CME required CUBE license for ISP SIP Trunk

Hi,
We have IP telephony setup with CME 9.1 with 2921 ISRG2 router.
Client would like to take ISP SIP trunk.
Do we require CUBE license for the same.Because I tried in the 2851 router without the CUBE license and its working fine.
How I can check CUBE license installed my CME.
When I check license in CME I can see that UC K9 and IPBasek9 license are permanent.
Do this enough for the SIP trunk configuration.
Thanks & Regards
Nithin Louis.

Just to add to George answer - CUBE licenses are somekind "honor" based right now which means you can configure everything and it will work but you need license paper for proving that you have adequate number of CUBE licenses for your system.
Maybe in future Cisco will improve this with option that you will need to install license and only then it will work - like many other licensing right now...
BR,
Dragan

Similar Messages

Need CUBE license for Existing 3945 GW Router ?

Hi
I already have a 3945 Router that have purchased long back
Now we are planning for a SIP TRUNK to Service provider ( PSTN).
We have UC License.
As we have not purchased CUBE license earlier with router , I would like to know
if its available to Purchase CUBE License and Session License
as spare ? if its Right to Use license..how to purchase and inegrate it with existing 3945 Router ?
==============================================================

It is a paper license. You need to contact you cisco reseller to purchase it.

View 'CUT Home Page View (CME)' is not licensed for this site.(SBL-DAT-0032

Hi Siebel guru,
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View 'CUT Home Page View (CME)' is not licensed for this site.(SBL-DAT-00327)
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Is it required separate License for Adobe Test & Target?

Hello Team,
I want to implement Adobe Test & Target for that is it requried to purchase separate License for that or it is included in CQ5 bundle?
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2901 CME: Problem with incoming call via SIP Trunk

Dear All,
I have seen some others posted similar question regarding this but mine still doesn't work by using the reference solution.
Mine is quite standard setup too -> CME setup on my 2901 router, analog phone attach to my FXS port my outgoing calls are working fine via SIP but my incoming calls are not. Caller only listen to engage tone and analog phone is not ringing at all. Attached with my config and trace log of ccsip messages. Kindly assist. Thank you so much.

Hi Carlo,
Here it is
CME_2901#show sip-ua timers
SIP UA Timer Values (millisecs unless noted)
trying 500, expires 180000, connect 500, disconnect 500
prack 500, rel1xx 500, notify 500, update 500
refer 500, register 500, info 500, options 500, hold 2880 minutes
, registrar-dns-cache 3600 seconds
tcp/udp aging 5 minutes
CME_2901#show sip-ua retry
SIP UA Retry Values
invite retry count = 6   response retry count = 6
bye retry count    = 10 cancel retry count   = 10
prack retry count = 10 update retry count    = 6
reliable 1xx count = 6   notify retry count   = 10
refer retry count = 10 register retry count = 6
info retry count   = 6   subscribe retry count = 6
options retry count = 6
CME_2901#show sip-ua min-se
SIP UA MIN-SE Value (seconds)
Min-SE: 1800

License for SIP trunk?

Hi Guys, we are going to setup a few SIP Trunks between CUCM 7.1.3 and third party FAX server. Do we need extra license for this SIP trunk? I had a look at the license calculator but it doesn't mention any SIP trunks. So I think it will not consume any license units, am I correct?
Thanks. Leo

Yes, correct, no licenses required/consumed for/by SIP Trunk.
GP.
Pls rate the post if it helps !!

Route pattern to SIP trunk problem

Hello, I have a 2801 router that has been configured with CME and a working SIP connection to my local ISP.
Tested with calls via CME so I know for sure that the SIP config and dial plan is fine on this gateway.
Next I wanted to try out CUCM so I set up a CUCM 8.6 box that is connected to the 2801 router to use as it's SIP gateway.
The only change I made to the gateway router config was to alter the "ip option 150" address so that the phones go to CUCM for their configs etc (which they do with no problems).
Then I set up a SIP trunk in CUCM along with a route pattern which is to use the SIP trunk within the Gateway/Route list option.
But when I make a call that matches this route pattern all I get is the intermittent beep message from the phone. I cannot route calls succesfully through it.
I have checked network connectivity and all is fine. The IP address I specfied in CUCM for the SIP trunk is simply one of the interfaces on the 2801 router and it is definitley reachable.
I also activated "debug ccsip all" on the 2801 gateway router but nothing appears. So it seems like the calls are not even reaching the 2801 gateway ?
Is the problem possibly a conflit between CME on the gateway router and my CUCM ?
Do I need to disable CME somehow on the gateway first ? Or am I not doing something correct in the CUCM config ?
Thank you kindly for any suggestions.
ps. I have attached a couple of screenshots of my config.

