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AP 1100F проблема с исходящей связью
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Автор:  viktorkc [ 01 авг 2012, 11:55 ]
Заголовок сообщения:  AP 1100F проблема с исходящей связью
Добрый день!
была ли у кого проблема с новой прошивкой ap1100f_g2_v8_41_086.bin?
у меня данная ситуация: после прошивки все sip-акаунты зарегистрировались, но на исходящие звонки работает только 0/0 порт остальные только на входящие...в чем может быть дело?



!
! APOS(tm) configuration saved from vty
! 2012/ 7/30 19:16:19 !
version 8.41.086
!
hostname voip.r-prof.pro
!
username root password administrator
username guest password guest user
!
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address dhcp
ip dhcp unicast
speed auto
!
interface FastEthernet0/1
ip address 192.168.10.1 255.255.255.0
half-duplex
!
! ip route 0.0.0.0 0.0.0.0 192.168.0.1 via dhcp
!
!
!
http server
!
!
! dns name-server 192.168.0.1 via dhcp
!
!
!
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
protocol sip
dtmf-relay rfc-2833
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
timing fxo-reconnecting-duration 500
qos-threshold delay 150
qos-threshold jitter 50
qos-threshold packet-loss 1
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
! FXS
voice-port 0/1
caller-id enable
!
!
! FXS
voice-port 0/2
caller-id enable
!
!
! FXS
voice-port 0/3
caller-id enable
!
!
! FXS
voice-port 1/0
caller-id enable
!
!
! FXS
voice-port 1/1
caller-id enable
!
!
! FXS
voice-port 1/2
caller-id enable
!
!
! FXS
voice-port 1/3
caller-id enable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 1000 pots
destination-pattern 100
port 0/0
user-name 2049-100
user-password xxxxx
!
dial-peer voice 1001 pots
destination-pattern 101
port 0/1
user-name 2049-101
user-password xxxxx
!
dial-peer voice 1002 pots
destination-pattern 102
port 0/2
user-name 2049-102
user-password xxxxx
!
dial-peer voice 1003 pots
destination-pattern 103
port 0/3
no register e164
user-name 2049-103
user-password xxxxx
!
dial-peer voice 1004 pots
destination-pattern 104
port 1/0
no register e164
user-name 2049-104
user-password xxxxx
!
dial-peer voice 1005 pots
destination-pattern 105
port 1/1
no register e164
user-name 2049-105
user-password xxxxx
!
!
!
! Voip peer configuration.
!
dial-peer voice 1006 voip
destination-pattern T
session target sip-server
session protocol sip
no vad
dtmf-relay info
!
!
!
dial-peer call-hold h
dial-peer call-transfer h
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.0.10
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-server xxx.xxx.xxx.xxx 5060 126
register e164
!
!
! Tones
!
!
!
!
line console
!
line vty
!
Автор:  Denis [ 02 авг 2012, 13:22 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
Пришлите результаты команды sho sip.
Автор:  viktorkc [ 07 авг 2012, 11:00 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
Proxyserver Registration Information
proxyserver registration option = e164
Proxyserver list :
--------------------------------------------------------------------------
---------------
Server address Port Priority Domain Status(LastFailReas
on)
--------------------------------------------------------------------------
---------------
xxx.xxx.xxx.xxx 5060 126 any Registered(E.164)(A
uth:alreadyAuth)

Proxyserver registration status :
--------------------------------------------------------------------------
---------
E.164 UserName Password Port Stat
us
--------------------------------------------------------------------------
---------
100 2049-100 <encrypt> 0/ 0 Regi
stered
101 2049-101 <encrypt> 0/ 1 Regi
stered
102 2049-102 <encrypt> 0/ 2 Regi
stered

