I'm trying to build by very own home CVP lab. I found on Craigslist someone who is selling 2x 2911 and 1x2901, but he doesn't have a console cable to do a show tech. I have a picture and I'm seeing the following:
I assume that besides this I will need some PVDMs. Is this a good start to achieve my goal? Anything which I absolutely need to see when I do a show tech? Thank you!
SOLVED: See end ..
Just installed Unity Connection 12 for the first time and noticed that the secure IMAP port (7993) is closed, which means I can't use COBRAS to export or import messages. I've never had to do anything special to enable this in previous versions, and I can't find any configuration options to turn it on; the documentation suggests that it should just be on by default. I've confirmed that the port is not open on the server ("show open ports all") and that it's not a firewall issue.
The only hits that turn up in Cisco communities are from people running "unrestricted" versions, but I am definitely running the "restricted" version.
Product Ver : 188.8.131.5200-10 Active Master Version: 184.108.40.20600-10
Anyone else run across this?
EDIT: Apparently you need to enable export-restricted functionality in the Smart License portal, then run "utils cuc encryption enable" and restart the IMAP server on both nodes. Thanks /u/rogue_ranga!
Looking to see what others use for all company calls. The past two we've had some issues since we recently moved to a centralized platform. In the past we had a CUBE at each remote location to handle call load, but now that we just have two vCUBE's in each location we are running into issues with CPS for call company calls. The last company call we we're seeing about 55 CPS when employee's were dialing into the third party hosting bridge we've been using (PGi formerly Globalmeet). Most end users would then have to call in with Cell Phones as the vCUBE's in our two datacenters could not handle the flood of calls right before the call starts. Our two vCUBE's are maxed out by Cisco's standards with 4vCPU's and 8Gbs of memory.
I know we could get some ASR's to do this for us, but i'd like to keep the vCUBE's if possible and see if there's anything we can run internally across the MPLS that will not use SIP trunks or impact the CPS.
Open to any and all suggestions! Thanks!
I am new to CUCM and PABXs in general so bear with me with this!
I am working on a project where I am renewing a Siemens HiPath Concentrator with a modern Cisco Unified Commmunications Manager CUCM utilising Cisco UCS C220 M4 Servers and Cisco VG350 Voice Gateways.
As part of this renewal the exisiting circuits that terminate onto the HiPath will require migrating to CUCM.
There are two circuits in particular I need help with. These circuits that conect to the HiPath originate from other PABXs: an Ericsson MD110 (circuit 1234) and a Meridian (circuit 5678). These circuits I know to be hotlines/PLARs and teminate onto the HiPath via exchange/FXO cards and subs/FXS cards respectively. I have included a drawing of the existing system and the proposed solution (a like for like replacement).
My query is regarding the CUCM configuration to aid with migration. I believe I need to know how the calls are routed from the MD110 and Meridian to the HiPath and then routed to the Admin phone. I will have access to the HiPath configuration but not the other PABXs. I believe it is particularly more important with circuit 1234 as this is an exchnage/FXO card on the CUCM/VG350.
Is this correct and if so, what information do I need to know?
Hope this maks sense; please advise if you like any further information.
I'll be taking that 210-065 to convert my valid CCNA voice to Collaboration. I was planning on taking it May the 9th and my CCNA Voice expires the 14th. If I pass (fingers crossed), do I need to worry about the Preliminary Score being finalized too late? Like is that 72 business hours or weekdays only (if so I'd be in danger because of the weekend)? Does anyone know?
Thanks in advance!
We are using Imagicle with BillyBlues except it has a few limitations in our deployment.
First, it does reporting in a way that seems to work around usage billing. Unfortunately, our environment ended up going more towards a flat rate billing based on the number of devices an organization has.
Second, for call reporting it can't seem to track calls for devices set as an "Anonymous (Public/Shared Space)" device in CUCM. The data is there, it just doesn't include it unless a user is associated either in CUCM or if it left as anonymous and then a dummy user account is created and assigned that extension in Imagicle. Definitely not ideal and it creates extra management overhead.
Just curious what others are using in this space?
Guys, i have been a UC engineer for couple years now. I seem to find it diffcult to troubleshoot transcoding issues, or when there is an issue with RTP. Do you guys know of good videos online that teach these scenarios? I cant seem go find any? Looking zpecifically for what to look in RTMT logs or even sip proxy logs.
