E164 dialling through Cisco Call Manager and its Gate Keeper

Hello,
one of our customers has succesfully setup a few Polycom VSX Endpoints and they are registered to a Gatekeeper.
Calls can be made manually in both directions but if a IP Communicator/IP Phone calls a VSX the recent call list will only include the IP Address of the Gatekeeper and not the E.164 Number so calls cannot be returned.
Is there a way to configure this ?
I am unaware of the Version of the CCM.
Regards
Steff

configuration has two gatekeeper zones. An internal gatekeeper
zone and an external gatekeeper zone.
Devices registered to the internal zone can call devices on the external
Calls through H.323 address but not by e164 alias. However the devices on external zone
Can call devices on internal zone through e164 alias.
Call Manager is not involved in this problem.

Similar Messages

Cisco Call Manager and LifeSize Endpoints

Hello dear support community,
last week I asked which version of Cisco Call Manager supports BFCP (https://supportforums.cisco.com/message/3966334).
Unfortunately I asked the wrong question.
As it turns out I just didn't need to know which version of Call Manager supports BFCP, which I know now, I also need to know if Call Manager is compatible with LifeSize systems?
Here is the setting, so you know what I'm dealing with:
The costumer uses Cisco Call Manager (Ver. 9.1) as SIP registrar for his LifeSize endpoints. The problem is, when he is making a video conference (via SIP) and he wants to share a presentation, the other side either sees him or the presentation, but not both.
We tried it with our Call Manager (Ver. 9.0) and some LifeSize endpoints. As soon as the LifeSize endpoint uses CUCM as SIP registrar, the option to share the presentation is completely gone.
I guess the costumer switched sources from cam to pc. That would explain why he just sees either video or presentation. But, as you might have guessed, it's not an acceptable solution for the costumer.
And yes, we made sure that BFCP is enabled for the endpoints.
So, what I'm asking myself and you is:
Is this known? And more important, is there a reliable workaround?
Thanks a lot in advance.
Best regards
Tobias

I've been down this road and when I asked the question Lifesize's stance was that it wasn't supported by THEM. I pointed out that CUCM now supports line-side BFCP but they essentially shrugged their shoulders. Either they are blatently disabling content sharing by virtue of using a SIP Proxy or they have a priorietary SDP/header that CUCM isn't passing through.
Either you can call Lifesize and shake them up a bit or you can wireshark the SIP dialog to see what the codec asks CUCM before disabling the sharing button. Once you know that you might be able to use SIP transparancy to pass the header through.

Cisco Call Manager and MGCP Question

Hello,
I appreciate if somebody can help.
Scenario:
Site 1 PSTN E1----VG----Call Manager----VG--- PABX---Site 2 PSTN1 E1
I have configured a dialing pattern on Cisco call manager 6.xxxxxx to Send to VG on Site1
Both VG routers are using MGCP with call Manager.
The problem if from Site 2 tries to call 6xxxxxx the call manager is not routing the call to the VG in site 1.
I did debug ccapi inout and on Site 2 VG the call response was the number unassigned. This means that the call Manager is searching the directory for the destination but it is not searching the route patterns.
Any ideas to override this and ask the call manager to check it's destination pattern?
Thanks,

Problem solved. The VG in Site 2 was in a CSS that is not allowed to dial PSTN.
Regards,

Cisco call manager and ip phone software.

Hi everybody.
Does Cisco call manager also include software required for ip phone? Or software for ip phone needs to be installed on tftp server and it does not come with cisco call manager.?
thanks and have a great weekend.

Hi Sarah,
by ip phone software, do you mean Cisco IP Phone Agent software?
if yes, then you need to have Customer Response Solution (CRS) and Call Manager together to setup ip phone software (services).
check this link for further info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a00801c5765.shtml
plz Rate if it helped.
Soroush.

