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addpac 200B, проблема с регистрацией
http://old.xdsl.ru/svpro/viewtopic.php?f=4&t=647		 Страница 1 из 2
Автор:  ggR [ 21 авг 2009, 02:27 ]
Заголовок сообщения:  addpac 200B, проблема с регистрацией
Доброго времени суток, уважаемые. Подскажите, в чем проблема может быть ? Имеем addpac 200B, настроен на коннект к АТСке по SIP. Но, проблема в том, что не номера не регистрируются на АТС. Подскажите, куда копать ?

Easy Setup показывает следующее в логе

Server address Port Priority Status 192.168.56.213 5060 128 Failed

Proxyserver registration status :
UserName Regist Status
192 yes Failed
193 yes Failed


конфа

Using 1862 out of 65332 bytes
!
version 8.237
!
hostname AP200
!
dhcp-list 0 type server
dhcp-list 0 address server interface ether0.0
dhcp-list 0 option dhcp-lease-time 7200
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.254 255.255.255.0
dhcp-list 1 option dhcp-lease-time 600
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.56.221 255.255.255.0
!
interface ether1.0
no ip address
ip dhcp-group 0
!
snmp name AP200B
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.56.33
!
dnshost nameserver 192.168.56.34
!
auto-script autorun.inf
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
no busyout monitor callagent
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
! FXS
voice-port 0/1
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 192
port 0/0
user-password 192
!
dial-peer voice 1 pots
destination-pattern 193
port 0/1
user-password 193
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.56.221
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g729
!
!
!
! SIP UA configuration.
!
sip-ua
sip-server 192.168.56.213
register e164
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
!
!
! Tones
!
!
!
!
Автор:  Geniu$$ [ 21 авг 2009, 09:12 ]
Заголовок сообщения: 
Добавьте
conf t
sip
user

Прошейтесь

Как прошиться www.svpro.ru/addpac/faq3.htm
Автор:  ggR [ 22 авг 2009, 00:18 ]
Заголовок сообщения: 
Обновил, прописал команды. Заработало. Спасибо.
Автор:  ggR [ 22 авг 2009, 02:02 ]
Заголовок сообщения: 
Возник еще вопрос.
К этой АТС еще подключен абонент посредством IP телефонии. С обычного телефона до него дозваниваемся без проблем, звоню сейчас с телефона, который подключен к addpac 200b, слышу с той стороны ответ, меня не слышат. Кодеки стоят что там, что здесь g729. В чем может быть проблема ?

upd: вообщем я звоню с телефона подключеного к addpac меня никто не слышит, звонят мне - все нормально

Using 2067 out of 65332 bytes
!
version 8.30U
!
hostname AP200
!
!
no bridge spanning-tree
!
dhcp-list 0 type server
dhcp-list 0 address server interface ether0.0
dhcp-list 0 option dhcp-lease-time 7200
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.254 255.255.255.0
dhcp-list 1 option dhcp-lease-time 600
!
!
no ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.56.221 255.255.255.0
!
interface ether1.0
no ip address
ip dhcp-group 0
!
snmp name AP200B
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.56.33
!
dnshost nameserver 192.168.56.34
!
pnp-sktelink debug on
!
auto-script autorun.inf
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
busyout monitor sip-server
no busyout monitor callagent
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
! FXS
voice-port 0/1
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 192
port 0/0
user-password 192
!
dial-peer voice 1 pots
destination-pattern 193
port 0/1
user-password 193
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
no vad
codec-variant g7231 standard
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.56.221
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g729
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-server 192.168.56.213
register e164
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad
!
!
! Tones
!
!
!
!
Автор:  ggR [ 22 авг 2009, 03:55 ]
Заголовок сообщения: 
debug

