CME to CUCM

[Robert Craig]

For the love of me, I cannot get my CME to dial out to a CUCM, or even full version CCM. I know it's rather simple. I created a dial-peer that looks something like this:

dial-peer voice 25 voip

destination pattern 1...  <<---This accounts for any phones that start with 1 and have 4 digits total

session target ipv4:192.168.1.252   <<<----IP Address of my CCM or CUCM

codec g711ulaw

no vad

I have created a route pattern on both the CCM or CUCM and can get calls from either to CME, but not the other way around. As soon as I finish the 4th number of an extension for either system (i have switched between both systems thinking it was some config error on either one), I get the fast busy signal. I simply don't get it. Anyone have any ideas? Thanks in advance for any help.

Rob

[Amer rajai Sha'er · Accepted solution]

Hello Rob,

Let me get this straight , you are able to dial from the CUCM to CCME but no the other way aroudn right ?

First , r u using a trunk on the CUCM to connect between CCME and CUCM ? if yes did you add a CSS on the inbound calls CSS on the trunk that has access to the ip phones you are dialing to?

second , if the above is ok , can you please do a debug voice ccapi inout and upload the debug along with the sh run of the CCME ?

Amer

[Robert Craig]

OK. Here is setup.

CCME ----- Phone with extension 5000

CUCM (Version 4.3.2) ---- Phone with extension 1000

1000 Can call 5000 by a simple route pattern and H323 Gateway that points to the CCME. There are no CSS because this is a fresh install with minimal config. No CSS or partitions have been created.

5000 Can NOT call 1000. Now quite sure how to display results of the debug (I've never used it before). Below is the config for my CCME.

no aaa new-model
voice-card 2
!
voice-card 3
!
ip cef
!
!
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.3.1 192.168.3.25
!
ip dhcp pool ITS
   network 192.168.3.0 255.255.255.0
   option 150 ip 192.168.3.1
   default-router 192.168.3.1
   dns-server 8.8.8.8
!
!
ip name-server 8.8.4.4
ip name-server 8.8.8.8
multilink bundle-name authenticated
!
!
!
voice service pots
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
sip
!
!
!
archive
log config
  hidekeys
!
!
interface FastEthernet0/0
no ip address
speed auto
!
interface FastEthernet0/0.1
encapsulation dot1Q 1 native
ip address 192.168.1.254 255.255.255.0

interface FastEthernet0/0.2
encapsulation dot1Q 2
ip address 192.168.2.1 255.255.255.0
!
interface FastEthernet0/0.3
description VOIP Network
encapsulation dot1Q 3
ip address 192.168.3.1 255.255.255.0
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.1.1
!
ip http server
no ip http secure-server
ip http path flash:/gui
!
tftp-server flash:SCCP41.8-2-2SR1S.loads
tftp-server flash:apps41.8-2-2es1.sbn
tftp-server flash:cnu41.8-2-2es1.sbn
tftp-server flash:cvm41sccp.8-2-2es1.sbn
tftp-server flash:dsp41.8-2-2es1.sbn
tftp-server flash:jar41sccp.8-2-2es1.sbn
tftp-server flash:term41.default.loads
tftp-server flash:term61.default.loads
tftp-server flash:/7970-7971/apps70.8-2-2es1.sbn alias apps70.8-2-2es1.sbn
tftp-server flash:/7970-7971/cnu70.8-2-2es1.sbn alias cnu70.8-2-2es1.sbn
tftp-server flash:/7970-7971/cvm70sccp.8-2-2es1.sbn alias cvm70sccp.8-2-2es1.sbn
tftp-server flash:/7970-7971/dsp70.8-2-2es1.sbn alias dsp70.8-2-2es1.sbn
tftp-server flash:/7970-7971/jar70sccp.8-2-2es1.sbn alias jar70sccp.8-2-2es1.sbn
tftp-server flash:/7970-7971/sccp70.8-2-2sr1s.loads alias sccp70.8-2-2sr1s.loads
tftp-server flash:/7970-7971/term70.default.loads alias term70.default.loads
tftp-server flash:/7970-7971/term71.default.loads alias term71.default.loads
tftp-server flash:desktops/320x212x12/list.xml
tftp-server flash:desktops/320x212x12/campusnight.png
tftp-server flash:desktops/320x212x12/ciscofountain.png
tftp-server flash:desktops/320x212x12/mountain.png
tftp-server flash:desktops/320x212x12/fountain.png
tftp-server flash:desktops/320x212x12/morrorock.png
!
control-plane
!
!
!
voice-port 2/0
caller-id enable
!
voice-port 2/1
!
voice-port 3/0
description Line to OOMA
caller-id enable
!
voice-port 3/1
!
!
!
sccp
!
!
!
dial-peer voice 10 voip
description *** 10 Digit Calls ***
destination-pattern [2-9]..[2-9]......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 100 voip
description *** Incoming Dial-Peer ***
session protocol sipv2
session target sip-server
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 11 voip
description *** 11 Digit Calls ***
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 911 voip
destination-pattern 911
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 5 voip
destination-pattern 1000
session target ipv4:192.168.1.252
codec g711ulaw
no vad
!
gateway
timer receive-rtp 1200
!
sip-ua
credentials username XXXXX password XXXXXXXX realm sip-ua.com
authentication username XXXXXXX password XXXXXXXXX
registrar dns:proxy.sip-ua.com expires 60
sip-server dns:proxy.sip-ua.com
!
!
telephony-service
load 7941GE SCCP41.8-2-2SR1S
load 7970 sccp70.8-2-2sr1s
max-ephones 5
max-dn 5
ip source-address 192.168.3.1 port 2000
auto assign 1 to 5
service phone videoCapability 1
timeouts interdigit 4
timeouts busy 25
system message Welcome home....
url services http://phone-xml.berbee.com/menu.xml
time-zone 7
voicemail 9999
max-conferences 4 gain -6
moh music-on-hold.au
web admin system name XXXXXXXX password XXXXXXXX
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Jun 04 2011 15:07:19
!
!
ephone-dn  1  dual-line
number 5000 no-reg both
label Desk-Phone
name Desk-Phone
call-forward busy 9999
call-forward noan 9999 timeout 45
!
!
ephone-dn  2  dual-line
number 5001 no-reg both
label House Line 2
call-forward busy 9999
call-forward noan 9999 timeout 45
!
!
ephone-dn  3  dual-line
number 5002 no-reg both
label Rob-Laptop
name Rob-Laptop
call-forward busy 9999
call-forward noan 9999 timeout 45
!
!
ephone-dn  4  dual-line
number 5202269505 no-reg both
label House Phone
name House Phone
call-forward busy 9999
call-forward noan 17605838270 timeout 60
!
!
ephone-dn  5  dual-line
number 5004 no-reg both
call-forward busy 9999
call-forward noan 9999 timeout 5
!
!
ephone  1
video
mac-address 0019.2FEE.A4C8
button  1:1 2:4 3:3
!
!
!
ephone  2
video
mac-address 0250.F200.0001
type CIPC
button  1:3
!
!
!
ephone  3
video
mac-address 000E.8349.D3F5
type ata
button  1:4
!
!
!
ephone  4
video
mac-address 0E83.49D3.F501
type ata
button  1:5
!
!
!
ephone  5
video
mac-address B8AC.6F79.3677
type CIPC
button  1:2
!
ntp clock-period 17208163
ntp server 209.81.9.7
end

