Cisco :: CUCM Calling Party Info - User Extension Number?
Mar 21, 2012
In CUCME if you do not configure any translation rules and leave the system mainly at default, when a call is routed to the PSTN the CUCME system sends the true calling party ID which would be a users extension number. Is it correct to assume that a CUCM server based system, when too left at the majority of default (without translation rules or stripping etc) that it will send the true calling ID to the gateway?
I have some tunnels which terminate to my home router. I'm allowing the other ends of the tunnels to use my voice setup. I need to prepend *67 to all called numbers which don't originate from my house. I don't want people calling my home number based on the caller-id number they see when someone across one of the tunnels calls.
So if 5008 calls 212-333-4444 I want it sent to my provider as *672123334444. If 5001 calls a number, I don't want it touched. Can I do this? I can use IOS or CUCM here.
I want to install some AIR-ANT2566P4W-R antennas outdoor against the wall, the 2602e access points will be mounted on the interior side of the wall, altough the wall is 30cm thick, so I need some antenna extension cables.What type of coax extension cables would be recommended ?
Found you on Google and prays that the regulars here will take pity on a former Juniper admin. I've got a brand new shop to handle that is all Cisco including CUCM 8.x and I have zero Call Manager experience. How to enable international calling for a single user
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brief flow/steps for making sure a user can dial international? I figured it was as easy as making sure their DN CSS had the ability to do so, but apparently not.
I'm working with a Cisco Unified CM 7.0There are sevral ip phones in the place.ach phone has a 3 digit extension tied to it. no matter where on the network that phone is plugged in that phone has the same extension. This is a problem because one phone (extension 134) is damaged and was swapped out with a phone that has extension 102. I want that phone that has extension 102 to have the extension 134 instead.So that is my problem. I must say I am merely a CCNA and I have no idea about this voip stuff. I thought it was a simple log in to the server and make a single change. boy was I wrong. I clicked on call routing, then I clicked on Directory number and I got a list of all the DNs. I then click on extension 102 and 134. I noticed the MAC of the respective phones was in a field titled associated devices. I then dissociated both phones thinking I could just associate the appropriate phones next. as soon as I hit save that entire section disappeared. Now those extension aren't tied to any phone and once more those phones now have no extensions. I am freaking out because I can't undo this. I've clicked on literally every single option inside this server. I am in way over my head. I will include a picture to illestrate what I am talking about.
I have a pbx avaya phone system with lines and extension number.I needed a phone with line and extension.So i run the ethernet cable to the office made all connection and connected ethernet wire directly to the pbx avaya, and i got the phone working with the extension.But my question is to get the line working and the extension i have to punch down the cable from the pbx, the ethernet wire and the pair of wire coming in with the line?
I want to be able to gather some time metrics based on source IP, and destination port. Is it possiable to track how much time a user spends using a service based on it's port number. I have figured out how to capture all the data, and I can then look at timestamps, but I would like a better way if possible. Can this be done at the firewall, or do I need a different appliance?
Both of these ISDNs are up, this gives us 4 channels. Someone said they recieved a busy tone when they attempted to dial out. I looked over the system and seen there are two outbound pots dial-peers. Each dial-peer references one of the BRI ports. The preferences are the same on each dial-peer. I think what is happening is that the system is randomly selecting one of the dial-peers due to the preference, even if both channels of the BRI are in use. How does the system know if that port has both channels in use? I've not used ISDN before, so tried to enter the B-channel sub interface and the system (UC500) tells me I cannot do this. I was thinking about adding each channel into a trunk group and then referencing the trunk group in the dial-peer. I can obviously add both BRI's into one trunk group.
Is it possible to connect an analog phone to an FXS port on a CME router and a VoIP phone to a switch connected to said router and have voice connectivity between the phones? Also, is it possible to connect an FXO port on that same CME to a RJ-11 wall jack to connect to the PSTN and be able to call that VoIP phone as well as the analog phone from my cellphone? I'm trying to tie as I read the CCNA Voice OCG.