AP200(config)# [1766.780] VM(0/0/0) vmOffHook
[1766.840] VM(0/0/0) vmTmoOffHook
[1766.840] VM(0/0/0) Rx OffHook
[1766.840] VM(0/0/0) Modem attribute disable
[1766.840] VM(0/0/0) Modem attribute G711A
[1766.840] VM(0/0/0) vopp enable
[1766.840] VM(0/0/0) Tx OFFHOOK_IND
[1766.840] VM(0/0/0) play Dial tone
268 <CEP 000000> : Call Received
269 <CEP 000000> : Call Initiated : calledNumber() crv(0) total(0)
270 <Call 28> : ****************** Call Created status(InitiatedByFXS
) *******************
271 <CEP 000000> : Calling number(192)
272 <CEP 000000> : Call id(0bdc8f4a-e0bd-d891-8048-0002a405a338) callNum(
28)
[1774.655] VM(0/0/0) Tx DIGIT_IND '1'
[1774.655] VM(0/0/0) play mute
273 <Call 28> : Digit(1) at InitiatedByFXS
274 <Call 28> : MatchedAll
[1774.895] VM(0/0/0) Tx DIGIT_IND '5'
275 <Call 28> : Digit(5) at CalleeDeterminedWaitDigit
276 <Call 28> : MatchedAll
[1775.175] VM(0/0/0) Tx DIGIT_IND '0'
277 <Call 28> : Digit(0) at CalleeDeterminedWaitDigit
278 <Call 28> : MatchedAll
279 <Time 28> : Inter digit timer timeout.
280 <Call 28> : Digit(#) at CalleeDeterminedWaitDigit
281 <Call 28> : MatchAllProcess After Sorted
<0> id(1000) dest(T) prefer(0) selected(18)
282 <Call 28> : Initiate callee with dial-peer(T) status(CalleeDetermi
nedAll) id(0bdc8f4a-e0bd-d891-8048-0002a405a338)
283 <NetEP 28> : InitiateOutCall: calledNum(150) callingNum(192) target
(sip-server)
[1778.175] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 0
[1778.175] VM(0/0/0) DTMF disable later
284 <NetEP 28> : DoCall: calledAddr(sip:150@192.168.56.213:5060) callin
gAddr(192)
[1778.175] VM(0/0/0) set T38 mode STD
[1778.175] VM(0/0/0) Fax rate 9600
285 <SIP 0> : No authentication information available
286 <SIP 28> : Send INVITE Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
INVITE sip:150@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a443
From: <sip:192@192.168.56.213>;tag=174a8a49a4
To: <sip:150@192.168.56.213>
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 43 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Sat, 22 Aug 2009 11:52:55 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:192@192.168.56.221>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 249
Max-Forwards: 70

v=0
o=192 1250941975 1250941975 IN IP4 192.168.56.221
s=AddPac Gateway SDP
c=IN IP4 192.168.56.221
t=1250941975 0
m=audio 23040 RTP/AVP 18 8 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=ptime:20

[1778.225] RTA(0/0/0) Rx RS_LISTEN_REQ callId=28 ssId=1 G711U
peer=0.0.0.0 mp=23040/23041 hp=0/0
[1778.225] VM(0/0/0) vopp idle
[1778.225] VM(0/0/0) start codec replace timer to G711U
[1778.225] VM(0/0/0) discard voice under codec replace
[1778.235] VM(0/0/0) discard voice under codec replace

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a443
To: sip:150@192.168.56.213
From: sip:192@192.168.56.213;tag=174a8a49a4
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 43 INVITE
Content-Length: 0

287 <SIP 28> : Receive 100 Trying
288 <SIP 28> : Transaction (43 INVITE) proceeding

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a443
To: sip:150@192.168.56.213;tag=8576
From: sip:192@192.168.56.213;tag=174a8a49a4
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 43 INVITE
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Proxy-Authenticate: Digest realm="Registered Users",nonce="be7cf8f1e3c78e1c3871e
2c58b162c58",algorithm=MD5
Content-Length: 0

289 <SIP 28> : Receive 407 Proxy Authentication Required
290 <SIP 28> : Transaction (43 INVITE) completed
[1778.285] VM(0/0/0) Modem attribute disable
[1778.285] VM(0/0/0) Modem attribute G711A
[1778.285] VM(0/0/0) vopp enable
[1778.285] VM(0/0/0) codec replaced to G711U
291 <SIP 28> : Send ACK Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
ACK sip:150@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a443
From: <sip:192@192.168.56.213>;tag=174a8a49a4
To: sip:150@192.168.56.213;tag=8576
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 43 ACK
Content-Length: 0
Max-Forwards: 70