SIP UA Timer counters
retry counter = 10
SIP UA Timer values
tretry (sip retry timer) = 500 msec.
tinterval (sip retry max interval timer) = 4 sec.
treg (sip register timer) = 60 sec.
tregtry (sip register retry timer) = 20 sec.
texpires (sip invite expire timer) = 180 sec.
tsipping (sip ping timer) = 45 sec.
tsrv (sip srv retry timer) = 60 sec.
thppolling (sip higher priority polling timer) = 30 sec.
SIP UA Session Timer value
Min-SE = 1800 sec.
Session-Expires = 1800 sec.
SIP DNS SRV Query : Disable
SIP Called-Party-Number : from URL
SIP FIR Support : Disable
SIP Call Transfer Mode : Basic
SIP Media Channel Start Mode : Default
SIP Reliable Provisional Response Option : Supported with value <100rel>
SIP Response Option : default
SIP Local Domain : NULL
SIP Special Char : NULL
SIP Routing Method of Incoming Call : Default
SIP Remote-Party-ID : Disabled
SIP Local Host Name : No
SIP Conference Server Info
Name (ID) = NULL
Related Voip Tag = -1
SIP NAT Info
PING = Disabled
Required = NULL
SIP Session Refresh Method = INVITE
SIP Keep Authentication information on registration = Yes
SIP Fixed CLID = NULL (add-line-number=no)
SIP Message Parameter Translation TRUE
SIP Higher Priority Polling = FALSE
SIP Switch Over Method = no-call
SIP Force-Forwarding Info
SIP Hook-Flash Event(INFO) Ignore = FALSE
SIP Q850 Reason support = FALSE
Автор:  genal [ 18 авг 2012, 18:53 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
Выложите пожалуйста еще дебаг во время звонка к примеру со второго порта.

conf term
term monit
ex
deb voip call
deb rta ipc
deb voip sip

Зделайте тестовый звонок (что вывалиться выложите)
(напишите какой номер набираете).

Остановить дебаги: no deb all.
Автор:  viktorkc [ 19 авг 2012, 08:24 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
набирал номер 89096890156

[71341.155] VM(0/1/0) play Dial tone
47 <CEP 000100> : Call Received
48 <CEP 000100> : Call Initiated : calledNumber() crv(0) total(0)
49 <Call 14> : ****** Call Created status(InitiatedByFXS) ver(8.28:2
006-02-06-00-00) time(1345396472) ****
50 <CEP 000100> : Calling number(102)
51 <CEP 000100> : Call id(f81e3150-3861-51aa-8024-0002a4072aaa) callNum(
14)
[71345.080] VM(0/1/0) Tx DIGIT_IND '8'
[71345.085] VM(0/1/0) play mute
52 <Call 14> : Digit(8) at InitiatedByFXS
53 <Call 14> : MatchedAll
[71345.630] VM(0/1/0) Tx DIGIT_IND '9'
54 <Call 14> : Digit(9) at CalleeDeterminedWaitDigit
55 <Call 14> : MatchedAll
[71345.970] VM(0/1/0) Tx DIGIT_IND '0'
56 <Call 14> : Digit(0) at CalleeDeterminedWaitDigit
57 <Call 14> : MatchedAll
[71346.290] VM(0/1/0) Tx DIGIT_IND '9'
58 <Call 14> : Digit(9) at CalleeDeterminedWaitDigit
59 <Call 14> : MatchedAll
[71346.720] VM(0/1/0) Tx DIGIT_IND '6'
60 <Call 14> : Digit(6) at CalleeDeterminedWaitDigit
61 <Call 14> : MatchedAll
[71347.080] VM(0/1/0) Tx DIGIT_IND '8'
62 <Call 14> : Digit(8) at CalleeDeterminedWaitDigit
63 <Call 14> : MatchedAll
[71347.440] VM(0/1/0) Tx DIGIT_IND '9'
64 <Call 14> : Digit(9) at CalleeDeterminedWaitDigit
65 <Call 14> : MatchedAll
[71347.820] VM(0/1/0) Tx DIGIT_IND '0'
66 <Call 14> : Digit(0) at CalleeDeterminedWaitDigit
67 <Call 14> : MatchedAll
[71348.260] VM(0/1/0) Tx DIGIT_IND '1'
68 <Call 14> : Digit(1) at CalleeDeterminedWaitDigit
69 <Call 14> : MatchedAll
[71348.630] VM(0/1/0) Tx DIGIT_IND '5'
70 <Call 14> : Digit(5) at CalleeDeterminedWaitDigit
71 <Call 14> : MatchedAll
[71348.990] VM(0/1/0) Tx DIGIT_IND '6'
72 <Call 14> : Digit(6) at CalleeDeterminedWaitDigit
73 <Call 14> : MatchedAll
74 <Time 14> : Inter digit timer timeout.
75 <Call 14> : Digit(#) at CalleeDeterminedWaitDigit
76 <Call 14> : MatchAllProcess After Sorted
<0> id(1006) dest(T) prefer(0) selected(8)
77 <Call 14> : Initiate callee with dial-peer(T) status(CalleeDetermi
nedAll) id(f81e3150-3861-51aa-8024-0002a4072aaa)
78 <NetEP 14> : InitiateOutCall: calledNum(89096890156) callingNum(102
) target(sip-server)
79 <NetEP 14> : DoCall: calledAddr(sip:89096890156@176.9.85.133:5060)
callingAddr(102)
[71351.900] VM(0/1/0) set T38 enable by CCC
[71351.900] VM(0/1/0) set T38 mode STD
[71351.900] VM(0/1/0) Fax rate 9600
80 <SIP 0> : No authentication information available
81 <SIP 14> : Send INVITE Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
INVITE sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK0250d225a420
From: <sip:102@176.9.85.133>;tag=0250d225a4
To: <sip:89096890156@176.9.85.133>
Call-ID: 021f3150-2115-d2bb-8025-0002a4072aaa@192.168.0.6
CSeq: 20 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Sun, 19 Aug 2012 17:14:42 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Contact: <sip:102@192.168.0.6>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 246
Max-Forwards: 70

