Пытаюсь прозвониться с сипа на gsm звонок приходит на шлюз там выбирается gsm а дальше не идет, в итоге потом возвращается на сип на тот же номер
Received SIP PDU from ( 192.168.0.185:5060 )
INVITE sip:89182010597@192.168.0.200 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>
Call-ID: 5c9973664db548a7b8242318818135bb
User-Agent: Infra Call Center Server 4.0.636.187
Contact: <sip:192.168.0.185:5060;transport=udp>
CSeq: 2 INVITE
Allow: INVITE,CANCEL,ACK,BYE,OPTIONS
Content-Type: application/sdp
Content-Length: 192
v=0
o=- 22190 1843 IN IP4 192.168.0.185
s=-
c=IN IP4 192.168.0.185
t=0 0
m=audio 7040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
[1964.301] Sending SIP PDU to ( 192.168.0.185:5060 ) from 5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0
38 <Call 2> : ****** Call Created status(InitiatedByNet) ver(8.28:2
006-02-06-00-00) time(1299114905) ****
39 <SIP 2> : Receive INVITE Request
40 <NetCon 2> : Found inbound voip peer by IP address id(100)
41 <Call 2> : From Net - calledParty(89182010597) callingParty(102)
42 <Call 2> : MatchedAll
43 <Call 2> : MatchAllProcess After Sorted
<0> id(4584) dest([78]9[18]T) prefer(0) selected(0)
<1> id(3561) dest(T) prefer(0) selected(0)
<2> id(3560) dest(T) prefer(0) selected(1)
44 <Call 2> : Initiate callee with dial-peer([78]9[18]T) status(Call
eeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
45 <CEP 000000> : InitiateOutCall : calledNum(89182010597), callingNum(
102), callerPort(ffffffff) type(GSM)
46 <CEP 000000> : Outbound call to CEP callId(00000000-0000-0000-0000-00
0000000000) callNum(2)
47 <SIP 2> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE
)
48 <SIP 2> : SetLocalAudioFormats : myVoipPeer(100) is not NULL, co
dec(0)
49 <PhonePlay 2> : Audio Count(1)
50 <PhonePlay 2> : rtpSessionId(1) Second Audio Port(-1)
51 <SIP 2> : SetAlerting
52 <Call 2> : PreConnected from(0)
53 <SIP 2> : SetLocalAudioFormats : outbound(FALSE) hqaEnable(FALSE
)
54 <SIP 2> : SetLocalAudioFormats : myVoipPeer(100) is not NULL, co
dec(0)
55 <SIP 2> : Add Local Audio MediaFormat : 8
[1964.325] Sending SIP PDU to ( 192.168.0.185:5060 ) from 5060
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>;tag=994d3001a4
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:89182010597@192.168.0.200
Content-Type: application/sdp
Content-Length: 196
v=0
o=89182010597 1299114905 1299114905 IN IP4 192.168.0.200
s=AddPac Gateway SDP
c=IN IP4 192.168.0.200
t=1299114905 0
m=audio 23004 RTP/AVP 8
a=ptime:20
a=rtpmap:8 PCMA/8000
a=sendrecv
56 <Time 2> : Call Forwarding No Answer timer timeout.
57 <CEP 000000> : Disconnected(16) at Busy
58 <Call 2> : Terminated from(0) this(Local:CallClear) before((null)
) forced(0) time(1299114922)
[1981.518] Sending SIP PDU to ( 192.168.0.185:5060 ) from 5060
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
To: <sip:89182010597@192.168.0.200>;tag=994d3001a4
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 INVITE
User-Agent: AddPac SIP Gateway
Content-Length: 0
59 <NetEP 2> : Call FROM <Oper2> terminated reason(Local:CallClear)
60 <CEP 000000> : DisconnectCall at Idle
Received SIP PDU from ( 192.168.0.185:5060 )
ACK sip:89182010597@192.168.0.200 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 192.168.0.185:5060;branch=z9hG4bK92e4056352ce4bc1aa9d259fe3e314
b
From: "Oper2" <sip:102@192.168.0.185>;tag=98c370edc21f424f85e2a5f8f6f684fa
User-Agent: Infra Call Center Server 4.0.636.187
To: <sip:89182010597@192.168.0.200>;tag=994d3001a4
Call-ID: 5c9973664db548a7b8242318818135bb
CSeq: 2 ACK
Content-Length: 0
61 <SIP 2> : Receive ACK Request
62 <SIP 2> : Set Terminated Success for 2 INVITE
Welcome, APOS(tm) Kernel Version 8.51.002.