Hello, is there a way that a line group can skip a member if that member is out of office and has forwarded their phone to voicemail ? We have a longest idle group configured for a team but it is including people who aren't at their desk and have forwarded all to to voicemail.
Webex CMR Cloud and MultiStream
Cisco can't seem to give me a straight answer on whether its supported or not but from my reading of the solution design guides, firmware and CSR releases notes etc etc it is.
My setup is as follows;
CUCM 11.0.1 Expressway C/E 8.10.4X Webex Meeting Center running WBS32.12 with multistream enabled and our 'tennant' running on the CMS bridges in Webex's DC's (confirmed by TAC) Endpoints (MX700 and SX20/80's) running CE9.2.4 and registered to CUCM
I should note I know multistream capabilities with Telepresence Server based conferences were removed from endpoints in CE9.X firmware but our Webex is running on the new CMS based servers ;)
Expressways zones and CUCM trunks are all set to allow multistream, iX protocol, appropriate bandwidth control and have the max SIP message size increased to 18000, endpoints are enabled for multistream.
I see the SIP invites to Webex out the Expressways with the multistream capability advertised but no multistream happening on the endpoints. Webex client applications appear to be working as intended and I'm able to use the new Webex layouts introduce for multistream capabilities etc.
I don't really know where to go from here (TAC are working on it but Webex's support has gone down hill recently so we'll wait and see) but it's really bugging me.
The SDP contents for these invites get pretty unruly so I'm not exactly sure what I need to be looking for in regards to multistream (bar its capability bring advertised)
So.... Is anyone using a similar setup and has multistream working for Webex cloud based CMR?
Hello, I've been tasked with looking into seeing if there is a way to add location data to directory numbers in CUCM. We are running version 9 currently. The point of this is for 911 calling. I know next to nothing about CUCM, I've learned just a few minor things in my role as Desktop Support, like how to add a new phone, or configure a phone, and set up a new name for an extension. I am not Cisco certified, so I don't really understand a lot of the terminology when I try to look things up online, so I'm reaching out here for some help.
TL/DR : I've been asked by CEO to see if we can add location data for 911 purposes to directory numbers, and I am totally lost!
Currently, we have an employee that cannot log into the Agent on their PC. We are receiving Configuration error. Please ask your sys admin to associate your phone with the RM JTAPI provider user ID. However, they can access it at a different location. I have disassociated and re-associated their phone with the RM JTAPI. I have also uninstalled and reinstalled Agent. Other users are able to log in on that PC also. Any ideas?
I tried searching and reading admin/configuration guides, diagrams of all kinds and I am just not getting it. I am not even close to a voice engineer. Just rudimentary knowledge by my estimates so I wanted to ask someone with some mercy to /ELI5 this list of things that confuse me.
Where are each of these configured?
What is the basic relationship between them all? Do you need CUBE to be able to run CME? Do you need CUCM to run Unity? etc.
Thanks in advance for the explanation and I apologize if it all sounds idiotic.
Hello, We recently added additional subscribers to our CUCM cluster, and added them to a new CM Group. We've noticed some strange behavior. Initially, outbound calls would fail for most devices that would register to the new sub with a fast busy (503 unavailable, disonnect code 41). We tried adding the IP address of the new sub to the SME SIP trunk to the UCM cluster. Calls are now working, but we see differences in how calls route out with phones registered to the new sub. Some go right out to the SME over the SIP trunk, while others proxy out of the original sub that they used to be registered to. If we build a brand new device with a brand new DN that registeres to the new sub, it doesn't proxy through the orginal sub. Any ideas on why that might be? Also is having a shared DN on phones registered to two different subscribers supported (by way of device pool)?
Are there any reports that can be run to pull endpoint Serial numbers? I've been tasked with finding these for all 3500+ of the endpoints in our environment to verify SmartNet coverage and I've been unable to locate anything solid (at least for 10.5.2+) in regards to reporting on this. We do have CSP Collector and that has a good chunk of them in there (2852), but when I query CUCM for 7841 endpoints I'm seeing 700 or so more devices so i'm not sure if SNTC is catching everything.
Any help is much appreciated. Thanks in advance!
We have recently been asked by the higher ups to allow video calls from our environment to a third party that is running Google Hangouts. We have call manager with expressways and are using the Cisco hosted Webex. Does anyone know of a working solution for this? We are happy to look at third party products to resolve this if they work.