Hello, thanks for helping.
I activated "debug voice ccapi inout" as well as "debug ccsip all" on the gateway but nothing showed up.
Therefore I deduce the call is not even making it to across the SIP trunk into the gateway router ?
As I am a newbie trying this out for the first time, it is guranteed to be something really simple.
I have included my running config from the gateway router below..
One addition I made was to add an incoming dial peer. That is "dial peer 5, description CUCM SIP trunk".
I set it up with a destination patter 2... to match my phone config on CUCM which have numbering in the 2000 range.
Sorry, I got RTMT up and running but could not get any meaningful results from it. I need to learn up on that.
I did however run a 'dialed number analysis' from CUCM direct and have attached the result. It seems the dialled number "99" is matching the route pattern OK.
So why is it not then moving down the SIP trunk to my gateway and getting picked up by the incoming dial peer ?
Thanks if you guys can offer any more help.
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot system flash:c2801-ipvoicek9-mz.151-2.T0a.bin
boot-end-marker
no aaa new-model
clock timezone nzst 13 0
dot11 syslog
ip source-route
ip dhcp pool DATA_SCOPE
   network 192.168.200.0 255.255.255.0
   default-router 192.168.200.1
   dns-server 8.8.8.8
ip dhcp pool VOICE_SCOPE
   network 192.168.100.0 255.255.255.0
   default-router 192.168.100.1
   option 150 ip 192.168.2.115
ip dhcp pool MGMT_SCOPE
   network 192.168.1.0 255.255.255.0
   default-router 192.168.1.99
ip cef
ip name-server 4.2.2.2
no ipv6 cef
multilink bundle-name authenticated
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
codec preference 3 g711ulaw
codec preference 4 ilbc
voice translation-rule 1
rule 1 /^9/ //
voice translation-profile Strip9ToGetOut
translate called 1
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-2995340181
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2995340181
revocation-check none
crypto pki certificate chain TP-self-signed-2995340181
certificate self-signed 01
3082023E 308201A7 A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32393935 33343031 3831301E 170D3733 30363034 31393534
32305A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 39393533
34303138 3130819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100C34D C8ECBB53 E01373A3 2E286B78 2D23042B 1C8588B1 A7861899 BA1C6860
AE1D7868 2A59E3BC 54D0A457 8FFDE27F C09104E5 C7A429F3 74CD9DA8 4A980366
675CC27C CDB94838 821CC05F 2C0AC2BC D882C132 6CAA1FA6 6DA740E4 562428B1
12B741F1 A50C9246 4CC35EDA DEE1D038 3883BB35 A91ABF8B 483E4160 F5FA4B5A
9A570203 010001A3 66306430 0F060355 1D130101 FF040530 030101FF 30110603
551D1104 0A300882 06526F75 74657230 1F060355 1D230418 30168014 72119640
F3396E1F E4168086 D31D8619 0D8337FF 301D0603 551D0E04 16041472 119640F3
