Group paging in Call manager
do call manager 5.x or unity has feature in present that enable us to use it for group paging or public address in case of emergencies like evacuation, or any third party integration with Call manager for group paging ?can you please help me out
you assisance will be higly appreciated
Hi Zeeshan,
This feature (Group Paging) is not built-in to any CCM version but is most possible (via IP Phone speaker and Overhead) using these highly rated Third party products;
CDW-Berbee Informacast
http://www.berbee.com/public/berbeesoftware/InformaCast.aspx
IPCelerate
http://www.ipcelerate.com/assets/docs/IPcelerate_ER.pdf
Hope this helps!
Rob
Similar Messages
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Does Call Manager v8.5 support paging on mulitple phones (groups of phones) simultaneously or does it require another product? We purchased IPSESSIONs a while back to use for paging but I was told that the function was in Call Manager. We have had CM for a few years and we are just now getting ready for paging and some other UC functions.
Next question. With IP sesssions I have to configure on IP multicasting throughout our network to support paging. Can I assume that even if Call Manager supported it internally without the necessity of IPSESSIONs that I would still need the IP Multicasting protocol support on our entire network for paging?Hello,
Chris is right that 3rd party app is needed. But I don't agree that Informacast is the best one.
What about an alternative in 2Ring PHONE SERVICES which can deliver much more than just paging?
If you are interested in getting more info, follow this link: http://www.2ring.com/NewJuice
FREE and TRIAL licenses are obviously available.
As for your 2nd question, yes, multicast should be enabled in entire network.
Regards,
Martin -
Paging feature on Call Manager 6.x
Is there a way to setup Paging on Call Manager 6.x without using third party software like IPCelerate,Berbee etc.
Paging functionality is not built-in to CUCM 6. You can either use 3rd party software as you mentioned or integrate with an overhead paging unit such as Valcolm.
Hope this helps.
Brandon -
Hunt Group with Pots lines in Call Manager
Hello all. I have a site with 12 Pots lines. It is all configured via Call Manager. We are using mgcp. I asked the telco to setup the numbers to hunt which they said they did but still when I call the main number second time it just rings and eventually get some voicemail that is not from the call manager. I think I need to setup Hunting in the Call Manager? Is that correct?
Line Group --> Add the extensions in here and set them up to hunt?
Hunt List --> Create a Hunt list and associate a Line Group with it?
Hunt Pilot --> I add all the extensions in here?Let's take one step back. What are you trying to accomplish? I am assuming you want to market one number but have the ability to accept multiple calls on that number. If that is the case, having the telco set up a hunt is the way to go.
Beyond that, where do you want those calls from the PSTN to route? Are you wanting those calls to ring a hunt group on your side? If so, you will need to do the following:
1. Configure a Line group. (Call Routing -> Route/Hunt -> Line Group) This is where you will be adding the extensions. You will also need to specify a distribution algorithm. The distribution algorith, is where you will be choosing how you want the extensions to ring. For instance, if you set the DA to broadcast, every extension in the line group will ring at the same time. Click Help --> this page, while you are in the line group configuration page to get more information about the settings you can modify and different algorithms you can use.
2. Configure a Hunt List. (Call Routing -> Route/Hunt -> Hunt List) You will need to create a new Hunt List and add your line group to it. The hunt list is evaluated in a top down fashion. You can get pretty tricky with multiple line groups with different distribution algorithms in your hunt list, but if you just want something basic,just add the line group you made.
3. Configure the Hunt Pilot. (Call Routing -> Route/Hunt -> Hunt Pilot) The hunt pilot will be the dialed number that triggers the hunt group. Make sure you select all the required settings. You will be selecting the hunt list you just made in this configuration page. Be sure to set a maximum hunt timer and final destination so that the calls do not just ring forever. You can get pretty granular with the settings in here. Reference the help --> this page document for more information.
