I need you help, now my Sip network is running, when I make a call it will no not ring but on the phone it will read call fail no answer. When I debugged the ccsip I get the following result:
debug ccsip:
Calling Number : 4047881723
Called Number : 4049212048
Source IP Address (Sig ): 72.244.194.141
Destn SIP Req Addr:Port : 216.143.130.94:0
Destn SIP Resp Addr:Port : 216.143.130.94:5060
Destination Name : 216.143.130.94
*Mar 23 01:40:06.507: //3/BCD0A45B8006/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729r8
Negotiated Codec Bytes : 0
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0
Source IP Address (Media): 0.0.0.0
Source IP Port (Media): 0
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 9428
Orig Destn IP Address:Port (Media): 0.0.0.0:0
*Mar 23 01:40:06.507: //3/BCD0A45B8006/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Debug ccsip messages:
Date: Sat, 23 Mar 2002 01:45:58 GMT
Call-ID: 506bfbb75329ebfe6e62868e0784c6a8@216.143.130.94
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
*Mar 23 01:45:58.271: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:4049212048@72.244.194.141 SIP/2.0
Via: SIP/2.0/UDP 216.143.130.94:5060;branch=z9hG4bK0dd142b9
From: "4047881723" <sip:4047881723@216.143.130.94>;tag=as0ad5ab52
To: <sip:4049212048@72.244.194.141>;tag=92914-11C0
Contact: <sip:4047881723@216.143.130.94>
Call-ID: 506bfbb75329ebfe6e62868e0784c6a8@216.143.130.94
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
Below is my cofig.....
version 12.4
hostname 3640
enable secret 5
enable password
resource policy
clock timezone GMT 5
voice-card 1
ip subnet-zero
ip tcp path-mtu-discovery
isdn switch-type primary-5ess
isdn voice-call-failure 0
!
voice service voip
signaling forward unconditional
sip
ds0-num
no call service stop
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g723r63
!
voice class codec 99
codec preference 1 g729r8
!
voice class codec 100
codec preference 1 g729r8
controller T1 1/0
framing esf
clock source internal
linecode b8zs
cablelength short 133
pri-group timeslots 1-24
!
controller T1 1/1
framing esf
linecode b8zs
cablelength short 133
pri-group timeslots 1-24
!
interface FastEthernet0/0
ip address 72.244.194.141 255.255.255.248
speed auto
half-duplex
!
interface Serial1/0:23
no ip address
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
isdn supp-service name calling
!
interface Serial1/1:23
no ip address
isdn switch-type primary-5ess
isdn protocol-emulate network
isdn incoming-voice voice
isdn supp-service name calling
!
ip default-gateway 72.244.194.137
ip http server
ip classless
ip route 0.0.0.0 0.0.0.0 72.244.194.137
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0
ip route 72.244.194.136 255.255.255.248 72.244.194.137
!
access-list 100 permit tcp any any eq telnet
access-list 101 permit ip any any
access-list 102 permit udp any any range 16384 32767
access-list 103 permit tcp any eq 1720 any
access-list 103 permit tcp any any eq 1720
snmp-server community RO
control-plane
!
voice-port 1/0:23
!
voice-port 1/1:23
!
dial-peer voice 1 pots
service session
destination-pattern .T
port 1/0:23
!
dial-peer voice 2 voip
service session
destination-pattern .T
session protocol sipv2
session target ipv4:216.143.130.94
session transport udp
!
dial-peer voice 4 pots
service session
destination-pattern .T
port 1/1:23
!
dial-peer voice 5 voip
service session
destination-pattern .T
session protocol sipv2
session target ipv4:216.143.130.92
session transport udp
!
gateway
timer receive-rtp 1200
!
sip-ua
nat symmetric role active
nat symmetric check-media-src
retry register 10
sip-server dns:sip-server
!
line con 0
line aux 0
line vty 0 4
password
login
!
end