292 <SIP 0> : No opaque in authentication
293 <SIP 0> : Adding authentication information
294 <SIP 28> : Send INVITE Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
INVITE sip:150@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a444
From: <sip:192@192.168.56.213>;tag=174a8a49a4
To: <sip:150@192.168.56.213>
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 44 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Sat, 22 Aug 2009 11:52:55 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:192@192.168.56.221>
Accept: application/sdp
Proxy-Authorization: Digest username="192", realm="Registered Users", nonce="be7
cf8f1e3c78e1c3871e2c58b162c58", uri="sip:150@192.168.56.213", response="e1d6282e
e65758d8916dcdd082d34db0", algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 253
Max-Forwards: 70

v=0
o=192 1250941975 1250941975 IN IP4 192.168.56.221
s=AddPac Gateway SDP
c=IN IP4 192.168.56.221
t=1250941975 0
m=audio 23040 RTP/AVP 18 8 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=ptime:20


Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a444
To: sip:150@192.168.56.213
From: sip:192@192.168.56.213;tag=174a8a49a4
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 44 INVITE
Content-Length: 0

295 <SIP 28> : Receive 100 Trying
296 <SIP 28> : Transaction (44 INVITE) proceeding

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a444
To: sip:150@192.168.56.213;tag=31560
From: sip:192@192.168.56.213;tag=174a8a49a4
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 44 INVITE
Contact: sip:192.168.56.213:5060
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Length: 0

297 <SIP 28> : Receive 180 Ringing
298 <SIP 28> : Transaction (44 INVITE) proceeding
299 <Call 28> : Alert from(fffffffe) pseudo(0) inband(0) status(Callee
Initiated)
[1778.390] RTA(0/0/0) Rx CC_ALERT_RSP peerId(0/0/0)
[1778.390] VM(0/0/0) play RingBack tone

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a444
To: sip:150@192.168.56.213;tag=31560
From: sip:192@192.168.56.213;tag=174a8a49a4
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 44 INVITE
Contact: sip:192.168.56.213:5060
Require: timer
Supported: timer
Session-Expires: 1800;refresher=uas
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Type: application/sdp
Content-Length: 149

v=0
o=- 1 1 IN IP4 192.168.56.214
s=-
c=IN IP4 192.168.56.214
t=0 0
m=audio 12056 RTP/AVP 18
a=rtpmap:18 G729/8000/1
a=sendrecv
a=ptime:20
Unexpected input, line 6, column 26
[1787.690] VM(0/0/0) vopp idle
[1787.690] VM(0/0/0) start codec replace timer to G729A
[1787.690] VM(0/0/0) Rx RTP replace codec to G729A
300 <SIP 28> : Receive 200 OK
[1787.690] VM(0/0/0) discard voice under codec replace
301 <Call 28> : Connected from(fffffffe)
[1787.700] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[1787.700] VM(0/0/0) VAD disable
[1787.700] VM(0/0/0) SID enable by CCC
[1787.700] RTA(0/0/0) Rx CC_CONNECT_RSP peerId(0/0/0)
[1787.700] VM(0/0/0) Fax enable
[1787.700] VM(0/0/0) discard voice under codec replace
302 <NetEP 28> : Call with sip:150@192.168.56.213 established
303 <SIP 28> : Received INVITE OK response
304 <SIP 28> : Send ACK Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
ACK sip:192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a444
From: <sip:192@192.168.56.213>;tag=174a8a49a4
To: sip:150@192.168.56.213;tag=31560
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 44 ACK
Content-Length: 0
Max-Forwards: 70


305 <SIP 28> : Check Event Relation
306 <SIP 28> : Set Terminated Success for 44 INVITE
[1787.750] VM(0/0/0) Modem attribute disable
[1787.750] VM(0/0/0) Modem attribute G711A
[1787.750] VM(0/0/0) vopp enable
[1787.750] VM(0/0/0) codec replaced to G729A
[1787.750] VM(0/0/0) Fax enable
[1787.750] VM(0/0/0) play mute
[1787.765] VM(0/0/0) DTMF disable
[1805.145] VM(0/0/0) vmOnHook
[1805.195] VM(0/0/0) vmTmoOnHook
[1805.245] VM(0/0/0) vmTmoOnHook
[1805.295] VM(0/0/0) vmTmoOnHook
[1805.345] VM(0/0/0) vmTmoOnHook
[1805.395] VM(0/0/0) vmTmoOnHook
[1805.445] VM(0/0/0) vmTmoOnHook
[1805.495] VM(0/0/0) vmTmoOnHook
[1805.545] VM(0/0/0) vmTmoOnHook
[1805.595] VM(0/0/0) vmTmoOnHook
[1805.645] VM(0/0/0) vmTmoOnHook
[1805.695] VM(0/0/0) vmTmoOnHook
[1805.745] VM(0/0/0) vmTmoOnHook
[1805.795] VM(0/0/0) vmTmoOnHook
[1805.845] VM(0/0/0) vmTmoOnHook
[1805.845] VM(0/0/0) Rx OnHook
[1805.845] VM(0/0/0) vopp idle
[1805.845] VM(0/0/0) Tx DISCONN_CNF
307 <CEP 000000> : Disconnected(16) at Busy
308 <Call 28> : Terminated from(0) this(Local:CallClear) before(NULL)
forced(0)
309 <CEP 000000> : DisconnectCall at Idle
310 <SIP 28> : ReleaseWithBYE
311 <SIP 28> : Send BYE Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
BYE sip:192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a445
From: <sip:192@192.168.56.213>;tag=174a8a49a4
To: sip:150@192.168.56.213;tag=31560
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 45 BYE
Date: Sat, 22 Aug 2009 11:53:22 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:192@192.168.56.221>
Content-Length: 0
Max-Forwards: 70