396E1FE4 168086D3 1D86190D 8337FF30 0D06092A 864886F7 0D010104 05000381
81003B5A 29DE3A1E C5AB6092 E8D90650 C80752FC 0AAC93FD C5DE3D69 071B08FA
D4013232 81CA07E7 15F90190 6A3AD6A0 1D05F0F2 13479568 888332A5 F81E2681
7DA44095 4D11CFB7 CA79579A 8D95DE54 7B00173C E2C50573 A310C8C9 1487FEFC
CE35B66E 9EF94CFA 8D6D6DCD ADC78132 2709F198 6DF2F0FA D80CC088 D0C4C7D1 080B
      quit
license udi pid CISCO2801 sn FTX0947W07M
username xxx privilege 15 password 0 xxx
interface FastEthernet0/0
ip address 192.168.3.50 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
duplex auto
speed auto
interface FastEthernet0/1.2
encapsulation dot1Q 2
ip address 192.168.2.1 255.255.255.0
interface FastEthernet0/1.99
encapsulation dot1Q 99
ip address 192.168.1.99 255.255.255.0
interface FastEthernet0/1.100
description voice_VLAN
encapsulation dot1Q 100
ip address 192.168.100.1 255.255.255.0
interface FastEthernet0/1.200
description data_VLAN
encapsulation dot1Q 200
ip address 192.168.200.1 255.255.255.0
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip route 0.0.0.0 0.0.0.0 192.168.3.1
logging esm config
tftp-server flash:/phone/7940-7960/P00307020200.bin alias P00307020200.bin
tftp-server flash:/phone/7940-7960/P00307020200.loads alias P00307020200.loads
tftp-server flash:/phone/7940-7960/P00307020200.sb2 alias P00307020200.sb2
tftp-server flash:/phone/7940-7960/P00307020200.sbn alias P00307020200.sbn
control-plane
mgcp fax t38 ecm
dial-peer voice 1 voip
description local_7_Digit_Calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 9[2-9]......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 2 voip
description international_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 900T
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 3 voip
description national_calling
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
session protocol sipv2
session target ipv4:203.184.16.2
voice-class codec 1
dial-peer voice 4 voip
translation-profile outgoing Strip9ToGetOut
destination-pattern 90[34679].......
dial-peer voice 5 voip
description CUCM SIP trunk
destination-pattern 2...
session protocol sipv2
session target ipv4:192.168.2.115
voice-class codec 1
sip-ua
authentication username xxxxxxxxxx password xxxxxxxx
060
telephony-service
max-ephones 10
max-dn 20
ip source-address 192.168.1.99 port 2000
load 7960-7940 P00307020200
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 1000
name Lydia Francis
ephone-dn 2 dual-line
number 1001
name Leah Francis
ephone-dn 3 dual-line
number 1002
n
ephone-dn 4 dual-line
number 1003
ephone 1
mac-address C80A.A970.01DE
type CIPC
button 2:2
ephone 2
mac-address 000C.3070.8705
button 1:1 2:15
ephone 3
mac-address 000C.8546.5954
button 1:3 2:15
line con 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
scheduler allocate 20000 1000
ntp server 195.43.74.123
end