How to get this VC traffic through this network. I am quite new to ASAs, and I feel I am making a tonne of headway
The setup: The VCS Expressway is currently sitting within the DMZ (ip 172.16.10.10) which is NAT'd to 208.118.125.130. The internal VCS Control is pointed to the the VCS Expressway within the DMZ (as it is designed to do).
I have accessibility from the DMZ to the internal network. And from the DMZ to outside seems works partially (more on that below).
The problem:
Calls signalling is able to get through my network, but not media. IE, the call initiates, but media does not connect. Furthermore, I registered an internal endpoints (10.2.20.118) to the DMZ expressway (172.16.10.10). The registration works fine, but again, when I call to another endpoint (internal GK register endpoint to external GK registered endpoint) the call sets up, but media doesn't establish.
Here is the network topology, and below that is the run config.
ASA Version 8.0(4) ! host name ignite CSGfw enable password awUSpLuFs5wdhqJE encrypted password 2KFQnbNIdI.2KYOU encrypted names name 10.0.0.0 inside-network name 172.16.10.10 VCSE
we have a Cisco 1801 at one of our remote sites that uses an ADSL line for it's primary connection and ISDN as backup. Despite no apparent problems with the ADSL line the ISDN is repeatedly dialling the internet every 2-3 mins during office hours and considerable expense to the customer.
Just recently bought a Linksys E4200 router and it's been rocking so far! I upgraded from a WRT320N. However it seems that calling with SIP (using my Voip Buster account) doesn't work anymore.I don't seem to be able to register with the Voip service. Port is the standard 5060 (if my memory serves me well, I'm not at home now) and I already activated on the router the special tick for Voip ALG calling. Router is upgraded to the latest firmware and it's the V1 router.With all of the above, I still can't connect using SIP to VoiceBuster. It worked like a charm with the WRT320N so it's something with the router?.
I can't make skype calling on any land line or cell phone anymore. I think I Isp have blocked it. Is it possible that any isp can block ant internet calling?
We are upgrading from Callmanager 4.0(2a) to new hardware and the latest version. We have Callmanger 4.0(2a) and would like to upgrade to the newest version, at least to CUCM 8. We already have the new hardware that is compatible, smartnets and licenses. After looking at the matrix sheet it appears (there were several different upgrade routes but I could only find documentation for upgradring to 7.1(2) and 7.0(1) from 4.x so I am going with 7.1(2)) should we go:
from 4.0(2a) to 4.3(1)
from 4.3(1) to 7.1(2)
from 7.1(2) to 8.0(1)
this gets us to 8 which we can then upgrade to 8.5 or 8.6
So, we have 4 servers, 2 old (Pub, Sub) and 2 new (Pub, Sub). What is the best plan of attack here:
1) Backup 4.0(2a), then do an inplace upgrade on old hardware from 4.0(2a) to 4.3(1). Then backup 4.3(1) and do another inplace upgrade on the same server to 7.1(2). Then load 7.1(2) on the new hardware and restore backup from 7.1(2) on the old hardware. Then backup and inplace upgrade to 8.0 on the new server?
Or
2) is there any way to not have to "touch" or do an inplace upgrade on the old hardware running 4.0(2a), and just run a backup on it, load 8.5 on the new server and import the data over from the backup?
What vmware i have to use to install cucm 7. vmware workstation 7 or vmware server or exsi 4 or 5. Any correct link to download and vmware config settings.
the customer has CUCM in the inventory database of LMS 4.1. He has all accesses from LMS to CUCM. One phone 7961 is seen in the UT report. When the customer click on the CUCM in the inventory - there is no IP phone registered in the CUCM.
the customer has CUCM in the inventory database of LMS 4.1. He has all accesses from LMS to CUCM. One phone 7961 is seen in the UT report. When the customer click on the CUCM in the inventory - there is no IP phone registered in the CUCM. What is wrong?:-( See the attachment.
I've looked in many places but cannot see how or if it is possible to configure a phone, in CUCM to have a feature ring instead of the normal ring.In CUCME you go into the ephone x configuration mode, and assign the DN to the phone with the button xfx command. What this gives you is a slightly different ring tone when a call comes through. If I am not mistaken it is the same ringtone they use on the show "24".Is there a way to do this "feature" with CUCM?