v=0
o=102 1345396482 1345396482 IN IP4 192.168.0.6
s=AddPac Gateway SDP
c=IN IP4 192.168.0.6
t=1345396482 0
m=audio 23038 RTP/AVP 4 18 0 8
a=ptime:30
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000

[71351.940] RTA(0/1/0) Rx RS_LISTEN_REQ callId=14 ssId=1 G711U
peer=0.0.0.0 mp=23038/23039 hp=0/0
[71351.945] VM(0/1/0) codec same G711U

Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK0250d225a420;received=77.232.154
.207;rport=5060
From: <sip:102@176.9.85.133>;tag=0250d225a4
To: <sip:89096890156@176.9.85.133>;tag=as17aa7d69
Call-ID: 021f3150-2115-d2bb-8025-0002a4072aaa@192.168.0.6
CSeq: 20 INVITE
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="pbx.zadara.com", nonce="725dc3c9"

Content-Length: 0


82 <SIP 14> : Receive 401 Unauthorized
83 <SIP 14> : Transaction (20 INVITE) completed
84 <SIP 14> : Send ACK Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
ACK sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK0250d225a420
From: <sip:102@176.9.85.133>;tag=0250d225a4
To: <sip:89096890156@176.9.85.133>;tag=as17aa7d69
Call-ID: 021f3150-2115-d2bb-8025-0002a4072aaa@192.168.0.6
CSeq: 20 ACK
Content-Length: 0
Max-Forwards: 70


85 <SIP 0> : No opaque in authentication
86 <SIP 0> : Adding authentication information
87 <SIP 14> : Send INVITE Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
INVITE sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK0250d225a421
From: <sip:102@176.9.85.133>;tag=0250d225a4
To: <sip:89096890156@176.9.85.133>
Call-ID: 021f3150-2115-d2bb-8025-0002a4072aaa@192.168.0.6
CSeq: 21 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Sun, 19 Aug 2012 17:14:42 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2049-102", realm="pbx.zadara.com", nonce="725dc3
c9", uri="sip:89096890156@176.9.85.133", response="f69e3412dea7e44433c4c79a4fc5d
e36", algorithm=MD5
Contact: <sip:102@192.168.0.6>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 254
Max-Forwards: 70

v=0
o=102 1345396482 1345396482 IN IP4 192.168.0.6
s=AddPac Gateway SDP
c=IN IP4 192.168.0.6
t=1345396482 0
m=audio 23038 RTP/AVP 4 18 0 8
a=ptime:30
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1


Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK0250d225a421;received=77.232.154
.207;rport=5060
From: <sip:102@176.9.85.133>;tag=0250d225a4
To: <sip:89096890156@176.9.85.133>;tag=as17aa7d69
Call-ID: 021f3150-2115-d2bb-8025-0002a4072aaa@192.168.0.6
CSeq: 21 INVITE
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Content-Length: 0


88 <SIP 14> : Receive 403 Forbidden
89 <SIP 14> : Transaction (21 INVITE) completed
90 <SIP 0> : Adding authentication information
91 <SIP 14> : Send ACK Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
ACK sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK0250d225a421
From: <sip:102@176.9.85.133>;tag=0250d225a4
To: <sip:89096890156@176.9.85.133>;tag=as17aa7d69
Call-ID: 021f3150-2115-d2bb-8025-0002a4072aaa@192.168.0.6
CSeq: 21 ACK
Authorization: Digest username="2049-102", realm="pbx.zadara.com", nonce="725dc3
c9", uri="sip:89096890156@176.9.85.133", response="a9a2b4e2ca3eb19a04e4e7aaad8fd
10c", algorithm=MD5
Content-Length: 0
Max-Forwards: 70


92 <SIP 14> : Check Event Relation
93 <SIP 14> : ReleaseWithNothing
[71352.175] RTA(0/1/0) Rx RS_CLOSE_REQ callId=14 ssId=1 dir=reve
[71352.175] RTA(0/1/0) close Media socket
[71352.175] RTA(0/1/0) close RTCP socket
94 <Call 14> : Terminated from(fffffffe) this(Remote:NoPermission) be
fore(NULL) forced(0) time(1345396483)
95 <CEP 000100> : DisconnectCall at Busy
96 <CEP 000100> : StopSignal
[71352.180] RTA(0/1/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[71352.180] VM(0/1/0) play mute
97 <CEP 000100> : Disconnect (0)
[71352.180] RTA(0/1/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[71352.180] VM(0/1/0) play Reorder tone
98 <NetEP 14> : Call TO <sip:89096890156@176.9.85.133> terminated reas
on(Remote:NoPermission)
[71356.275] VM(0/1/0) vmOnHook
[71356.325] VM(0/1/0) vmTmoOnHook
[71356.375] VM(0/1/0) vmTmoOnHook
[71356.425] VM(0/1/0) vmTmoOnHook
[71356.475] VM(0/1/0) vmTmoOnHook
[71356.525] VM(0/1/0) vmTmoOnHook
[71356.575] VM(0/1/0) vmTmoOnHook
[71356.625] VM(0/1/0) vmTmoOnHook
[71356.675] VM(0/1/0) vmTmoOnHook
[71356.725] VM(0/1/0) vmTmoOnHook
[71356.775] VM(0/1/0) vmTmoOnHook
[71356.825] VM(0/1/0) vmTmoOnHook
[71356.875] VM(0/1/0) vmTmoOnHook
[71356.925] VM(0/1/0) vmTmoOnHook
[71356.975] VM(0/1/0) vmTmoOnHook
[71356.975] VM(0/1/0) Rx OnHook
[71356.975] VM(0/1/0) vopp idle
[71356.975] VM(0/1/0) vopp disabled
[71356.985] VM(0/1/0) VoPP ready
[71356.985] VM(0/1/0) Tx DISCONN_CNF
99 <CEP 000100> : Disconnected(16) at Disconnecting
no deb all
clear all debug informations
voip#
Автор:  genal [ 22 авг 2012, 10:54 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
А с порта с номером 100 звонки проходят на тот же номер? Если да выложите точно такой же лог, нас будет интересовать именно:


INVITE sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK0250d225a421
From: <sip:102@176.9.85.133>;tag=0250d225a4
To: <sip:89096890156@176.9.85.133>
Call-ID: 021f3150-2115-d2bb-8025-0002a4072aaa@192.168.0.6
CSeq: 21 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Sun, 19 Aug 2012 17:14:42 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2049-102", realm="pbx.zadara.com", nonce="725dc3
c9", uri="sip:89096890156@176.9.85.133", response="f69e3412dea7e44433c4c79a4fc5d
e36", algorithm=MD5
Contact: <sip:102@192.168.0.6>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 254
Max-Forwards: 70

v=0
o=102 1345396482 1345396482 IN IP4 192.168.0.6
s=AddPac Gateway SDP
c=IN IP4 192.168.0.6
t=1345396482 0
m=audio 23038 RTP/AVP 4 18 0 8
a=ptime:30
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1


Нужно будет сравнить в чем разница.
Автор:  viktorkc [ 31 авг 2012, 06:13 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
INVITE sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 INVITE
Supported: replaces, timer, 100rel, early-session
Min-SE: 1800
Date: Fri, 31 Aug 2012 15:10:16 GMT
Session-Expires: 1800
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2049-100", realm="pbx.zadara.com", nonce="5afe55
79", uri="sip:89096890156@176.9.85.133", response="a2f54ee7d2c8d5457a58f0b13c716
c41", algorithm=MD5
Contact: <sip:100@192.168.0.2>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 254
Max-Forwards: 70

v=0
o=100 1346425816 1346425816 IN IP4 192.168.0.2
s=AddPac Gateway SDP
c=IN IP4 192.168.0.2
t=1346425816 0
m=audio 24014 RTP/AVP 4 18 0 8
a=ptime:30
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1


Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509;received=93.188.18
8.162;rport=5060
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 INVITE
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89096890156@176.9.85.133:5060>
Content-Length: 0


121 <SIP 375> : Receive 100 Trying
122 <SIP 375> : Transaction (509 INVITE) proceeding

Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509;received=93.188.18
8.162;rport=5060
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>;tag=as5412da85
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 INVITE
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89096890156@176.9.85.133:5060>
Content-Length: 0


123 <SIP 375> : Receive 180 Ringing
124 <SIP 375> : Transaction (509 INVITE) proceeding
125 <Call 375> : Alert from(fffffffe) pseudo(0) inband(0) status(Callee
Initiated)
[1028319.080] RTA(0/0/0) Rx CC_ALERT_RSP peerId(0/0/0)
[1028319.080] VM(0/0/0) play RingBack tone

Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509;received=93.188.18
8.162;rport=5060
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>;tag=as5412da85
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 INVITE
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89096890156@176.9.85.133:5060>
Content-Type: application/sdp
Content-Length: 277

v=0
o=root 1212709662 1212709662 IN IP4 176.9.85.133
s=Asterisk PBX 10.3.0
c=IN IP4 176.9.85.133
t=0 0
m=audio 19246 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