Nortel Call CS 1000 / Cisco Call Manager

I know kind of a oxymoron, but currently tenants are to be only on Nortel. Now no other phone and this one in particular is having problems. But one is given to the manager to have Voicemail. But because nobody (including me) has any knowledge of Nortel except Cisco CUCM, a outsource tech directed the phone to be sent to the Call Manager(Cisco) as the rest of the city employees is on Cisco. Now it is not working and instead its going to 5117 when he goes to voicemail and not 1900 which is through Cisco Call Manager.
All I want is to get this Nortel phone and have it sent to the Call Manager (Ext 1900). I am using the web GUI so how do I get this done?
This topic first appeared in the Spiceworks Community

Hi!
Just for curiosity; why wouldn´t you install COP addon?

Cisco call manager express : desactivate silencious mode in Ip telephony

Hello All,
I configured cisco call manager and it works well. I would like now not to allow users to make their phone in silencious mode by using down volum. May I do it with cisco commands lines? Thanks in advance for any help.
PS : I have a Cisco IOS Software, C2951 Software (C2951-UNIVERSALK9-M), Version 15.2(3)T1, REL
EASE SOFTWARE (fc1)
Regards.

Hi Carl Messer ,
just one doubt, in fast track method can we give any phone model with any protocol(sccp,sip) type as a reference?
for example for 8831 sip phone reference phone type given is 7937 which is a SCCP phone so my doubt if i gave 7975 sccp will it work?
from above thread:
voice register pool-type 8831
xml-config maxNumCalls 2
xml-config busyTrigger 1
xml-config custom sshAccess 0
num-lines 2
description Cisco IP Phone 8831
reference-pooltype 7937
voice register pool 1
busy-trigger-per-button 1
id mac ####.####.####
type 8831
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
description 233
codec g711ulaw
no vad
Rgds
Vigeesh

Cisco Call Manager MIB and OID.

Hi,
One of our customer has developed their own NMS software to manage multiple OEM products.
They want to integrate Cisco Call Manager 7.0 and Cisco Router with the NMS software for monitoring/managing of Voice traffic.
The monitoring of CUCM 7.0 will consists of the following-
1. Call History
2. Missed calls.
3. Calls in progress,
4, Dropped Calls.
The Monitoring of Cisco Router will Consists of Interface statistics and Portwise Traffic statistics of LAN and WAN Ports.
I believe this monitoring is done through Call Detailed Record and if I will load the CDR MIBs in NMS software it will be able to log CRD logs from CUCM 7.0.
Please suggest that if the same is possible.??.
kindly help me in getting the MIBs/OIDs for the same.
Best regards,
Puneet K.

Hi,
For 200+ users you should better go with Cisco CallManager (CUCM). Although ISR G2 supports 250+ users, it will not provide clustering mechanism and additional HA capabilities as CUCM does.
You can read the Call Processing chapter of the CUCM SRND:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/9x/callpros.html
Hope to help, please remember to rate helpful posts.
Kind regards,
- Adrian.

Cisco Call manager 7.2 through ASA firewall

Hi,
We have a part of our building that we have sold to another company. We still have to provide them with some resources until they can install their own network. We have a 6500 switch there and we are going to implement a ASA in between and lock down most communication. One of the resources required are Cisco IP phones.
Does anyone know which ports etc are required to be opened to allow communication between these phones and Call manager and other IP phones on the site?
Any help would be appreciated

Hi Andrew
The attached document may assist:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/7_0/CCM_7.0PortList.pdf
A lot depends on topology etc, and the handset registration protocol you are using (SIP vs SCCP).
Hope this helps.
Barry Hesk
Intrinsic Network Solutions

Cisco Call Manager 8.5 and Lync 2013

Dear all,
Our customer wants to integrate their Lync 2013 to the Cisco Call Manager 8.5.
Out of some not so recent information i thought that if u want to use the enterprise voice of lync you'll have to make a sip trunk between Cisco Call Manager & Lync 2013.
I think that this still counts, i've also heard that there have been a moderate amount of changes between cucm 8.5 and cucm 8.6 on matters of sip.
I'm unable to find them though, also what does CUCI - Lync do?
Kr,
Yannick Vranckx