I have a problem with my Cisco 7961 phones not registering on my CUCM 8.6 install on ESXi 5. Weird because I have Cisco IP communicator phones that register with no problem. You guys know what I can be missing. I have restarted the CUCM and services multiple times. The phone log on my phones say it can't find dhcp and DNS unknown host but my CUCM is configured by IP address. I also attached some screenshots
I have a CUOM 2.3 and CUCM 7.0 installed, we are planned to have a major upgrade to CUCM to 8.5 soon, do i need to upgrade my CUOM ,since CUOM is a Network management system? If CUOM to be upgraded to 8.X do we need to purchase the upgrade license? Is OM 2.3 to 8.X is a direct upgrade?
My Customer today have CUOM 2.3 and CUCM 7.1.3. He is planning to do upgrade CUOM 8.6 and CUCM 8.6. My question is: Is it possible to do upgrade CUOM directly to 8.6? If CUOM to be upgraded to 8.X do we need to purchase the upgrade license?
Looks like I still have an issue with LMS to recognize the IP Phones in UT as IP Phones. SNMP RO on Call Manager is enabled and is green in CM (e.g. topology) - so SNMP get is basically fine. The Phones are recognised as End Devices in UT.
As far as I understand, now if I start a Phone Aquisition, the CUCM is polled by LMS to gather additional information about the phones. So it seems there is a problem with the SNMP polling of the Call manager?
bit of a newbie to this call manager stuff but I was having a bit of a play earlier and for the life of me I can not see an easy way to list which directory numbers are in a particular pickup group. I can see the list of pickup groups and how to assign the directory number to a pickup group but surely there's an easy way to find out which numbers are in a particular pickup group???
I was thinking, is there a way to view a report on the configured numbers within CUCM? E.g. not only extension numbers but hunt pilots and other things such as that? Rather than manually going to each page.
I am trying to configure my ASA 7.2(4) to inspect SCCP traffic from a CUCM v7.I have been advised that the ASA device needs to support the version of Skinny I am running.What version of Skinny does ASA 7.2(4) support? How can I find out what version my phones are using?
I have a 2951 which i want to integrate to the CUCM and wish to plug a Siemens ISDX into it which is the best card to use NM-HDV2-1T1/E1 or WIC2-2MFT-T1/E1? its QSIG
can i use a cat 5 cable to extend a 9 pin serial cable to connect my PC to my home audio controller? I have a serial to USB converter but my PC is upstairs and my audio component is downstairs?
Any 3rd party CA to provide a root cert for the ACS & verify csr's generated by the 7925's? I've spoken with customer service at Verisign and GeoTrust and either i'm explaining it totally wrong or their not providing CA services for this type of secured environment. sHA1 using EAP-TLS.
I've recently obtained a Cisco Aironet 1242AG WAP. I do realize this is not a standalone router. I would like to use it as an extension of my current router. So right now, the status light is alternating colors. If I understand this correctly, it cannot find the controller. I cannot telnet into it, or use the web broswer interface. They both say the connection is not available. However, when I check my DHCP table from my router, I see that an IP has been assigned to the MAC for the WAP. I've tried doing a factory reset by holding MODE before I plug in the power, and waiting until it turns amber. After a few minutes and going through a few LED changes, it goes back to continuosly alternating colors.
We have CUCM & lync server setup and its working fine if any one calling each other with DNS load balancing. But customer want ot use Cisco ACE 4710 instead of DNS LB.We have configured Cisco LB and deploy but when CUCM user calling to lync user call going to disconnect after 4 sec and also Lync users unable to make call to CUCM.And also LB deploye between share point but when users are trying to open web session that time web page going to show page cannot display.
I would like to use the NCS 1.2 to monitor Juniper SRX 210 firewall. When I try to import the MIB File from NCS, which show "Error: Failed to load MIB File "mib-802" because it is not in the resource path.what I can upload the MIB File from Juniper. [code]