126 <SIP 375> : Receive 183 Session Progress
127 <SIP 375> : Transaction (509 INVITE) proceeding
128 <SIP 375> : Received Session Progress response
129 <SIP 375> : Get SIP Audio MediaFormat : 18
130 <Call 375> : PreConnected from(fffffffe)
[1028319.115] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
VAD_CTRL 0
[1028319.115] VM(0/0/0) VAD disable
[1028319.115] VM(0/0/0) SID enable by CCC
[1028319.115] RTA(0/0/0) Rx CC_CONNECT_RSP peerId(0/0/0)
[1028319.115] VM(0/0/0) Fax enable
[1028319.115] VM(0/0/0) play mute
[1028319.120] RTA(0/0/0) Rx RS_OPEN_REQ callId=375 ssId=1 G729A
peer=176.9.85.133 mp=24014/24015 hp=19246/19247
[1028319.120] VM(0/0/0) vopp idle
[1028319.120] VM(0/0/0) vopp disabled
[1028319.120] VM(0/0/0) vopp skip sanity by codec change
[1028319.120] VM(0/0/0) start codec replace timer to G729A
[1028319.120] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_CTRL 1
[1028319.120] VM(0/0/0) DTMF_Signal enable
[1028319.125] VM(0/0/0) discard voice under codec replace
[1028319.130] VM(0/0/0) discard voice under codec replace
[1028319.180] VM(0/0/0) vopp enable
[1028319.180] VM(0/0/0) codec replaced to G729A
[1028319.180] VM(0/0/0) Fax enable
[1028319.180] VM(0/0/0) play mute
[1028319.335] VM(0/0/0) codec same G729A
[1028319.335] VM(0/0/0) Rx RTP replace codec to G729A
[1028330.065] VM(0/0/0) vmOnHook
[1028330.115] VM(0/0/0) vmTmoOnHook
[1028330.165] VM(0/0/0) vmTmoOnHook
[1028330.215] VM(0/0/0) vmTmoOnHook
[1028330.265] VM(0/0/0) vmTmoOnHook
[1028330.315] VM(0/0/0) vmTmoOnHook
[1028330.335] VM(0/0/0) vmOffHook
[1028330.395] VM(0/0/0) vmTmoOffHook
[1028330.395] VM(0/0/0) Rx OffHook
[1028330.395] VM(0/0/0) Tx FLASH_IND
131 <CEP 000000> : Hook Flashed
132 <Call 375> : HookFlashed from(0)
[1028330.545] VM(0/0/0) vmOnHook
[1028330.595] VM(0/0/0) vmTmoOnHook
[1028330.645] VM(0/0/0) vmTmoOnHook
[1028330.695] VM(0/0/0) vmTmoOnHook
[1028330.745] VM(0/0/0) vmTmoOnHook
[1028330.795] VM(0/0/0) vmTmoOnHook
[1028330.845] VM(0/0/0) vmTmoOnHook
[1028330.895] VM(0/0/0) vmTmoOnHook
[1028330.945] VM(0/0/0) vmTmoOnHook
[1028330.995] VM(0/0/0) vmTmoOnHook
[1028331.045] VM(0/0/0) vmTmoOnHook
[1028331.095] VM(0/0/0) vmTmoOnHook
[1028331.145] VM(0/0/0) vmTmoOnHook
[1028331.195] VM(0/0/0) vmTmoOnHook
[1028331.245] VM(0/0/0) vmTmoOnHook
[1028331.245] VM(0/0/0) Rx OnHook
[1028331.245] VM(0/0/0) vopp idle
[1028331.245] VM(0/0/0) vopp disabled
[1028331.255] VM(0/0/0) VoPP ready
[1028331.255] VM(0/0/0) Tx DISCONN_CNF
133 <CEP 000000> : Disconnected(16) at Busy
134 <Call 375> : Terminated from(0) this(Local:CallClear) before(NULL)
forced(0) time(1346425838)
135 <CEP 000000> : DisconnectCall at Idle
136 <SIP 375> : ReleaseWithCANCEL for 1 INVITEs
137 <SIP 375> : Send CANCEL Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
CANCEL sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 CANCEL
Date: Fri, 31 Aug 2012 15:10:38 GMT
User-Agent: AddPac SIP Gateway
Content-Length: 0
Max-Forwards: 70


[1028331.275] RTA(0/0/0) Rx RS_CLOSE_REQ callId=375 ssId=1 dir=all
[1028331.275] RTA(0/0/0) close Media socket
[1028331.275] RTA(0/0/0) close RTCP socket
138 <NetEP 375> : Call TO <sip:89096890156@176.9.85.133> terminated reas
on(Local:CallClear)

Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509;received=93.188.18
8.162;rport=5060
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>;tag=as5412da85
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 INVITE
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Content-Length: 0


139 <SIP 375> : Receive 487 Request Terminated
140 <SIP 375> : Transaction (509 INVITE) completed
141 <SIP 375> : Send ACK Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
ACK sip:89096890156@176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>;tag=as5412da85
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 ACK
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKd8500673a4509;received=93.188.18
8.162;rport=5060
From: <sip:100@176.9.85.133>;tag=d8500673a4
To: <sip:89096890156@176.9.85.133>;tag=as5412da85
Call-ID: d8d34050-c3bc-0630-8373-0002a4072aaa@192.168.0.2
CSeq: 509 CANCEL
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Content-Length: 0