Best Practice to Integrate CER with RedSky E911 Anywhere via SIP Trunk

We are trying to integrate CER 9 with RedSky for V911 using a SIP trunk and need assistance with best practice and configuration. There is very little documentation regarding "best practice" for routing these calls to RedSky. This trunk will be handling the majority of our geographically dispersed company's 911 calls.
My question is: should we use an IPsec tunnel for this? The only reference I found was this: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/virtual-office/deployment_guide_c07-636876.htmlm which recommends an IPsec tunnel for the SIP trunk to Intrado. I would think there are issues with an unsecure SIP trunk for 911 calls. Looking for advice or specifics on how to configure this. Does the SIP trunk require a CUBE or is a CUBE only required for the IPsec tunnel?
Any insight is appreciated.
Thank you.

you can use Session Trace in RTMT to check who is disconnecting the call and why.

Confused by basic SIP Trunk configuration.

I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing. I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
If someone can make some sense of this, I'd truly appreciate it!

Hello Aditya and thanks for the consideration!
I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production.
I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
==============================================
Branch_SIP#show run
Building configuration...
Current configuration : 3529 bytes
! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Branch_SIP
boot-start-marker
boot-end-marker
! card type command needed for slot/vwic-slot 0/2
enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 10.0.10.1 10.0.10.10
ip dhcp excluded-address 10.0.30.1 10.0.30.10
ip dhcp pool Data
network 10.0.10.0 255.255.255.0
default-router 10.0.10.254
option 150 ip 192.168.5.250
dns-server 192.168.5.200
ip dhcp pool Voice
network 10.0.30.0 255.255.255.0
default-router 10.0.30.254
dns-server 192.168.5.200
option 150 ip 172.16.0.1
ip dhcp pool data
option 150 ip 172.16.0.2
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
bind media source-interface Loopback1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2851 sn FTX1031A2FM
redundancy
interface Loopback1
ip address 2.2.2.2 255.255.255.255
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 10.0.10.254 255.255.255.0
interface GigabitEthernet0/0.30
encapsulation dot1Q 30
ip address 10.0.30.254 255.255.255.0
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Serial0/3/0
no ip address
shutdown
clock rate 2000000
interface Serial0/3/1
ip address 172.16.0.1 255.255.255.0
clock rate 250000
interface Internal-Service-Module0/0
no ip address
shutdown
!Application: CUE Running on AIM2
hold-queue 512 out
router eigrp 1
network 0.0.0.0
network 2.2.2.2 0.0.0.0
network 10.0.0.0
network 10.0.10.0 0.0.0.255
network 10.0.30.0 0.0.0.255
network 172.16.0.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.16.0.2
tftp-server flash:term45.default.loads
tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
tftp-server flash:cnu45.8-5-3TH1-6.sbn
tftp-server flash:apps45.8-5-3TH1-6.sbn
tftp-server flash:dsp45.8-5-3TH1-6.sbn
tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
control-plane
voice-port 0/0/0
voice-port 0/0/1
mgcp profile default
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
session protocol sipv2
session target sip-server
codec g711ulaw
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 5...
session protocol sipv2
session target sip-server
codec g711ulaw
sip-ua
sip-server ipv4:192.168.5.250
telephony-service
codec g711ulaw
max-ephones 24
max-dn 48
ip source-address 172.16.0.1 port 2000
system message SIP Branch Site
cnf-file location flash:
load 7960-7940 P00308010200.bin
max-conferences 8 gain -6
transfer-system full-consult
ephone-dn 1
number 4008
ephone-dn 2
number 4005
ephone 1
device-security-mode none
mac-address 001D.A21A.2065
button 1:1
line con 0
exec-timeout 0 0
line aux 0
line 194
no activation-character
no exec
transport preferred none
transport input all
transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
speed 115200
line vty 0 4
password cisco
login
transport input all
line vty 5 15
password cisco
login
transport input all
scheduler allocate 20000 1000
end
Branch_SIP#show debug
TFTP:
TFTP Event debugging is on
CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)
Branch_SIP#
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B6C5C28
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 4008
Called Number            : 5005
Source IP Address (Sig ): 172.16.0.1
Destn SIP Req Addr:Port : 192.168.5.250:5060
Destn SIP Resp Addr:Port : 192.168.5.250:5060
Destination Name         : 192.168.5.250
*Apr 2 03:20:32.351: //25/0C496935804A/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): 2.2.2.2
Source IP Port    (Media): 19472
Destn IP Address (Media): -
Destn IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Apr 2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503
Branch_SIP#

Design Question Sip Trunk

Hi,
I have a requirement to move from a h323 environment to a SIP environment. I am looking for best practises especially around security. I have 2 CUCM servers (8.5) located in separate cities for redundancy. I also have 2 voice gateways which at the moment are h323 to the PSTN, each located at different cities.
My requirements are:
1. Creat a sip trunk to the provider instaed of using PRI.
2. If the Wan link fails on one gateway to provider, the alternate router in the other location should be able to receive the setup messages and if a user logs on via extension mobility, should be able to answer the call.
Are there any simplified design docos about for this? I am hesitant to create a SIP trunk straight to the provider for security, so thinking of terminating the call on the voice routers with CUBE. I'm pretty sure this is run of the mill and would appreciate some input.
Cheers!
Pieter

