Hi,
I need when the calls come from my PSTN line number 88887777 (voice port 0/1/1), the router dial over my sip trunk to number 03399883381..
But when i do the test, the follow error happens:
003481: Jul 16 22:12:07.926: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:03399883381@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>
Date: Tue, 16 Jul 2013 22:12:07 GMT
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 2528974697-3986362850-2933118957-3360635254
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF Y, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 15
Timestamp: 1374012727
Contact: <sip:y.y.y.y:5060>
Diversion: <sip:787@y.y.y.y>;privacy=off;reason=unconditional;counter=1;s creen=no
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 310
v=0
o=CiscoSystemsSIP-GW-UserAgent 3685 6234 IN IP4 y.y.y.y
s=SIP Call
c=IN IP4 y.y.y.y
t=0 0
m=audio 17930 RTP/AVP 0 8 101 19
c=IN IP4 y.y.y.y
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=direction:passive
003482: Jul 16 22:12:07.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9;rport=60371;received= y.y.y.y
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
CSeq: 101 INVITE
Server: nt
Content-Length: 0
003483: Jul 16 22:12:07.942: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9;rport=60371;received= y.y.y.y
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>;tag=18f01cbe78aa7d69fd8f3e0a8ea294be.a643
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
CSeq: 101 INVITE
Server: nt
Content-Length: 0
003484: Jul 16 22:12:07.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:03399883381@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK1365BB9
From: <sip:gila.local>;tag=146C004-236C
To: <sip:03399883381@X.X.X.X>;tag=18f01cbe78aa7d69fd8f3e0a8ea294be.a643
Date: Tue, 16 Jul 2013 22:12:07 GMT
Call-ID: 96BD1769-ED9B11E2-AED7D7ED-C84F3976@gila.local
Max-Forwards: 15
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
The router config is:
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server expires max 3600 min 3600
localhost dns:XXXX.local
no update-callerid
!
voice class codec 2
codec preference 1 g729r8
codec preference 2 g729br8
voice-port 0/1/1
supervisory disconnect dualtone mid-call
connection plar 787
caller-id enable
sip-ua
nat symmetric role passive
nat symmetric check-media-src
max-forwards 15
timers trying 1000
timers connect 1000
timers disconnect 1000
timers notify 1000
timers info 1000
sip-server ipv4:X.X.X.X:5060
telephony-service
video
no auto-reg-ephone
max-ephones 22
max-dn 88
ip source-address 10.176.10.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
time-zone 18
time-format 24
date-format dd-mm-yy
voicemail 300
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh music-on-hold.au
multicast moh 239.10.16.16 port 2000
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
transfer-pattern 0.T
secondary-dialtone 0
create cnf-files version-stamp 7960 May 27 2013 20:13:10
ephone-dn 31 dual-line
number 787 no-reg primary
call-forward all 03399883381
!
dial-peer voice 1 voip
destination-pattern 03399883381
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