4. POINT THE FXO PORTS TO THE HUNT PILOT. This is very important. Calls to your analog lines will not trigger the hunt group unless you point the ports to the Hunt Pilot. If you are using MGCP/SCCP you will need to enter the hunt pilot in the "Attendant DN" field for EVERY FXO port. If you do not enter a value here, CUCM will have no idea what to do when the port rings. It will not answer the call if there is no value specified. This would explain why your calls were ending up with the telco voicemail service; CUCM was never answering the call.
If you completed the above steps correctly, you should be able to call the main number multiple times and have the calls go to a hunt group on your side, no matter which line the call ends up coming in on.
Let me know if you have any questions, this can be kind of tricky at first.
PS, if the calls were going to a voicemail that wasn't from Call Manager (Unity Connection) call the telco and have them remove voicemail service from the lines, assuming you have configured Unity Connection to handle that.
Hope this helps,
Dallan -
Can't remove registered ephone in call-manager-fallback
This ephone and dn keeps registering so long as call-manager-fallback is not shutdown.
RTR001#show ephone registered
ephone-1[0] Mac:0FD4.9DA0.D415 TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 6/5 max_streams=1
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:10.32.21.183 * 26602 SCCP Gateway (AN) keepalive 4 max_line 2 available_line 1 dual-line
port 2/0/21
button 1: cw:1 ccw:(0)
dn 2 number 4851 CM Fallback CH1 IDLE
Preferred Codec: g711ulaw
Lpcor Type: none
The MAC 0FD4.9DA0.D415 identifies port 21 on a Cisco VG224. After shutting down that voice-port, the ephone doesn't register when call-manager-fallback is enabled.Well, that is one idea that I've already had, Linc, but I'm reluctant to use the "nuclear option" for obvious reasons. I'm actually wondering now if the Secure Cert / OD problem is affecting Profile Manager. See this thread: https://discussions.apple.com/message/23686348#23686348
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Active Directory integration with call manager
Hi,
I am facing issues while Integrating the CCM to my Active Directory using AD Plug-in.
SITE SETUP:
1. Windows 2003 Parent Domain Controller located remotely with GC.
2. Windows 2003 Child Domain for the Parent DC located Locally with GC.
3. Cisco CallManager 4.1.3 sr3b
My Requirement is to integrate CCM with my Windows 2003 AD.
My Questions are:
1. Do I need to Provide the Parent Domain name or the Child Domain name while performing the AD Plug-in Setup?
2. Does my Call Manager need to have the Forest access of the Active Directory (i.e., Does it perform some modifications in the Parent Domain)?
3. Does the user account (which is used for Directory Integration) need to have direct members of Schema Admins or thru some other domain admin groups (i.e., Admin user -> Child Domain Admins Groups -> Parent Domain and Schema Admin Groups)?
Can anyone can help me on this?
Thanks,
V.Kumar1. Do I need to Provide the Parent Domain name or the Child Domain name while performing the AD Plug-in Setup?
Use the root domain, in this case the Parent domain.
Cisco does not recommend having a Cisco Unified CallManager cluster service users in different domains because response times while user data is being retrieved might be less than optimal if domain controllers for all included domains are not local.
2. Does my Call Manager need to have the Forest access of the Active Directory (i.e., Does it perform some modifications in the Parent Domain)?
Yes, actually all domains in the forest share the same Schema, which will be modified after running the AD plugin.
3. Does the user account (which is used for Directory Integration) need to have direct members of Schema Admins or thru some other domain admin groups (i.e., Admin user -> Child Domain Admins Groups -> Parent Domain and Schema Admin Groups)?
Account should be a member of the Schema Admins group in Active Directory, try the one in parent domain.
Correct permissions for CCMAdministration and similar example for your setup:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_chapter09186a00806e8c04.html#wp1043057
HTH -
Cisco Call Manager DR situation
Hello,
We have 2 Cisco Call Manager in our site and place in 2 different locations connecting via a layer 2 network.
They form a cluster group and the production one is the publisher and the DR one is the subscriber.
In the DR situation, if the layer 2 connection between 2 CCM is broken, is it possible to change the subscriber to be the publisher and make changes on that?
Would you please guide me how to do so?
