Доброе утро. Есть AP1000f 4 FXS. Настрое на Астериск по сип. На 0/0-0/3 порты прописаны учетки, каждая со своим номером, при звонке на шлюз по каждому из номеров - звонок проходит и связь работает, но если звонить с любого из номеров то звонок определяется номером с порта 0/0. При этом, если позвонить с аккаунта на астериске то нормер определится верно.
В чем может быть проблема?
Код:
AP1000# sh run
!
version 8.30K
!
hostname AP1000
!
!
no bridge spanning-tree
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.126 255.255.255.128
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address dhcp
!
interface ether1.0
no ip address
!
snmp name AP1000
!
no arp reset
!
!
!
!
!
! 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
!
!
! FXS
voice-port 0/2
caller-id enable
!
!
! FXS
voice-port 0/3
caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern ***7969
port 0/0
user-name 6*****a
user-password ********
!
dial-peer voice 2 pots
destination-pattern ***7989
port 0/1
user-name c*****0
user-password ********
!
!
!
! Voip peer configuration.
!
dial-peer voice 100 voip
destination-pattern T
session target sip-server
session protocol sip
codec g729
dtmf-relay rtp-2833
no vad
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.21.32.5
no ignore-msg-from-other-gk
!
!
! SIP UA configuration.
!
sip-ua
user-register
sip-server 46.2*.*.*
register e164
!
!
! MGCP configuration.
!
mgcp
codec g711ulaw
vad
!
!
! Tones
!
!
!
!
Лог исходящего звонка
deb voip call:
Код:
AP1000# deb voip call
AP1000# 1 <CEP 000100> : Call Received
2 <CEP 000100> : Call Initiated : calledNumber() crv(0) total(0)
3 <Call 8> : ****************** Call Created status(InitiatedByFXS
) *******************
4 <CEP 000100> : Calling number(***7989)
5 <CEP 000100> : Call id(f4a8a54e-5815-6576-8012-0002a404f92a) callNum(
8)
6 <Call 8> : Digit(8) at InitiatedByFXS
7 <Call 8> : MatchedAll
8 <Call 8> : Digit(9) at CalleeDeterminedWaitDigit
9 <Call 8> : MatchedAll
10 <Call 8> : Digit(2) at CalleeDeterminedWaitDigit
11 <Call 8> : MatchedAll
12 <Call 8> : Digit(7) at CalleeDeterminedWaitDigit
13 <Call 8> : MatchedAll
14 <Call 8> : Digit(2) at CalleeDeterminedWaitDigit
15 <Call 8> : MatchedAll
16 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit
17 <Call 8> : MatchedAll
18 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit
19 <Call 8> : MatchedAll
20 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit
21 <Call 8> : MatchedAll
22 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit
23 <Call 8> : MatchedAll
24 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit
25 <Call 8> : MatchedAll
26 <Call 8> : Digit(*) at CalleeDeterminedWaitDigit
27 <Call 8> : MatchedAll
28 <Time 8> : Inter digit timer timeout.
29 <Call 8> : Digit(#) at CalleeDeterminedWaitDigit
30 <Call 8> : MatchAllProcess After Sorted
<0> id(100) dest(T) prefer(0) selected(2)
31 <Call 8> : Initiate callee with dial-peer(T) status(CalleeDetermi
nedAll) id(f4a8a54e-5815-6576-8012-0002a404f92a)
32 <NetEP 8> : InitiateOutCall: calledNum(89272******) callingNum(205
7989) target(sip-server)
33 <NetEP 8> : DoCall: calledAddr(sip:89272******@46.2*.*.*:5060) ca
llingAddr(2057989)
34 <SIP 0> : No authentication information available
35 <SIP 8> : Send INVITE Request
36 <SIP 8> : Receive 100 Trying
37 <SIP 8> : Transaction (6 INVITE) proceeding
38 <SIP 8> : Receive 200 OK
39 <SIP 8> : Get SIP Audio MediaFormat : 18
40 <Call 8> : Connected from(fffffffe)
41 <NetEP 8> : Call with sip:89272******@46.2*.*.* established
42 <SIP 8> : Received INVITE OK response
43 <SIP 8> : Send ACK Request
44 <SIP 8> : Check Event Relation
45 <SIP 8> : Set Terminated Success for 6 INVITE
46 <Time 0> : SIP_TREGISTER timer timeout.
47 <SIP 0> : Adding authentication information
48 <SIP 4735> : Send REGISTER Request
49 <SIP 4735> : Receive 401 Unauthorized
50 <SIP 4735> : Transaction (4735 REGISTER) completed
51 <SIP 0> : No opaque in authentication
52 <SIP 0> : Adding authentication information
53 <SIP 4736> : Send REGISTER Request
54 <SIP 4736> : Receive 200 OK
55 <SIP 4736> : Transaction (4736 REGISTER) completed
56 <SIP 4736> : Send SNMP TRAP (success with code - 200) previous succ
ess
57 <SIP 0> : Adding authentication information
58 <SIP 4737> : Send REGISTER Request
59 <SIP 4737> : Receive 401 Unauthorized
60 <SIP 4737> : Transaction (4737 REGISTER) completed
61 <SIP 0> : No opaque in authentication
62 <SIP 0> : Adding authentication information
63 <SIP 4738> : Send REGISTER Request
64 <SIP 4738> : Receive 200 OK
65 <SIP 4738> : Transaction (4738 REGISTER) completed
66 <SIP 4738> : Send SNMP TRAP (success with code - 200) previous succ
ess
67 <SIP 4735> : Set Terminated Success for 4735 REGISTER
68 <SIP 4736> : Set Terminated Success for 4736 REGISTER
69 <SIP 4737> : Set Terminated Success for 4737 REGISTER
70 <SIP 4738> : Set Terminated Success for 4738 REGISTER
71 <CEP 000100> : Disconnected(16) at Busy
72 <Call 8> : Terminated from(100) this(Local:CallClear) before(NUL
L) forced(0)
73 <CEP 000100> : DisconnectCall at Idle
74 <SIP 8> : ReleaseWithBYE
75 <SIP 8> : Send BYE Request
76 <NetEP 8> : Call TO <sip:89272******@46.2*.*.*> terminated reason
(Local:CallClear)
77 <SIP 8> : Receive 200 OK
78 <SIP 8> : Transaction (7 BYE) completed
79 <SIP 8> : Set Terminated Success for 7 BYE
Звонок пришел с номера ***7969
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