[1805.865] RTA(0/0/0) Rx RS_CLOSE_REQ callId=28 ssId=1 dir=reve
[1805.865] RTA(0/0/0) close Media socket
[1805.865] RTA(0/0/0) close RTCP socket
312 <NetEP 28> : Call TO <sip:150@192.168.56.213> terminated reason(Loc
al:CallClear)

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK174a8a49a445
To: sip:150@192.168.56.213;tag=31560
From: sip:192@192.168.56.213;tag=174a8a49a4
Call-ID: 17dc8f4a-3613-8a12-8049-0002a405a338@192.168.56.221
CSeq: 45 BYE
Content-Length: 0

313 <SIP 28> : Receive 200 OK
314 <SIP 28> : Transaction (45 BYE) completed
315 <SIP 28> : Set Terminated Success for 43 INVITE
316 <SIP 28> : Set Terminated Success for 45 BYE
Автор:  Geniu$$ [ 24 авг 2009, 07:13 ]
Заголовок сообщения: 
Попробуйте кодек другой.

А вообще не нравится мне, то что на адпак приходит. Например вот это:
To: sip:150@192.168.56.213;tag=31560
From: sip:192@192.168.56.213;tag=174a8a49a4

Вот выдержка из RFC:
The Contact, From, and To header fields contain a URI. If the URI
contains a comma, question mark or semicolon, the URI MUST be
enclosed in angle brackets (< and >).

И адпаку тоже что-то не нравиться:
Unexpected input, line 6, column 26

Что у вас за станция такая?
Автор:  ggR [ 25 авг 2009, 08:41 ]
Заголовок сообщения: 
Кодеки принудительно ставил все, и гонял все.
АТСка Panasonic TDE200
Что можно еще попробовать?
Автор:  ggR [ 26 авг 2009, 05:40 ]
Заголовок сообщения: 
вообще ничего не понимаю, теперь постоянно идет сразу сброс. На АТСке аддпак регистрируется, все нормально

Using 1988 out of 65332 bytes
!
version 8.30U
!
hostname AP200
!
!
no bridge spanning-tree
!
dhcp-list 0 type server
dhcp-list 0 address server interface ether0.0
dhcp-list 0 option dhcp-lease-time 7200
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.254 255.255.255.0
dhcp-list 1 option dhcp-lease-time 600
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.56.221 255.255.255.0
!
interface ether1.0
no ip address
ip dhcp-group 0
!
snmp name AP200B
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.56.33
!
pnp-sktelink debug on
!
auto-script autorun.inf
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
busyout monitor sip-server
no busyout monitor callagent
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
! FXS
voice-port 0/1
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 205
port 0/0
user-password 205
!
dial-peer voice 1 pots
destination-pattern 206
port 0/1
user-password 206
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.56.221
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g729
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-server 192.168.56.213
register e164
!
!
! MGCP configuration.
!
mgcp
no codec
vad
!
!
! Tones
!
!
!
!
Автор:  Geniu$$ [ 26 авг 2009, 07:38 ]
Заголовок сообщения: 
Сделайте ещё:
conf t
no ip-s