Thank you.
Best Regards,
Terry ChowHi Terry,
Thank you for your question. However, the Small Business Support Community is limited to Cisco Small Business Products.
Your question below relates to a Cisco Classic Product which our community would not be able to help you with.
The best area for you to post your question would be at the Cisco NetPro Collaboration-Voice-video forums: https://supportforums.cisco.com/community/netpro/collaboration-voice-video
Best regards,
Cindy
Cindy Toy
Small Business Community Manager
Customer Advocacy
Cisco Systems, Inc.
www.cisco.com/go/smallbizsupport -
Call Manager 4.1.3 VS 4.2.3
Looking for recommendations and experiences with implementing 4.2.3. I have to decide between 4.1.3 and 4.2.3 with Call Manager and Unified Messaging. Also looking to eventually upgrade to 5.1. Any help is appreciated
a quick list of 4.2 features/enhancements includes:
user features/enhancements -
* call pickup notification
* one touch call pickup
* one touch group pickup
* other group pickup
* directed call park w/BLF
* login/logout of hunt groups
* CCM assistant on phone
* complete transfer on-hook
system features/enhancements -
* AAR support for calls on no bandwidth
* call forward on no-register
* device mobility
* h323 overlap sending/receiving
* h323 annex M.1 support for h323 GW & h225 trunks
* mlpp enhancements
* v.150 secure modem support
administrative features/enhancements -
* support for password aging, complex PWs, oneTime PWs with LDAP
* voice quality stats on cal-by-call basis
please see the following link for more CCM 4.2 info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_data_sheet0900aecd8042402c.html
a quick list of 5.0 features/enhancements includes:
sip trunk & endpoint support features/enhancements -
* native support for sip devices
* cti for ISP phones
* presense information for sip devices
* fault, config, accounting, performance, security enhancements for sip support
* sip trunk enhancements for external applications
* third party sip devices supporting RFC 3261
* sip line side RFCs
* sip trunk RFC support
licensing features/enhancements -
* each device corresponds to a device license unit
* DLUs must be purchased to cover the number of devices connected to CCM
* third party sip devices require DLUs
localization features/enhancements -
* many language support enhancements
please see the following link for more CCM 5.0 info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_data_sheet0900aecd8042403e.html -
Hi Everyone
I'm taking my PUB out of my call manager group as the 3rd option for phones to register to.I'm moving another server into it's place and I want to turn off the call processing function on my PUB. Is the Cisco Call Manger service the correct service that I need to deactivate under the service activation section?
EricYes
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How does Tandberg Gatekeeper call Call Manager SCCP extensions
Am I missing something here? I have CUCM 8.6 with h225 trunk built to Tandberg Gatekeeper (works fine). I have two Tandberg 150 video endpoints that are registered to the Tandberg Gatekeeper, extensions 8001 and 8002 (alias is what you call it I guess). The video enpoints can dial eachother by their extensions and work fine. The CUCM SCCP phones can dial extension 8001 and 8002 and ring the video phones to make voice only calls, which is fine. Had to build route patterns/lists/groups for this. How the heck do you dial a CUCM SCCP phone, like extension 1450 from the Tandberg video endpoint? Theres nothing in the Tandberg Gatekeeper to configure to route this call from a Tandberg Gateeper registerd phone to the Call Manager SCCP phone. This is just a simple lab setup.
Anthony,
Its interesting you say that because that is the total opposite of my experience. I'm currently working on a CME 9.5 to Exchange 2013 deployment where most users have DIDs.
When a call comes in to the extension to which the DID points, and the call rolls over to Exchange typically we receive a message stating that there is no mailbox associated with the extension because Exchange is reading the first diversion header information
(Wireshark shows two headers and the DID is always numbered 1) which has the DID number rather than the redirecting extension. I've had to either add the last 3 digits of the DID to the users mailbox or manually transform the SIP INVITE diversion header
to get around this problem. -
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 -
View Call Manager Call Real-Time?
Cisco Forum Members:
Is there a way to view Call Manager calls real-time?