Hi Yannick,
There can be three types of integrations between Lync and Cisco Systems
1) Direct SIP Trunk between Lync Server and Call Manager - In this, we can setup shared line between Cisco extensions and Lync extensions, and Sip trunk between the servers. Any Cisco phone / Lync client can call each other vice-versa. Initial setup needs planning, but later it does not need much configurations in client side. The following document explains the configurations needed.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucilync/9/CUCI_BK_C0B36AC1_00_cisco-uc-integration-for-microsoft.pdf
2) CUCILYNC, which is like Cisco Jabber client but connects with Lync and Call Manager: Lync Client <----> CUCILYNC <-----> CUCM
In this option, a plug-in is installed in each client side. Although initial configurations is easier, each user needs to install the client and operate it. The below document explains the configuration and setup.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucilync/9/CUCI_BK_C0545A41_00_cisco-uc-integration-for-microsoft.html
3) RCC (Remote Call Plugin) which just provides basic call control feature but for that you need Cisco Unified Presence Server: Lync Client <----> CUPS <-----> CUCM
HTH
Manish

Cisco Call Manager 5.1 and Cisco VoIP Gateways

Hi,
I have Cisco Call Manager 5.1 with Cisco 2800 series routers as a ISDN gateways and Cisco VG224/ATA 186 as POTS gateways. While Cisco Call Manager collects call statistics from 7900 series SCCP handsets managed by the Call Manager (number of VoIP packets, dropped frames etc.), it shows only one side of the call as it does not collect statistics from the Cisco routers (configured as H.323 gateways), the Cisco VG224 (with ports configured as H.323 gateways and as a SCCP devices) and Cisco ATA 186 (configured for SCCP).
Is the collection of these statistics possible? What needs to be done?
Thanks,
Paul

Update: Logging of QoS/Call Management Records (CMR) to Call Manager CDR database is not possible with H.323 gateways; only MGCP gateways:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a008020650a.shtml#qa4
Anyone know how to get the Cisco router gateway to log these statistics anyway using standard router logging mechanisms (e.g. syslog)?
Anyone know about calls to SCCP ports on Cisco VG224 or ATA 186?
Thanks,
Paul

Connectivity Issue between ASA 5520 firewall and Cisco Call Manager

Recently i have installed ASA 5520 firewall, Below is the detail for my network
ASA 5520 inside ip: 10.12.10.2/24
Cisco Switch 3560 IP: 10.12.10.1/24 for Data and 10.12.110.2/24 for Voice
Cisco Call Manager 3825 IP: 10.12.110.2/24
The users and the IP phone are getting IP from the DHCP server which configured on cisco 3560 Switch.
the Default Gateway for Data user is 10.12.10.2/24 and
for the voice users is 10.12.110.2/24
now the problem is that the users is not able to ping 10.12.110.2 call manager. please if somebody can help in this regard. i will appreciate the prompt response against this issues.

