Hi everyone,
I'm new with CME and I'm have a problem to make a call from SCCP to SIP phone, SIP phone registered fine under Voice Register Global, I can make the call from SIP to SCCP and working perfectly, example: SIP(ext 2001) dial 1002 ---> SCCP(1002) work fine, SCCP(ext 1002) dial 2001 ---> SIP(ext 2001) not working, below is my config and debug output, please help as I have tried to sovel this problem for a last few days ..without luck
I have modified the config to make it shorter.
I made a call from ext 1002 to ext 2001 and debuged.
-------------------------------------------------------------------
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
no fax-relay sg3-to-g3
h323
modem passthrough nse codec g711ulaw
sip
registrar server expires max 250 min 200
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
!
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 20
max-pool 10
authenticate register
voicemail 5003
tftp-path flash:
file text
create profile sync 001902402159062A
!
voice register dn 1
number 2001
allow watch
mwi
!
voice register dn 2
number 2002
allow watch
mwi
!
voice register pool 1
id mac 08GF.45NH.33BG
number 1 dn 1
template 1
presence call-list
dtmf-relay sip-notify
username 2001 password test1
codec g711ulaw
!
voice register pool 2
id mac DF34.98KJ.FGH1
number 1 dn 2
presence call-list
dtmf-relay rtp-nte
username 2002 password test2
codec g711ulaw
------------------------------------------------------------------------------------
!
dial-peer voice 13 voip
translation-profile outgoing PSTN_OUTGOING
destination-pattern 9[0-1][2-9]........
session protocol sipv2
session target sip-server
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 11 voip
translation-profile outgoing PSTN_OUTGOING
destination-pattern 9001T
session protocol sipv2
session target sip-server
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 12 voip
translation-profile outgoing PSTN_OUTGOING
destination-pattern 9[2-9].......
session protocol sipv2
session target sip-server
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 21 pots
service aa
incoming called-number .
direct-inward-dial
port 0/3/0
!
dial-peer voice 100 voip
service aa
destination-pattern 123456789
session target sip-server
incoming called-number 123456789
dtmf-relay rtp-nte
codec g711ulaw
no vad
-------------------------------------------------------------------------
interface Loopback0
ip address 100.100.100.100 255.255.255.0
!
!
interface FastEthernet0/0
ip address 172.16.3.1 255.255.255.0
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
!
!
interface FastEthernet0/1
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
!
Interface vlan 1
ip address 10.0.0.1 255.255.255.0
ip nat inside
--------------------------------------------------------------------------------
telephony-service
em logout 0:0 0:0 0:0
max-ephones 20
max-dn 50
ip source-address 10.0.0.1 port 2000
max-conferences 8 gain -6
moh en_bacd_music_on_hold.au
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
ephone-dn 2 dual-line
number 1002
!
ephone 2
device-security-mode none
mac-address 1234.339C.4321
button 1:2
----------------------------------------------------------------------
DEBUG OUTPUT:
The Call Setup Information is:
Call Control Block (CCB) : 0x4A6F46F0
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 1002
Called Number : 2001
Source IP Address (Sig ): 124.184.95.113
Destn SIP Req Addr:Port : 203.161.160.71:5060
Destn SIP Resp Addr:Port : 203.161.160.71:5060
Destination Name : trunk.engin.com.au
*Oct 9 17:17:53.509: //4319/178076CE8755/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 124.184.95.113
Source IP Port (Media): 19548
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Oct 9 17:17:53.509: //4319/178076CE8755/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 1
Disconnect Cause (SIP) : 604
---------------------------------------------------------------------------
DEBUG OUTPUT:
Router#
*Oct 9 17:21:15.261: //4348/8FD13E9C8769/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
*Oct 9 17:21:15.269: //4348/8FD13E9C8769/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 124.184.95.113
*Oct 9 17:21:15.269: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 18630 for stream 1
*Oct 9 17:21:15.273: //4348/8FD13E9C8769/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
*Oct 9 17:21:15.273: //4348/8FD13E9C8769/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Oct 9 17:21:15.273: //4348/8FD13E9C8769/SIP/Media/sipSPIProcessRtpSessions: sipSPIProcessRtpSessions
*Oct 9 17:21:15.273: //4348/8FD13E9C8769/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 4348) to the VOIP RTP library
*Oct 9 17:21:15.273: //4348/8FD13E9C8769/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 124.184.95.113
*Oct 9 17:21:15.277: //4348/8FD13E9C8769/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Oct 9 17:21:15.277: //4348/8FD13E9C8769/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 124.184.95.113, lport = 18630, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 4348, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1
*Oct 9 17:21:15.277: //4348/8FD13E9C8769/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Oct 9 17:21:15.277: //4348/8FD13E9C8769/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Oct 9 17:21:15.345: //4348/8FD13E9C8769/SIP/Media/sipSPIDestroyRtpSession: stream:4AB72544
---------------------------------------------------------------------------
DEBUG OUTPUT:
Router#debug ccsip messages
SIP Call messages tracing is enabled
Router#
*Oct 9 17:28:53.909: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:2001@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 124.184.95.113:5060;branch=z9hG4bKA890
From: <sip:1002@trunk.engin.com.au>;tag=27EF894-76E
To: <sip:2001@192.168.1.2>
Date: Tue, 09 Oct 2012 17:28:53 GMT
Call-ID: A1D7FDE3-116D11E2-8795CE13-5722CA25@124.184.95.113
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2704086970-0292360674-2274414099-1461897765
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1349803733
Contact: <sip:1002@124.184.95.113:5060>
Call-Info: <sip:124.184.95.113:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 270
v=0
o=CiscoSystemsSIP-GW-UserAgent 8919 3652 IN IP4 124.184.95.113
s=SIP Call
c=IN IP4 124.184.95.113
t=0 0
m=audio 18106 RTP/AVP 0 100 19
c=IN IP4 124.184.95.113
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:19 CN/8000
a=ptime:20
*Oct 9 17:28:53.953: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 124.184.95.113:5060;branch=z9hG4bKA890
From: <sip:1002@trunk.engin.com.au>;tag=27EF894-76E
To: <sip:2001@192.168.1.2>
Call-ID: A1D7FDE3-116D11E2-8795CE13-5722CA25@124.184.95.113
CSeq: 101 INVITE
Timestamp: 1349803733
*Oct 9 17:28:53.969: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 604 Does not exist anywhere
Via: SIP/2.0/UDP 124.184.95.113:5060;branch=z9hG4bKA890
From: <sip:1002@trunk.engin.com.au>;tag=27EF894-76E
To: <sip:2001@192.168.1.2>;tag=SD0s3fa99-1560815025-1349800544451
Call-ID: A1D7FDE3-116D11E2-8795CE13-5722CA25@124.184.95.113
CSeq: 101 INVITE
Timestamp: 1349803733
Content-Length: 0
*Oct 9 17:28:53.981: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:2001@192.168.1.2:5060 SIP/2.0
Via: SIP/2.0/UDP 124.184.95.113:5060;branch=z9hG4bKA890
From: <sip:1002@trunk.engin.com.au>;tag=27EF894-76E
To: <sip:2001@192.168.1.2>;tag=SD0s3fa99-1560815025-1349800544451
Date: Tue, 09 Oct 2012 17:28:53 GMT
Call-ID: A1D7FDE3-116D11E2-8795CE13-5722CA25@124.184.95.113
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= unknown
Content-Length: 0
