Harry,
Enabling or disabling ANI and DNIS will not effect the routing, if anything enabling them will allow you too see what is being sent by CM to Nortel on Nortel phone display.
Joel Sisko
Main Topics
Browse All TopicsI currently have a VOIP setup with an H.323 Gateway and CallManager 4.0 using MGCP using a T1 CAS E&M-Wink-Start to connect to a Nortel MICS. I have configured the T1 to use E&M Wink Start. Signalling seems to be working perfect, the only issue I have is calling from the Cisco to the Nortel.
From the Nortel MICS, I am able to seize the trunks and make calls to the call Manager. However, when I try to call from a Cisco IP phone to the Nortel MICS, it does not work. The prime set on the MICS also does not pick up so i am guessing some kind of digit forwarding is the cause of the problem. I have enabled both H.323 and MGCP on this router to determine if H.323 or MGCP is the cause. I get the same problem for both so I assume it is some kind of router setting. I don't think it is my nortel side since i have an Analog E&M setup right now and it works perfect both ways. I am guessing it is a problem with the T1 but the nortel to cisco works perfectly, just the cisco to nortel doesn't work right now.
I issue the Debug Statement on the router : debug vtsp all or debug vtsp dsl or debug voip vstp all
*Dec 29 14:12:30.731: //581/82E364CF818D/VTSP:(0
Feature ID=0, Feature Status=1
*Dec 29 14:12:30.731: //581/82E364CF818D/VTSP:(0
*Dec 29 14:12:30.731: //581/82E364CF818D/VTSP:(0
*Dec 29 14:12:31.136: //581/82E364CF818D/VTSP:(0
[state:S_SETUP_REQ_PROC, event:E_TSP_CALL_FEATURE_I
*Dec 29 14:12:31.140: //581/82E364CF818D/VTSP:(0
Feature Type=31
*Dec 29 14:12:31.208: //581/82E364CF818D/VTSP:(0
[state:S_SETUP_REQ_PROC, event:E_TSP_DIAL]
*Dec 29 14:12:31.208: //581/82E364CF818D/VTSP:(0
Digits=101, Tone Mode=0
*Dec 29 14:12:31.208: //581/82E364CF818D/VTSP:(0
*Dec 29 14:12:31.208: //581/82E364CF818D/VTSP:(0
Digits To Dial=101
*Dec 29 14:12:31.829: //581/82E364CF818D/VTSP:(0
*Dec 29 14:12:31.829: //581/82E364CF818D/VTSP:(0
[state:S_DS_DIALING, event:E_VTSP_DSM_DIALING_C
*Dec 29 14:12:31.829: //581/82E364CF818D/VTSP:(0
*Dec 29 14:12:31.829: //581/82E364CF818D/VTSP:(0
Digits To Dial=
*Dec 29 14:12:31.833: //581/82E364CF818D/VTSP:(0
*Dec 29 14:12:31.833: //581/82E364CF818D/VTSP:(0
*Dec 29 14:12:31.833: //581/82E364CF818D/VTSP:(0
[state:S_SETUP_REQ_PROC, event:E_TSP_PROGRESS]
The only thing out of the ordinary I see is that Digits To Dial=
is blank, but the line before it is fine. I am trying to dial xtension 101, but you have to press 6101 (6 is the access code to get out)
I also have a sh voice port summary
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
========= == ============ ===== ==== ======== ======== ==
0/0:1 03 e&m-wnk up dorm idle idle y
0/0:0 01 e&m-wnk up up seized seized y
0/0:0 02 e&m-wnk up dorm idle idle y
1/0 -- fxo-ls up up idle off-hook y
1/1 -- fxo-ls up up idle off-hook y
2/0 -- fxo-ls up dorm idle on-hook y
2/1 -- fxo-ls up dorm idle on-hook y
3/0 -- fxs-ls up dorm on-hook idle y
3/1 -- fxs-ls up dorm on-hook idle y
Both the lines are seized properly
Here is a copy of my current Cisco 1760 Config: I have enough DSP etc
clock timezone PST -8
tdm clock T1 0/0 voice export line
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 0
!
voice-card 1
!
voice-card 2
!
voice-card 3
!
no aaa new-model
ip subnet-zero
ip cef
!
!
!
!
ip name-server 10.100.0.2
ip name-server 10.100.0.3
no ftp-server write-enable
!
!
!
!
!
!
voice class h323 1
h225 timeout tcp establish 3
!
!
!
!
!
!
!
ccm-manager redundant-host 10.100.0.41
ccm-manager mgcp
!
!
controller T1 0/0
framing esf
linecode b8zs
ds0-group 0 timeslots 1-2 type e&m-wink-start
ds0-group 1 timeslots 3 type e&m-wink-start
!
controller T1 0/1
framing esf
linecode b8zs
!
!
!
interface FastEthernet0/0
ip address 10.100.0.34 255.255.255.0
speed auto
full-duplex
h323-gateway voip interface
h323-gateway voip h323-id cscso1760voipgw01@cometcom
!
ip classless
ip route 0.0.0.0 0.0.0.0 10.100.0.1
!
no ip http server
!
dialer-list 1 protocol ip permit
!
control-plane
!
!
!
voice-port 0/0:1
!
voice-port 0/0:0
!
voice-port 1/0
!
voice-port 1/1
!
voice-port 2/0
!
voice-port 2/1
!
voice-port 3/0
!
voice-port 3/1
!
mgcp
mgcp call-agent 10.100.0.40 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
no mgcp timer receive-rtcp
mgcp sdp simple
!
mgcp profile default
!
!
!
!
dial-peer voice 6001 pots
application mgcpapp
port 3/0
!
dial-peer voice 6002 pots
application mgcpapp
port 3/1
!
dial-peer voice 7000 pots
application mgcpapp
port 0/0:0
forward-digits 0
!
dial-peer voice 91 pots
application mgcpapp
port 1/1
!
dial-peer voice 90 pots
application mgcpapp
port 1/0
!
dial-peer voice 8000 voip
preference 1
destination-pattern [6,8]...
voice-class h323 1
session target ipv4:10.100.0.40
dtmf-relay h245-alphanumeric
!
dial-peer voice 7001 pots
destination-pattern 6...
incoming called-number 1..
port 0/0:1
!
gateway
timer receive-rtp 1200
I have exhausted every single resource I have, this is the only part left in our VOIP integration setup. Any feedback you can provide will be greatly appreciated.
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by: harryyehPosted on 2004-12-30 at 15:38:50ID: 12929602
Currently with this configuration below I am able to call the prime set:
Setting the MICS Line to Manual Answer Mode with ANI disabled and DNIS disabled. When I do this and I set the primeset on the line, it is able to call in. What exactly is ANI or DNIS? I know these are for Caller ID are they not?