[Amer rajai Sha'er]

Ok robert,

Can you please add these to the gateway and give it a try:

dial-peer voice 120 pots

incoming called-number .

direct-inward-dial

Amer

[Robert Craig]

Just out of curiosity, how would that dial-peer apply in my situation? I'm not using any pots phones at the moment.

[Amer rajai Sha'er]

Hello,

Always the call must be matched with two dial-peer (one incoming and one outgoing) , usualy when it's CME there should be a system pots dial-peer for the ip phone itself , i asked you to try this since we don't have d debug to check that , the debug voice ccapi inout can show us exactly what is happeneing to the call from the time it start until the end.

To enable the debug it is easy , use any telnet software so you can capture the lines , on the router term level (which is the #) type : debug voice ccapi inout , then press enter then type terminal monitor and do a call , the debug will appear and we can have a better idea for what is happening .

I reliazed something else , if you are using the gateway (CME router) as a H323 gateway on CUCM , you should add to commands to the ethernet interface which they are :

h323-gateway voip interface

h323-gateway voip bind srcadd XXX.XXX.XXX.XXX (this is the ip address of the interface itself)

Please try all the commands i gave you and then enable the debug and give it a try.

Amer

[Robert Craig]

Amer,

     I tried just the POTS dial peer with no success. I then added the statement to the VOIP interface and it worked! I took the pots dial-peer back out and it still works. Thanks for your help. Now, for future reference, what would be the difference between an H323 gateway pointing to the CCME or a trunk?

[Amer rajai Sha'er · Accepted solution]

Hello Robert,

If you can check my latest reply , i said that in case of CCME it is different before every e-phone-dn when it is created it will get a dial-peer pots , so you should be ok , in case the gateway was not a CME this won't work without the dial-peer pots.

In your case there is no much difference from H323-gateway to H323 trunk , but i am guessing you went to the h-323 gateway becasue you wil gonna use the CCME gateway as a gateway for the CUCM users .

You can try it , try to make a new H323 non gatekeeper controlled and point it to the gateway , then associate the route pattern with the trunk and give it a shot with the same config , there will be no difference.

As for future , don't forget to add three things to the gateway (in case of H323 gateway)

1- H323-voip interface (same config i gave you before) this so that the gateway can know the h323 gateway to comunicate with.

2- VOIP dial-peers , create a voip dial-peer with a session target ipv4 for every call manager server you have even if there is no destination-patern , if there will be destination-patern , put the incoming called-number . comand into the voip dial-peer.

3- for evey PSTN port (BRI , PRI, FXO , etc....) create a Pots dial-peer with direc-inward-dial command without a destination pattern so it can match in case of incoming call so it can be matched tothe VOIP dial-peer and continue the call.

Hope this helps.

Amer

[Robert Craig]

I will remember that. Thanks again!