The application is that someone picks up an IP Phone, dials an access code to overhead paging, then fails to properly hang up the call as which point their subsequent conversation is broadcast over the paging speakers. Can it be determined which Directory Number is connected to the paging port (a router FXO/FXS port)?
Thank you.
DanThe easiest way would be to look in the GW that houses the paging port with the show call active voice command. For more info look at this link, http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008019ab88.shtml
Please remember to rate all useful posts.
Sent from Cisco Technical Support iPhone App -
Issue with User web page into Call Manager
Hello
I have an issue where all my users can log into their User page on UCM but when they make a change to speed dial or anything else, the change does not reflect on the phone in UCM. Doing a reset from their web page doesn't reset the phone. This is UCM 7.1.5 and the standard ccm end user group is enable for all end users.
thanksHi Bill -
Have you verified your User Management Roles and effective permissions? Go to User Management - Roles - select Standard CCMUSER Administration and for the permissions you want, ensure both Read and Update are selected.
Role Information
Application
Cisco Call Manager End User
Name
Description
Resource Access Information
">Resource Description Privilege
read update
CCMUser: Access List
read update
CCMUser: Device
read update
CCMUser: Directory
read update
CCMUser: Fast Dials
read update
CCMUser: IP Phone Services
read update
CCMUser: Line Settings
read update
CCMUser: Personal Address Book
read update
CCMUser: Plugins
read update
CCMUser: RemoteDestination
read update
CCMUser: Service URL
read update
CCMUser: Speed Dial User
read update
CCMUser: User Settings
read update
You can also check an individual's effective permissions by using these procedures:
Viewing a User's Roles, User Groups, and Permissions
This section describes how to view the roles, user groups, and permissions that are assigned to a user that belongs to a specified user group. Use the next procedure to view the roles, user groups, and permissions that are assigned to a user in a user group.
Note: You can also view user roles by using User Management > Application User (for application users) or User Management >End User (for end users) to view a particular user and then display the user roles.
Choose User Management > User Group.
The Find and List User Groups window displays.
Find the user group that has the users for which you want to display assigned roles.
Click the name of the user group for which you want to view the roles that are assigned to the users.
The User Group Configuration window displays for the user group that you chose. The Users in Group pane shows the users that belong to the user group.
For a particular user, click the i icon in the Permission column for the user.
The User Privilege window displays. For the user that you chose, this information displays:
User groups to which the user belongs
Roles that are assigned to the user
Resources to which the user has access. For each resource, this information displays:
Application
Resource
Permission (read and/or update)
Now if both items above look OK, you might check your DB replication status. I assume you have a Publisher and one or more Subscribers? User phones registered to Subscriber? You can check replication several ways:
CLI
RTMT
Unified Reporting on CUCM administrator web page - select "Unified CM Database Replication Debug" report. This is the easiest.
The desired Replication State is 2.
Here is some further information:
Check the DB replication status on all the Cisco Unified Communications Manager nodes in the cluster to ensure that all servers are replicating database changes successfully. You can check by using either RTMT or a CLI command.
• To check by using RTMT, access the Database Summary and inspect the replication status.
• To check by using the CLI, enter the command that is shown in the following example: admin: show perf query class "Number of Replicates Created and State of Replication"
==>query class :
- Perf class (Number of Replicates Created and State of Replication) has instances and values:
ReplicateCount -> Number of Replicates Created = 344 ReplicateCount -> Replicate_State = 2
Be aware that the Replicate_State object shows a value of 2 in this case. The following list shows the possible values for Replicate_State:
0—Replication Not Started. Either no subscribers exist, or the Database Layer Monitor service is not running and has not been running since the subscriber was installed.
1—Replicates have been created, but their count is incorrect.
2—Replication is good.
3—Replication is bad in the cluster.
4—Replication setup did not succeed.
I'm thinking it's more your Roles/permissions and not replication, but I included just in case. Hope this helps!
Ginger -
Call Manager register fxs port with voice gateway- problem
I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager -
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
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