+5 to Chris..Always use CUBE...
Here are more ideas..
1. Create two sip trunks, first one to Cube 1, second to CUBE 2
on your sip trunk set DTMF as no preference
2. Assign the trunks to CUCM group with your two servers in it
3. Configure route groups with Circular algorithm distribution (this way you have load balancing on your two cubes)
4. Configure dial-peers on your CUBE gateway to point in preferential order to your cucm servers
5. Use dtmf-relay rtp-nte on your dial-peer (ensure you have sip as the protocol on your dial-peers)
6. configure your codec selection properly on your dial-peers
7. configure your region settings properly between your sip trunk and phones.
8. Evaluate if you need xcoders, provide one if you do and ensure you set your region correctly between your xcoder and CUBE
9. Is there voicemail involved? Ensure you set your region settings between cube and voicemail correctly, otherwise calls to voicemail may invoke xcoder (from experience)
10. Provision adequate bandwidth for your calls. Once you move to sip, you loose the luxury of e1 channels. You are solely relying on Bandwidth. Ensure you have adequate bandwidth for your concurrent calls
11. Provision QoS for your calls
12. Is there fax involved? You need to think carefully on this one. What fax method do you want to use. T.38/pass through. Does your provider support T.38?
13. Have a thorough test plan. Test call transfers, call on hold, call forward etc
Just a few pointers..Careful planning and implementation is required for a successful SIP implementation

+ Globalization and SIP Trunks

I'm trying to add the +E.164 number to Missed Call Field. On a SIP Trunk, I know I don't have support for ISDN TON (Subscriber, National, International), so when I push the Calling Party Number to a Transformation how do I ensure I tag Local, National, and International Calling Party Number correctly? Configuration examples for those who have done this would be appreciated.

Hello Sam,
After a lot of research on my personal Lab, I was able to achieve Globalization in SIP Trunks. This is how I did:
I created two partitions, CALLING-XFORM-DP-PT, and CALLING-XFORM-TRK-PT.
One is for the Device Pool, and the other one is for the SIP Trunk. Create the CSSs and add your partitions > CALLING-XFORM-DP-CSS with CALLING-XFORM-DP-PT inside and CALLING-XFORM-DP-CSS with CALLING-XFORM-TRK-PT inside.
Now create two Device Pools, one for the Trunk and one for your Phones.
PHONES-DP and TRUNK-DP were the Device Pools I created. I am trying to be as specific as possible so everybody understands, so bear with me. :)
Now, put the CSSs accordingly. Go to PHONES-DP "Device Mobility Related Information" field and put CALLING-XFORM-DP-CSS there. Save and reset.
On TRUNK-DP you are not require to do anything, of course there is an "unless", but let's stick with your main concern, otherwise this post will be bigger than it will be.
Go to the Trunk itself and add the CALLING-XFORM-TRK-CSS to the "Incoming Calling Party Settings" field. Uncheck the "Use Device Pool CSS" and leave "Strip Digits" and "Prefix" fields in its defaults. Save and Reset.
Now, create your Calling Party Transformation Patterns. I will explain to you using the Brazilian national standard.
For Local Calls, here in Sao Paulo, we receive 11 which is the area code and 8 or 9 digits after that for local number. So, I created the Calling Party Xform mask as follows:
1 - 11.9XXXXXXXX > discard digit instruction: Predot > Prefix +5511
2 - 11.[2-9]XXXXXXX > discard digit instruction: Predot > Prefix +5511
I didn't change the TON or the Plan because it doesn't matter. SIP doesn't recognize these fields.
Now, same thing for national calls. We receive the area code and 8 or 9 digits. Since we already have specific Patterns for Local calls, it is ok to have a more generic pattern for other area codes
1 - [1-9][1-9]9XXXXXXXX > No Discard Digit Instruction > Prefix +55
For international calls, our SP always sends 00 in front of the number. Knowing that, it was easy to simply add the following pattern.
1 - 00.! > discard digit instruction: Predot > Prefix +
Now, you create the Calling Party Transformation Patterns for Phone Device Pools. Achieving Localization as well.
Local Numbers
1 - \+5511.9XXXXXXXX > discard digit instruction: Predot > No need to prefix anything
2 - \+5511.[2-9]XXXXXXX > discard digit instruction: Predot > No need to prefix anything
National Numbers
3 - \+55.[1-9][1-9]9XXXXXXXX > discard digit instruction: Predot > Prefix 0
4 - \+55.[1-9][1-9][2-9]XXXXXXX > discard digit instruction: Predot > Prefix 0
(in Brazil we used to have the numbers delivered to us as 0 + area code + number. That Prefix is just to achieve localization for us)
There is no need to do anything for International Numbers at this level, because if the number is different from +55, there is nothing to be done, just receive the +E164 format.
After all that you will be able to even do Plus Dialing. If you need any help setting it up, please let me know.
Don't forget that if your provider sends you the digits in a way that you can't do much on CUCM, try to identify the numbers by adding a translation pattern in your Trunk (if it is Cisco, of course) before the numbers enter on CUCM. Then use the Calling Party Transformation Patterns to strip off any identification you gave for the inbound numbers.
I hope this helps.
Please rate useful posts.

