeq: 10565 REGISTER
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Content-Length: 0

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

<--- Transmitting (NAT) to 85.142.210.125:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.15.61:5060;branch=z9hG4bK364c9000a410565;received=85.14
2.210.125;rport=5060
From: <sip:565@109.073.033.053>;tag=364c9000a4
To: sip:565@109.073.033.053;tag=as58a0c5b7
Call-ID: 36bfa04c-3849-90b9-8000-0002a4fffff0@192.168.15.61
CSeq: 10565 REGISTER
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="38d98925"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '36bfa04c-3849-90b9-8000-0002a4fffff0@192.1
68.15.61' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:85.142.210.125:5060 --->
REGISTER sip:109.073.033.053 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.61:5060;branch=z9hG4bK364c9000a410566
From: <sip:565@109.073.033.053>;tag=364c9000a4
To: sip:565@109.073.033.053
Call-ID: 36bfa04c-3849-90b9-8000-0002a4fffff0@192.168.15.61
CSeq: 10566 REGISTER
Date: Thu, 30 Sep 2010 18:44:55 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="565", realm="asterisk", nonce="38d98925", uri="s
ip:109.073.033.053", response="2cea3587c64c07c44a197e85cac885dc", algorithm=MD5
Contact: <sip:565@192.168.15.61>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70

<------------->
--- (13 headers 0 lines) ---
Sending to 85.142.210.125:5060 (NAT)

<--- Transmitting (NAT) to 85.142.210.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.61:5060;branch=z9hG4bK364c9000a410566;received=85.14
2.210.125;rport=5060
From: <sip:565@109.073.033.053>;tag=364c9000a4
To: sip:565@109.073.033.053
Call-ID: 36bfa04c-3849-90b9-8000-0002a4fffff0@192.168.15.61
CSeq: 10566 REGISTER
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Content-Length: 0

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

<--- Transmitting (NAT) to 85.142.210.125:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.61:5060;branch=z9hG4bK364c9000a410566;received=85.14
2.210.125;rport=5060
From: <sip:565@109.073.033.053>;tag=364c9000a4
To: sip:565@109.073.033.053;tag=as58a0c5b7
Call-ID: 36bfa04c-3849-90b9-8000-0002a4fffff0@192.168.15.61
CSeq: 10566 REGISTER
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Expires: 60
Contact: <sip:565@192.168.15.61>;expires=60
Date: Mon, 08 Nov 2010 10:27:53 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '36bfa04c-3849-90b9-8000-0002a4fffff0@192.1
68.15.61' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.220.0.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.220.9.5:5060;branch=z9hG4bK1783e083;rport=5060
Call-ID: 10f1e5be2eac3c716e0adb7d03f9dcb0@10.220.0.2
From: "4957480025"<sip:4957480025@10.220.0.2>;tag=as3e8c8b46
To: <sip:84855287992@10.220.0.2;user=phone>;tag=7565b2ba
CSeq: 105 INVITE
Contact: <sip:84855287992@10.220.0.2:5060;user=phone>
Content-Length: 222
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 918067 918071 IN IP4 10.220.0.2
s=Sip Call
c=IN IP4 10.220.0.2
t=0 0
m=audio 38896 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
<------------->
--- (9 headers 10 lines) ---

<--- SIP read from UDP:192.168.97.102:5060 --->
BYE sip:135@192.168.97.100:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.97.102:5060;branch=z9hG4bK9236aefe49750f15
Max-Forwards: 70
To: <sip:135@192.168.97.100>;tag=as3ac4c708
From: <sip:132@192.168.97.100;user=phone>;tag=4E0NykDM1gD
Call-ID: 6240D1B97277D130@192.168.97.102
CSeq: 3 BYE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Sending to 192.168.97.102:5060 (no NAT)
Scheduling destruction of SIP dialog '6240D1B97277D130@192.168.97.102' in 32000
ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.97.102:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.97.102:5060;branch=z9hG4bK9236aefe49750f15;received=192
.168.97.102
From: <sip:132@192.168.97.100;user=phone>;tag=4E0NykDM1gD
To: <sip:135@192.168.97.100>;tag=as3ac4c708
Call-ID: 6240D1B97277D130@192.168.97.102
CSeq: 3 BYE
Server: Asterisk PBX 1.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '459aa44a2e2ccdad3dea8a4b174735a9@192.168.9
7.100:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:135@192.168.97.102:5060> for address/port to send
to
set_destination: set destination to 192.168.97.102:5060
Reliably Transmitting (no NAT) to 192.168.97.102:5060:
BYE sip:135@192.168.97.102:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.97.100:5060;branch=z9hG4bK532e5790
Max-Forwards: 70
From: "132" <sip:132@192.168.97.100>;tag=as570fb92e
To: <sip:135@192.168.97.102:5060>;tag=yM0NzMDMGFD
Call-ID: 459aa44a2e2ccdad3dea8a4b174735a9@192.168.97.100:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

2010-11-08 15:40

neumax

Откуда: г.Щёлково МО

Сообщений: 3

Re: провайдер не видит моего ack'а

уф-ф-ф! разрешилась ситуация

в процессе усиленного "допроса" представителя поддержки, он произнес сокровенные слова "зачем-то тут ваш ре-инвайт.."

тут же на проводе добавил в sip.conf

canreinvite=no

и всё.. прошу прощения у всей честной компании за непредоставление sip и extensions настроек.

2010-11-08 16:52