Actually i don't wana to insert new subnet and complicate the nework. i need a simple way to solve the problem. below is the details for the asa 5520 config.
ASA Version 8.2(1)
name x.x.x.x Mobily
interface GigabitEthernet0/0
nameif inside
security-level 99
ip address 10.12.10.2 255.255.255.0
interface GigabitEthernet0/1
nameif outside
security-level 0
ip address x.x.x.x 255.255.255.252
object-group service DM_INLINE_SERVICE_1
service-object tcp-udp
service-object ip
service-object icmp
service-object udp
service-object tcp eq ftp
service-object tcp eq www
service-object tcp eq https
service-object tcp eq ssh
service-object tcp eq telnet
access-list RA_VPN_splitTunnelAcl_1 standard permit Inside-Network 255.255.255.0
access-list RA_VPN_splitTunnelAcl standard permit Inside-Network 255.255.255.0
access-list inside_nat0_outbound extended permit ip Inside-Network 255.255.255.0 10.12.10.16 255.255.255.240
access-list inside_nat0_outbound extended permit object-group DM_INLINE_SERVICE_1 10.12.10.16 255.255.255.240 Inside-Network 255.255.255.0
access-list inside_nat0_outbound_1 extended permit ip Inside-Network 255.255.255.0 10.12.10.16 255.255.255.240
pager lines 24
logging enable
logging asdm informational
mtu inside 1500
mtu outside 1500
mtu mgmt 1500
ip local pool VPN-Pool 172.16.1.1-172.16.1.30 mask 255.255.255.0
ip local pool VPN-Users 10.12.10.21-10.12.10.30 mask 255.255.255.0
no failover
icmp unreachable rate-limit 1 burst-size 1
asdm image disk0:/asdm-641.bin
asdm history enable
arp timeout 14400
global (inside) 2 interface
global (outside) 1 interface
nat (inside) 0 access-list inside_nat0_outbound_1
nat (inside) 1 Inside-Network 255.255.255.0
route outside 0.0.0.0 0.0.0.0 Mobily 1
timeout xlate 3:00:00
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02
timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00
timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00
timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolute
timeout tcp-proxy-reassembly 0:01:00
dynamic-access-policy-record DfltAccessPolicy
http server enable
http Mgmt-Network 255.255.255.0 mgmt
http Inside-Network 255.255.255.0 inside
no snmp-server location
no snmp-server contact
snmp-server enable traps snmp authentication linkup linkdown coldstart
crypto ipsec transform-set ESP-AES-256-MD5 esp-aes-256 esp-md5-hmac
crypto ipsec transform-set ESP-DES-SHA esp-des esp-sha-hmac
crypto ipsec transform-set ESP-DES-MD5 esp-des esp-md5-hmac
crypto ipsec transform-set ESP-AES-192-MD5 esp-aes-192 esp-md5-hmac
crypto ipsec transform-set ESP-3DES-MD5 esp-3des esp-md5-hmac
crypto ipsec transform-set ESP-AES-256-SHA esp-aes-256 esp-sha-hmac
crypto ipsec transform-set ESP-AES-128-SHA esp-aes esp-sha-hmac
crypto ipsec transform-set ESP-AES-192-SHA esp-aes-192 esp-sha-hmac
crypto ipsec transform-set ESP-AES-128-MD5 esp-aes esp-md5-hmac
crypto ipsec transform-set ESP-3DES-SHA esp-3des esp-sha-hmac
crypto ipsec security-association lifetime seconds 28800
crypto ipsec security-association lifetime kilobytes 4608000
crypto dynamic-map SYSTEM_DEFAULT_CRYPTO_MAP 65535 set pfs
crypto dynamic-map SYSTEM_DEFAULT_CRYPTO_MAP 65535 set transform-set ESP-AES-128-SHA ESP-AES-128-MD5 ESP-AES-192-SHA ESP-AES-192-MD5 ESP-AES-256-SHA ESP-AES-256-MD5 ESP-3DES-SHA ESP-3DES-MD5 ESP-DES-SHA ESP-DES-MD5
crypto map outside_map 65535 ipsec-isakmp dynamic SYSTEM_DEFAULT_CRYPTO_MAP
crypto map outside_map interface outside
crypto isakmp enable outside
crypto isakmp policy 10
authentication pre-share
encryption 3des
hash sha
group 2
lifetime 86400
crypto isakmp policy 30
authentication pre-share
encryption 3des
hash md5
group 2
lifetime 86400
telnet Inside-Network 255.255.255.0 inside
telnet timeout 5
ssh Inside-Network 255.255.255.255 inside
<--- More ---> ssh timeout 5
console timeout 0
threat-detection basic-threat
threat-detection statistics access-list
no threat-detection statistics tcp-intercept
webvpn
group-policy RA_VPN internal
group-policy RA_VPN attributes
dns-server value 86.51.34.17 8.8.8.8
vpn-tunnel-protocol IPSec
split-tunnel-policy tunnelspecified
split-tunnel-network-list value RA_VPN_splitTunnelAcl
username admin password LPtK/u1LnvHTA2vO encrypted privilege 15
tunnel-group RA_VPN type remote-access
tunnel-group RA_VPN general-attributes
address-pool VPN-Users
default-group-policy RA_VPN
tunnel-group RA_VPN ipsec-attributes
pre-shared-key *
class-map inspection_default
match default-inspection-traffic
policy-map type inspect dns preset_dns_map
parameters
message-length maximum 512
policy-map global_policy
class inspection_default
inspect dns preset_dns_map
inspect ftp
inspect h323 h225
inspect h323 ras
inspect netbios
inspect rsh
inspect rtsp
inspect skinny
inspect esmtp
inspect sqlnet
inspect sunrpc
inspect tftp
inspect sip
inspect xdmcp
service-policy global_policy global
prompt hostname context
Cryptochecksum:e5a64fa92ae465cd7dabd01ce605307d
: end