Покажите ещё дебаг
deb rta ipc
deb voip sip
deb voip call

Я думаю тут косяк в протоколе у панасоника о котором я раньше написал. Спросите панасоник.
Автор:  ggR [ 26 авг 2009, 08:03 ]
Заголовок сообщения: 
Походу все не влезло
Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
INVITE sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a412
From: <sip:205@192.168.56.213>;tag=164a8c12a4
To: <sip:171@192.168.56.213>
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 12 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Wed, 26 Aug 2009 16:00:22 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:205@192.168.56.221>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 349
Max-Forwards: 70

v=0
o=205 1251302422 1251302422 IN IP4 192.168.56.221
s=AddPac Gateway SDP
c=IN IP4 192.168.56.221
t=1251302422 0
m=audio 23014 RTP/AVP 4 18 0 8 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30

[8522.010] RTA(0/0/0) Rx RS_LISTEN_REQ callId=7 ssId=1 G711U
peer=0.0.0.0 mp=23014/23015 hp=0/0
[8522.010] VM(0/0/0) vopp idle
[8522.010] VM(0/0/0) start codec replace timer to G711U
[8522.010] VM(0/0/0) discard voice under codec replace
[8522.020] VM(0/0/0) discard voice under codec replace

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a412
To: sip:171@192.168.56.213
From: sip:205@192.168.56.213;tag=164a8c12a4
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 12 INVITE
Content-Length: 0

142 <SIP 7> : Receive 100 Trying
143 <SIP 7> : Transaction (12 INVITE) proceeding

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a412
To: sip:171@192.168.56.213;tag=15790
From: sip:205@192.168.56.213;tag=164a8c12a4
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 12 INVITE
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Proxy-Authenticate: Digest realm="Registered Users",nonce="b871e2c48912254b972f5
fbf7ffefdfb",algorithm=MD5
Content-Length: 0

144 <SIP 7> : Receive 407 Proxy Authentication Required
145 <SIP 7> : Transaction (12 INVITE) completed
[8522.070] VM(0/0/0) Modem attribute disable
[8522.070] VM(0/0/0) Modem attribute G711A
[8522.070] VM(0/0/0) vopp enable
[8522.070] VM(0/0/0) codec replaced to G711U
146 <SIP 7> : Send ACK Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
ACK sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a412
From: <sip:205@192.168.56.213>;tag=164a8c12a4
To: sip:171@192.168.56.213;tag=15790
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 12 ACK
Content-Length: 0
Max-Forwards: 70


147 <SIP 0> : No opaque in authentication
148 <SIP 0> : Adding authentication information
149 <SIP 7> : Send INVITE Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
INVITE sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a413
From: <sip:205@192.168.56.213>;tag=164a8c12a4
To: <sip:171@192.168.56.213>
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 13 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Wed, 26 Aug 2009 16:00:22 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:205@192.168.56.221>
Accept: application/sdp
Proxy-Authorization: Digest username="205", realm="Registered Users", nonce="b87
1e2c48912254b972f5fbf7ffefdfb", uri="sip:171@192.168.56.213", response="c1eabba3
c83c7beaf386699456875f40", algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 357
Max-Forwards: 70

v=0
o=205 1251302422 1251302422 IN IP4 192.168.56.221
s=AddPac Gateway SDP
c=IN IP4 192.168.56.221
t=1251302422 0
m=audio 23014 RTP/AVP 4 18 0 8 101
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:30


Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a413
To: sip:171@192.168.56.213
From: sip:205@192.168.56.213;tag=164a8c12a4
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 13 INVITE
Content-Length: 0

150 <SIP 7> : Receive 100 Trying
151 <SIP 7> : Transaction (13 INVITE) proceeding

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a413
To: sip:171@192.168.56.213;tag=4796
From: sip:205@192.168.56.213;tag=164a8c12a4
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 13 INVITE
Contact: sip:192.168.56.213:5060
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Length: 0