MGCP FXS ports requires a license in CUCM9

Hello!
I am connecting Some analoge phones to VG350 FXS ports which is configured as a MGCP Gateway in CUCM. I beleive MGCP did not requires any license for it. Can some confirm this ? is there any Cisco doc on it ?
Thanks & Regards,

Hi Sambit,
Technically u are correct but legally I think u would be requiring license.
even the ordering guide says Analog devices are supported with Essential USer license and must be purchased through UCL.
http://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/finalcopy.pdf
regds,
aman

Third Party Phone over SIP Trunk with CUCM 9.x

Hi all,
I have a problem where my Third Party SIP phones wont go over the SIP trunk configured in my CUCM 9.x cluster. My Cisco phones work fine and goes out the trunk. I have noticed a distinct difference in wireshark with the invite packets from Third Party SIP phones and the Cisco ones.
I have configured the SIP trunk in CUCM with the following route pattern (60.!#)and configured it with associated group and list. Heres the differense between the invite packets from Cisco and Third Party phones.
Cisco Phone: INVITE sip.60xxxx%23@ipadress
Third Party SIP Phone: INVITE sip:[email protected]
It seems the Cisco phones gets some extra configured the Third Party ones dont...
Thanks in advance for any help.
//Per

Thanks for the answer
Yeah i have DNS configured and i have the trunk pointed to a domain destination SRV record and like i said it works fine when calling from a Cisco phone. I tried changing the domain to an ip address but same result. I also changed the Plycom phone from being registered towards the domain of CUCM to an IP adress of CUCM and then the SIP INVITE messages in wireshark began to look kinda the same expet for the "%23" section but it still dont work.
When i look at the Real Time Data in RTMT the orig and final called from the cisco phone has stripped the 60 and forwared the rest of the number towards the correct domain for the SIP trunk.
When looking at the data from the Polycom phone the orig and final called data still contains the 60 prefix part and the called device name field is empty. The termination Cause Code is that the number requested is Unallocated/Unassigned..
In other words something is missing to get CUCM to strip 60 from the Polycom phones dialed number and send it towards the SIP trunk like it does when the Cisco phones call it.
Unfortunatley i dont have the meens to attach the trace...
Thanks again for any help/advice
With regards, Per.

Forwarding with SIP Trunking and Retaining CID

We are using SIP trunking to get to the PSTN and CUCM 7.1. The SIP provider only permits calls originating from their own DIDs.
So how can we allow our users to forward all calls to say their cell phone. AND when a call comes in we want them to be able to see the original caller ID. Is it possible? What is the mojo? Thanks!

Hi,
for the SIP trunks there are no limits from the system. Check this out: https://supportforums.cisco.com/message/3795863#3795863
If you have different voice codecs for your phones and the trunk you need DSP ressources for transcoding.
As I know the UC520 have a PVDM2-64 with 4 DSP chips. You can use the DSP calculator from cisco to find out how many DSPs you need. But keep in mind that conferencing and transcoding can't share a DSP processor.
For example 1 DSP for conferencing and 3 for transcoding.
best regards
Christian

License for ECC Sandbox

Hello Experts,
We want to create a sandbox for SAP WM ECC 6.0 system. We have the licenses for dev/qa/prod.
Do we require separate license for sandbox?
Thanks & best regards,
- Praveen Soni

Hi,
SAP* should be able to login.
1. Delete the SAP* user from client 000.
2. set the below parameter in the instance profile of your system
login/no_automatic_user_sapstar=0
this needs to be done from the OS. Bounce your system for the parameter to take effect. Login with SAP*/pass
This should work.
Do let me know if this helps.
Mohan.

CME required CUBE license for ISP SIP Trunk

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