What is the best cisco ip phone for call manager and ipcc practicals

Hi i have recently started my training on Cisco Call Manger and CCIE voice from a leading cisco voice training institute (http://networkerszone.com/), and am working on 7900 series phones, is there any other phone that i should use or this is fine.

Naval,
7900 phones are good enough for both CUCM & UCCE. UCCE however doesnt support all models of 7900 series, please refer UCCE Compatibility Matrix for supported phone models.
You may also use CIPC as agent phone.
GP.
Pls rate the post if it helps !!

E1 no voice on Cisco Call Manager version 6

Hello,
We are running Cisco Call Manager with MGCP and E1 connection to a Siemens PSTN
We have started experiencing no voice when calling to Siemens or calling from Siemens to Cisco. The call goes through but there is no ring tone and no voice when the call is answered.
Also, I noticed that MGCP fails back to our Sub sometime, but I am not sure if these issues are somehow related
How can I troubleshoot voice problems with E1?
Thank you

Hi fgasimzade,
Can you check with Siemen what channel order do they follow.
Cisco VG follows descending order for outbound calls and expect ascending for incoming calls.
You can toggle config using below commands
interface SerialX/Y/Z:15
isdn bchan-number-order {ascending | descending}.
HTH,
Regards,
Mohammed Noor

Hi Team, I wuold like to know if you have any app to make Firefox OS working with cisco Call Manager 10.5. Something like Cisco Jabber for Android or iOS.

I'm interesting on buying a Firefox Smart Phone, but
I would like to know if are any app to install on Firefox OS smart phone in order to work with cisco call manager 10.5.
Something like Cisco Jabber for Android o iOS.
Thanks,

Hi Itech,
If Cisco Jabber has a webapp, or mobile version of their website available, you should technically be able to access it through Firefox OS.
You may also search Firefox Marketplace for an alternative solution:
* [https://marketplace.firefox.com/]
- Ralph

Cisco Call Manager Administration Page Drop down menu not working

I'm running the mobile app browser on my Xoom tablet and when I log into Cisco Call Manager Administration page I get the menu's but it will not pull down the options. The same issue happens in google chrome. I use your browser on my desktop and have no issues logging into Cisco Call Manager Administration page and getting the drop down menu's to appear but the mobile version of your browser is not working. Is there a work around? I tried changing the UAString to Desktop, iPad etc and the issue is still there.

Thanks but I've tried all that. It happens in Chrome also. I've tried Opera and it works on that. I've read other forums and it looks like a Java script issue that Chrome and now Firefox Mobile cannot seem to resolve. My preference is Firefox and I'm sure as more tablets get out then this will be solved at some point. I just don't know how it is prioritized. I do know that in order for Androids to be viable in the business sector they need to address issues like these with a higher priority.

E164 dialling through Cisco Call Manager and its Gate Keeper

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