152 <SIP 7> : Receive 180 Ringing
153 <SIP 7> : Transaction (13 INVITE) proceeding
154 <Call 7> : Alert from(fffffffe) pseudo(0) inband(0) status(Callee
Initiated)
[8522.185] RTA(0/0/0) Rx CC_ALERT_RSP peerId(0/0/0)
[8522.185] VM(0/0/0) play RingBack tone
[8532.620] VM(0/0/0) vmOnHook
[8532.670] VM(0/0/0) vmTmoOnHook
[8532.720] VM(0/0/0) vmTmoOnHook
[8532.770] VM(0/0/0) vmTmoOnHook
[8532.820] VM(0/0/0) vmTmoOnHook
[8532.870] VM(0/0/0) vmTmoOnHook
[8532.920] VM(0/0/0) vmTmoOnHook
[8532.970] VM(0/0/0) vmTmoOnHook
[8533.020] VM(0/0/0) vmTmoOnHook
[8533.070] VM(0/0/0) vmTmoOnHook
[8533.120] VM(0/0/0) vmTmoOnHook
[8533.170] VM(0/0/0) vmTmoOnHook
[8533.220] VM(0/0/0) vmTmoOnHook
[8533.270] VM(0/0/0) vmTmoOnHook
[8533.320] VM(0/0/0) vmTmoOnHook
[8533.320] VM(0/0/0) Rx OnHook
[8533.320] VM(0/0/0) vopp idle
[8533.320] VM(0/0/0) Tx DISCONN_CNF
155 <CEP 000000> : Disconnected(16) at Busy
156 <Call 7> : Terminated from(0) this(Local:CallClear) before(NULL)
forced(0)
157 <CEP 000000> : DisconnectCall at Idle
158 <SIP 7> : ReleaseWithCANCEL for 1 INVITEs)
159 <SIP 7> : Send CANCEL Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
CANCEL sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a413
From: <sip:205@192.168.56.213>;tag=164a8c12a4
To: <sip:171@192.168.56.213>
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 13 CANCEL
Date: Wed, 26 Aug 2009 16:00:33 GMT
User-Agent: AddPac SIP Gateway
Content-Length: 0
Max-Forwards: 70


[8533.340] RTA(0/0/0) Rx RS_CLOSE_REQ callId=7 ssId=1 dir=reve
[8533.340] RTA(0/0/0) close Media socket
[8533.340] RTA(0/0/0) close RTCP socket
160 <NetEP 7> : Call TO <sip:171@192.168.56.213> terminated reason(Loc
al:CallClear)

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a413
To: sip:171@192.168.56.213;tag=4796
From: sip:205@192.168.56.213;tag=164a8c12a4
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 13 CANCEL
Content-Length: 0

161 <SIP 7> : Receive 200 OK
162 <SIP 7> : Transaction (13 CANCEL) completed

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a413
To: sip:171@192.168.56.213;tag=4796
From: sip:205@192.168.56.213;tag=164a8c12a4
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 13 INVITE
Content-Length: 0

163 <SIP 7> : Receive 487 Request Terminated
164 <SIP 7> : Transaction (13 INVITE) completed
165 <SIP 7> : Send ACK Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
ACK sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK164a8c12a413
From: <sip:205@192.168.56.213>;tag=164a8c12a4
To: sip:171@192.168.56.213;tag=4796
Call-ID: 165c954a-0ea0-8cef-8012-0002a405a338@192.168.56.221
CSeq: 13 ACK
Content-Length: 0
Max-Forwards: 70


166 <SIP 7> : Set Terminated Success for 13 CANCEL
167 <SIP 7> : Set Terminated Success for 12 INVITE
168 <SIP 7> : Set Terminated Success for 13 INVITE

AP200# ex

The System is ready. Please login to system.
login: root
password:
AP200B - Login : root at Console on Wed Aug 26 16:01:34 2009
AP200# conf t
Enter configuration commands, one per line. End with CNTL/Z
AP200(config)# [8601.580] VM(0/0/0) vmOffHook
[8601.640] VM(0/0/0) vmTmoOffHook
[8601.640] VM(0/0/0) Rx OffHook
[8601.640] VM(0/0/0) Modem attribute disable
[8601.640] VM(0/0/0) Modem attribute G711A
[8601.640] VM(0/0/0) vopp enable
[8601.640] VM(0/0/0) Tx OFFHOOK_IND
[8601.640] VM(0/0/0) play Dial tone
169 <CEP 000000> : Call Received
170 <CEP 000000> : Call Initiated : calledNumber() crv(0) total(0)
171 <Call 8> : ****************** Call Created status(InitiatedByFXS
) *******************
172 <CEP 000000> : Calling number(205)
173 <CEP 000000> : Call id(655c954a-29d6-f55f-8014-0002a405a338) callNum(
8)
[8604.350] VM(0/0/0) Tx DIGIT_IND '1'
[8604.350] VM(0/0/0) play mute
174 <Call 8> : Digit(1) at InitiatedByFXS
175 <Call 8> : MatchedAll
[8604.660] VM(0/0/0) Tx DIGIT_IND '7'
176 <Call 8> : Digit(7) at CalleeDeterminedWaitDigit
177 <Call 8> : MatchedAll
[8604.930] VM(0/0/0) Tx DIGIT_IND '1'
178 <Call 8> : Digit(1) at CalleeDeterminedWaitDigit
179 <Call 8> : MatchedAll
180 <Time 8> : Inter digit timer timeout.
181 <Call 8> : Digit(#) at CalleeDeterminedWaitDigit
182 <Call 8> : MatchAllProcess After Sorted
<0> id(1000) dest(T) prefer(0) selected(5)
183 <Call 8> : Initiate callee with dial-peer(T) status(CalleeDetermi
nedAll) id(655c954a-29d6-f55f-8014-0002a405a338)
184 <NetEP 8> : InitiateOutCall: calledNum(171) callingNum(205) target
(sip-server)
185 <NetEP 8> : DoCall: calledAddr(sip:171@192.168.56.213:5060) callin
gAddr(205)
[8607.930] VM(0/0/0) set T38 mode STD
[8607.930] VM(0/0/0) Fax rate 9600
186 <SIP 0> : No authentication information available
187 <SIP 8> : Send INVITE Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
INVITE sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a414
From: <sip:205@192.168.56.213>;tag=6c4ab015a4
To: <sip:171@192.168.56.213>
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 14 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Wed, 26 Aug 2009 16:01:48 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:205@192.168.56.221>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 349
Max-Forwards: 70