Copyright (c) 1999-2010 AddPac Technology Co., Ltd.
Login:
Login: root
Password:
GS1002> enable
GS1002# show running-config
Building configuration...
Current configuration:
!
version 8.51.002
!
hostname GS1002
!
username root password router administrator
username guest password guest user
!
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 192.168.0.200 255.255.255.0
speed auto
no qos-control
!
interface FastEthernet0/1
ip address 192.168.10.1 255.255.255.0
speed auto
no qos-control
!
ip route 0.0.0.0 0.0.0.0 192.168.0.1 10
!
!
!
!
http server
!
logging command
logging event 4-warning
logging on
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
protocol sip
dtmf-relay out-of-band
fax protocol t38 redundancy 0
fax rate disable
h323 call start fast
h323 call tunnel enable
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
connection plar 79881413242
translate-incoming called-number 900
caller-id enable
!
!
! GSM
voice-port 0/1
connection plar 79284452546
translate-incoming called-number 1
caller-id enable
!
!
! FXO
voice-port 0/2
connection plar 78622649060
ring detect-timeout 50
ring detect-timer 500
caller-id enable
caller-id type etsi
caller-id name disable
shutdown
!
!
! FXO
voice-port 0/3
connection plar 78622644314
ring detect-timeout 50
ring detect-timer 500
caller-id enable
caller-id type etsi
caller-id name disable
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
shutdown
!
dial-peer voice 3560 pots
destination-pattern T
port 0/2
!
dial-peer voice 3561 pots
destination-pattern T
port 0/3
!
dial-peer voice 4584 pots
destination-pattern [78]9[18]T
port 0/0
!
dial-peer voice 4585 pots
destination-pattern 79[2345679]T
port 0/1
!
!
!
! Voip peer configuration.
!
dial-peer voice 100 voip
destination-pattern 7T
session target ip 192.168.0.185
session protocol sip
codec g711alaw
no vad
dtmf-relay info
!
dial-peer voice 101 voip
destination-pattern 79881413242
session target ip 192.168.0.185
session protocol sip
codec g711alaw
no vad
dtmf-relay info
!
dial-peer voice 102 voip
destination-pattern 79284452546
session target ip 192.168.0.185
session protocol sip
codec g711alaw
no vad
dtmf-relay info
!
dial-peer voice 103 voip
destination-pattern 78622649060
session target ip 192.168.0.185
session protocol sip
codec g711alaw
no vad
dtmf-relay info
!
dial-peer voice 104 voip
destination-pattern 78622644314
session target ip 192.168.0.185
session protocol sip
codec g711alaw
no vad
dtmf-relay info
!
!
!
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.0.200
no ignore-msg-from-other-gk
!
!
! Translation Rule configuration.
!
translation-rule 1
rule 1 T 111T
!
translation-rule 2
rule 1 111T T
!
translation-rule 3
rule 0 7T 8T
!
translation-rule 900
rule 0 T 79881413242
!
!
!
! SIP UA configuration.
!
sip-ua
!
!
! Tones
voice class clear-down-tone 0 432 0 352 346
!
!
!
!
!
line console
!
line vty
!
gsm dev-restart-by-unreg 300
!
gsm 0/0
sms-language utf8
!
gsm 0/1
sms-language utf8
!
end
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