142 <SIP 375> : Receive 200 OK
143 <SIP 375> : Transaction (509 CANCEL) completed
144 <SIP 375> : Set Terminated Success for 509 CANCEL
145 <Time 0> : SIP_TREGISTER timer timeout.
146 <SIP 0> : Adding authentication information
147 <SIP 84709> : Send REGISTER Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
REGISTER sip:176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43503f00a484709
From: <sip:2049-100@176.9.85.133>;tag=43503f00a4
To: sip:2049-100@176.9.85.133
Call-ID: 43233150-46e9-3f9c-8000-0002a4072aaa@192.168.0.6
CSeq: 84709 REGISTER
Date: Fri, 31 Aug 2012 15:10:46 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2049-100", realm="pbx.zadara.com", nonce="37bf36
a1", uri="sip:176.9.85.133", response="bf77983fa9d10a379c194656be285918", algori
thm=MD5
Contact: <sip:2049-100@192.168.0.2>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43503f00a484709;received=93.188.
188.162;rport=5060
From: <sip:2049-100@176.9.85.133>;tag=43503f00a4
To: sip:2049-100@176.9.85.133;tag=as4b571acb
Call-ID: 43233150-46e9-3f9c-8000-0002a4072aaa@192.168.0.6
CSeq: 84709 REGISTER
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="pbx.zadara.com", nonce="42d01c06"

Content-Length: 0


148 <SIP 84709> : Receive 401 Unauthorized
149 <SIP 84709> : Transaction (84709 REGISTER) completed
150 <SIP 0> : No opaque in authentication
151 <SIP 0> : Adding authentication information
152 <SIP 84710> : Send REGISTER Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
REGISTER sip:176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43503f00a484710
From: <sip:2049-100@176.9.85.133>;tag=43503f00a4
To: sip:2049-100@176.9.85.133
Call-ID: 43233150-46e9-3f9c-8000-0002a4072aaa@192.168.0.6
CSeq: 84710 REGISTER
Date: Fri, 31 Aug 2012 15:10:46 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2049-100", realm="pbx.zadara.com", nonce="42d01c
06", uri="sip:176.9.85.133", response="5f101a1ff7b33072997f8d9f51200fdb", algori
thm=MD5
Contact: <sip:2049-100@192.168.0.2>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43503f00a484710;received=93.188.
188.162;rport=5060
From: <sip:2049-100@176.9.85.133>;tag=43503f00a4
To: sip:2049-100@176.9.85.133;tag=as4b571acb
Call-ID: 43233150-46e9-3f9c-8000-0002a4072aaa@192.168.0.6
CSeq: 84710 REGISTER
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Expires: 60
Contact: <sip:2049-100@192.168.0.2>;expires=60
Date: Fri, 31 Aug 2012 06:07:42 GMT
Content-Length: 0


153 <SIP 84710> : Receive 200 OK
154 <SIP 84710> : Transaction (84710 REGISTER) completed
155 <SIP 0> : Adding authentication information
156 <SIP 84711> : Send REGISTER Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
REGISTER sip:176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43509301a484711
From: <sip:2049-102@176.9.85.133>;tag=43509301a4
To: sip:2049-102@176.9.85.133
Call-ID: 43233150-b804-9323-8001-0002a4072aaa@192.168.0.6
CSeq: 84711 REGISTER
Date: Fri, 31 Aug 2012 15:10:47 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2049-102", realm="pbx.zadara.com", nonce="475659
13", uri="sip:176.9.85.133", response="484e17c1579a0b232435bcb8b67a5f82", algori
thm=MD5
Contact: <sip:2049-102@192.168.0.2>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 176.9.85.133:5060 )
NOTIFY sip:2049-100@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 176.9.85.133:5060;branch=z9hG4bK0cf1697d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@176.9.85.133>;tag=as748c67e9
To: <sip:2049-100@192.168.0.2>
Contact: <sip:asterisk@176.9.85.133:5060>
Call-ID: 0c4329d225810b0c1bde786d553c11e3@176.9.85.133:5060
CSeq: 102 NOTIFY
User-Agent: Zadarma PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:8500@176.9.85.133
Voice-Message: 0/0 (0/0)


Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 176.9.85.133:5060;branch=z9hG4bK0cf1697d;rport
From: "asterisk" <sip:asterisk@176.9.85.133>;tag=as748c67e9
To: <sip:2049-100@192.168.0.2>
Call-ID: 0c4329d225810b0c1bde786d553c11e3@176.9.85.133:5060
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0



Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43509301a484711;received=93.188.
188.162;rport=5060
From: <sip:2049-102@176.9.85.133>;tag=43509301a4
To: sip:2049-102@176.9.85.133;tag=as716f887b
Call-ID: 43233150-b804-9323-8001-0002a4072aaa@192.168.0.6
CSeq: 84711 REGISTER
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="pbx.zadara.com", nonce="04942132"

Content-Length: 0


157 <SIP 84711> : Receive 401 Unauthorized
158 <SIP 84711> : Transaction (84711 REGISTER) completed
159 <SIP 0> : No opaque in authentication
160 <SIP 0> : Adding authentication information
161 <SIP 84712> : Send REGISTER Request

Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
REGISTER sip:176.9.85.133 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43509301a484712
From: <sip:2049-102@176.9.85.133>;tag=43509301a4
To: sip:2049-102@176.9.85.133
Call-ID: 43233150-b804-9323-8001-0002a4072aaa@192.168.0.6
CSeq: 84712 REGISTER
Date: Fri, 31 Aug 2012 15:10:47 GMT
User-Agent: AddPac SIP Gateway
Authorization: Digest username="2049-102", realm="pbx.zadara.com", nonce="049421
32", uri="sip:176.9.85.133", response="452960453551528011a12301160f4773", algori
thm=MD5
Contact: <sip:2049-102@192.168.0.2>;expires=60
Expires: 60
Content-Length: 0
Max-Forwards: 70



Received SIP PDU from ( 176.9.85.133:5060 )
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK43509301a484712;received=93.188.
188.162;rport=5060
From: <sip:2049-102@176.9.85.133>;tag=43509301a4
To: sip:2049-102@176.9.85.133;tag=as716f887b
Call-ID: 43233150-b804-9323-8001-0002a4072aaa@192.168.0.6
CSeq: 84712 REGISTER
Server: Zadarma PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLIS
H
Supported: replaces, timer
Expires: 60
Contact: <sip:2049-102@192.168.0.2>;expires=60
Date: Fri, 31 Aug 2012 06:07:42 GMT
Content-Length: 0


162 <SIP 84712> : Receive 200 OK
163 <SIP 84712> : Transaction (84712 REGISTER) completed

Received SIP PDU from ( 176.9.85.133:5060 )
NOTIFY sip:2049-102@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 176.9.85.133:5060;branch=z9hG4bK41e7d25f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@176.9.85.133>;tag=as2ca454ca
To: <sip:2049-102@192.168.0.2>
Contact: <sip:asterisk@176.9.85.133:5060>
Call-ID: 392d767669be924363dbbbff66705072@176.9.85.133:5060
CSeq: 102 NOTIFY
User-Agent: Zadarma PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88

Messages-Waiting: no
Message-Account: sip:8500@176.9.85.133
Voice-Message: 0/0 (0/0)


Sending SIP PDU to ( 176.9.85.133:5060 ) from 5060
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 176.9.85.133:5060;branch=z9hG4bK41e7d25f;rport
From: "asterisk" <sip:asterisk@176.9.85.133>;tag=as2ca454ca
To: <sip:2049-102@192.168.0.2>
Call-ID: 392d767669be924363dbbbff66705072@176.9.85.133:5060
CSeq: 102 NOTIFY
User-Agent: AddPac SIP Gateway
Content-Length: 0
Автор:  genal [ 31 авг 2012, 07:29 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
Хм. Странно, не вижу разницы в инвайтах, но в первом случае сип-сервер дает нам 403 ошибку (с номера 102), а с номера 100 мы получаем trying и ringing (все нормально).
На сип сервере нет никаких ограничений?
Автор:  viktorkc [ 01 сен 2012, 01:31 ]
Заголовок сообщения:  Re: AP 1100F проблема с исходящей связью
с тех.под общался...говорят проблема в оборудовании, поставил доп voip-шлюз...работает
Страница 1 из 1 Часовой пояс: UTC
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