v=0
o=205 1251302508 1251302508 IN IP4 192.168.56.221
s=AddPac Gateway SDP
c=IN IP4 192.168.56.221
t=1251302508 0
m=audio 23016 RTP/AVP 4 18 0 8 101
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30

[8607.990] RTA(0/0/0) Rx RS_LISTEN_REQ callId=8 ssId=1 G711U
peer=0.0.0.0 mp=23016/23017 hp=0/0
[8607.990] VM(0/0/0) vopp idle
[8607.990] VM(0/0/0) start codec replace timer to G711U
[8607.995] VM(0/0/0) discard voice under codec replace

Received SIP PDU from ( 192.168.56.213:5060 )
[8608.005] VM(0/0/0) discard voice under codec replace
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a414
To: sip:171@192.168.56.213
From: sip:205@192.168.56.213;tag=6c4ab015a4
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 14 INVITE
Content-Length: 0

188 <SIP 8> : Receive 100 Trying
189 <SIP 8> : Transaction (14 INVITE) proceeding

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a414
To: sip:171@192.168.56.213;tag=10985
From: sip:205@192.168.56.213;tag=6c4ab015a4
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 14 INVITE
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Proxy-Authenticate: Digest realm="Registered Users",nonce="f7efdfbf7ffefcf9f2e4c
99224489021",algorithm=MD5
Content-Length: 0

190 <SIP 8> : Receive 407 Proxy Authentication Required
191 <SIP 8> : Transaction (14 INVITE) completed
[8608.050] VM(0/0/0) Modem attribute disable
[8608.050] VM(0/0/0) Modem attribute G711A
[8608.050] VM(0/0/0) vopp enable
[8608.050] VM(0/0/0) codec replaced to G711U
192 <SIP 8> : Send ACK Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
ACK sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a414
From: <sip:205@192.168.56.213>;tag=6c4ab015a4
To: sip:171@192.168.56.213;tag=10985
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 14 ACK
Content-Length: 0
Max-Forwards: 70


193 <SIP 0> : No opaque in authentication
194 <SIP 0> : Adding authentication information
195 <SIP 8> : Send INVITE Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
INVITE sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a415
From: <sip:205@192.168.56.213>;tag=6c4ab015a4
To: <sip:171@192.168.56.213>
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 15 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Wed, 26 Aug 2009 16:01:48 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:205@192.168.56.221>
Accept: application/sdp
Proxy-Authorization: Digest username="205", realm="Registered Users", nonce="f7e
fdfbf7ffefcf9f2e4c99224489021", uri="sip:171@192.168.56.213", response="94b5dda6
b9a4a65f108fefb45ecaff49", algorithm=MD5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 357
Max-Forwards: 70

v=0
o=205 1251302508 1251302508 IN IP4 192.168.56.221
s=AddPac Gateway SDP
c=IN IP4 192.168.56.221
t=1251302508 0
m=audio 23016 RTP/AVP 4 18 0 8 101
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=ptime:30


Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a415
To: sip:171@192.168.56.213
From: sip:205@192.168.56.213;tag=6c4ab015a4
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 15 INVITE
Content-Length: 0

196 <SIP 8> : Receive 100 Trying
197 <SIP 8> : Transaction (15 INVITE) proceeding

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a415
To: sip:171@192.168.56.213;tag=7973
From: sip:205@192.168.56.213;tag=6c4ab015a4
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 15 INVITE
Contact: sip:192.168.56.213:5060
Allow: INVITE,ACK,CANCEL,BYE,REGISTER
Content-Length: 0

198 <SIP 8> : Receive 180 Ringing
199 <SIP 8> : Transaction (15 INVITE) proceeding
200 <Call 8> : Alert from(fffffffe) pseudo(0) inband(0) status(Callee
Initiated)
[8608.165] RTA(0/0/0) Rx CC_ALERT_RSP peerId(0/0/0)
[8608.165] VM(0/0/0) play RingBack tone
[8612.775] VM(0/0/0) vmOnHook
[8612.825] VM(0/0/0) vmTmoOnHook
[8612.875] VM(0/0/0) vmTmoOnHook
[8612.925] VM(0/0/0) vmTmoOnHook
[8612.975] VM(0/0/0) vmTmoOnHook
[8613.025] VM(0/0/0) vmTmoOnHook
[8613.075] VM(0/0/0) vmTmoOnHook
[8613.125] VM(0/0/0) vmTmoOnHook
[8613.175] VM(0/0/0) vmTmoOnHook
[8613.225] VM(0/0/0) vmTmoOnHook
[8613.275] VM(0/0/0) vmTmoOnHook
[8613.325] VM(0/0/0) vmTmoOnHook
[8613.375] VM(0/0/0) vmTmoOnHook
[8613.425] VM(0/0/0) vmTmoOnHook
[8613.475] VM(0/0/0) vmTmoOnHook
[8613.475] VM(0/0/0) Rx OnHook
[8613.475] VM(0/0/0) vopp idle
[8613.475] VM(0/0/0) Tx DISCONN_CNF
201 <CEP 000000> : Disconnected(16) at Busy
202 <Call 8> : Terminated from(0) this(Local:CallClear) before(NULL)
forced(0)
203 <CEP 000000> : DisconnectCall at Idle
204 <SIP 8> : ReleaseWithCANCEL for 1 INVITEs)
205 <SIP 8> : Send CANCEL Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
CANCEL sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a415
From: <sip:205@192.168.56.213>;tag=6c4ab015a4
To: <sip:171@192.168.56.213>
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 15 CANCEL
Date: Wed, 26 Aug 2009 16:01:53 GMT
User-Agent: AddPac SIP Gateway
Content-Length: 0
Max-Forwards: 70


[8613.495] RTA(0/0/0) Rx RS_CLOSE_REQ callId=8 ssId=1 dir=reve
[8613.495] RTA(0/0/0) close Media socket
[8613.495] RTA(0/0/0) close RTCP socket
206 <NetEP 8> : Call TO <sip:171@192.168.56.213> terminated reason(Loc
al:CallClear)

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a415
To: sip:171@192.168.56.213;tag=7973
From: sip:205@192.168.56.213;tag=6c4ab015a4
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 15 CANCEL
Content-Length: 0

207 <SIP 8> : Receive 200 OK
208 <SIP 8> : Transaction (15 CANCEL) completed

Received SIP PDU from ( 192.168.56.213:5060 )
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a415
To: sip:171@192.168.56.213;tag=7973
From: sip:205@192.168.56.213;tag=6c4ab015a4
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 15 INVITE
Content-Length: 0

209 <SIP 8> : Receive 487 Request Terminated
210 <SIP 8> : Transaction (15 INVITE) completed
211 <SIP 8> : Send ACK Request

Sending SIP PDU to ( 192.168.56.213:5060 ) from 5060
ACK sip:171@192.168.56.213 SIP/2.0
Via: SIP/2.0/UDP 192.168.56.221:5060;branch=z9hG4bK6c4ab015a415
From: <sip:205@192.168.56.213>;tag=6c4ab015a4
To: sip:171@192.168.56.213;tag=7973
Call-ID: 6c5c954a-e6fa-b00a-8015-0002a405a338@192.168.56.221
CSeq: 15 ACK
Content-Length: 0
Max-Forwards: 70


212 <SIP 8> : Set Terminated Success for 15 CANCEL
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