I have followed the deployment guide to a T. I was able to enable users in the Spark Admin Hub for Call Aware and Connect and the Spark remote device was automatically created in CUCM by the Expressway connector for said users. I am able to see call history and make outbound calls from the Spark app but inbound calls to the Spark app do not work. I am not seeing any evidence of the inbound calls in the logs on my Expressways so I suspect CUCM isn't routing the calls over. I did create the SIP route pattern for ciscospark.com that is routed to the SIP trunk that points to Expressway-C. Not sure what else to check as the documentation doesn't have details troubleshooting. Thanks in advance.
This is what I'm trying to accomplish: I would like to implement a paging/intercom system in Call Manager where when you push the soft key button, it rings all phones in our location, and then we can say a message (such as weather alert, etc). Is this possible?
I've recently dove into learning Python to assist with some various job functions (BAT sheet I'm looking at you) and then I had the bright idea to try to connect to the AXL API to query various data from CUCM.
After a couple of days of googling and trying some things out I've hit a bit of a wall. I started here and that sort of got me working but one of the URL's he hard codes is no longer valid and I'm not sure how important it is.
I've also looked through this thread on Cisco's Dev Net. Right away he mentions that Suds (and I assume by extension suds-jurko) doesn't work with Python 3 and of course I'm using 3.6.
So I guess in short is anyone using Python 3.6 to do AXL queries and more importantly does anyone know of a good tutorial on how to actually do these queries? It's one thing to give me a bunch of code and say do this, but I'd really like to understand all the various arguments and how to build these queries on my own.
First, off, TIA!
I have inherited the SysAdmin duties of a small to medium sized company. My experience is several years' as HelpDesk, a BS in IT, and am completing semester 3 of 4 at the local Community college in pursuit of a CCNA. I can add new users, new phones (physical and software) into the PCCE system, and make changes to the call script.
One of my tasks is to switch over from one SIP provider to a new SIP provider. Both the old and new SIP providers are active at the moment.
My plan to transition is two fold: 1. Test the connection to the new SIP provider. 2. Schedule a handoff between the new and old SIP provider. This will involve transferring control of the toll free numbers and local area code block that the company owns.
I'm stuck at #1, trying to add the new SIP provider into our system for the purposes of testing/verification.
We have a Cisco PCCE system with an A and B side. Two CUBE/VXML routers serve as gateways. SIP A - the old SIP provider - is connected through VXML2. All incoming and outgoing calls go through SIP A.
I've been able to configure our ASA firewall and VXML1 gateway router so that SIP B (the new provider) can get a SIP "Ping" to succeed. However, incoming calls cannot connect. When attempting outgoing calls, DNA reports that I am trying to use an "Unallocated" number. Note: the outgoing number assigned for testing is a different area code than the block that our company currently uses/owns.
Where should I look next?
Edit: Additional information:
I have done some work in CUCM and on the CUBE router to bring me to this point:
New Partition (PT)
New 4-digit extension using the test phone number provided by ITSP2 This 4-digit extension uses a CSS set up for this test.
New Calling Search Space (CSS)
Tied to the PT above
New Device Pool (DP)
New Cisco VOIP phone:
Associated with the DN above.
Bound to the DP above.
Associated to the CSS above.
New Route Filter (RF), with appropriate area code for the test number provided by ITSP2
New Route Group (RG), with only VXML1 in the selected devices.
New Route List (RL), with the RG mentioned above, and the Standard Local Route Group
New Route Pattern (RP):
Uses the PT mentioned above. Uses the RL above.
New Translation Pattern (TP)
Bound to the PT above.
Bound to the CSS above.
Bound to the RF above.
Bound to DP above.
pointing to the internal IP/Port of VXML1
We are using a SX20 which is on version 7.3.6, and we need to downgrade it to 7.1.4 to use the Vyopta software. The firmware we received from Vyopta, however, it is failing on downgrade. The error message is: The installation failed: Failed to extract file: /mnt/base/image1/tmp/web-upload.tmp: The new software is not compatible with this system, make sure you use the correct software image.
We have new users that are not showing up on our Wallboard system. They are showing up on reports that are ran in Unified Intelligence. This is not a 3rd part Wallboard system. Any ideas on what we need to do to get the new people to show up?
We have a CUCM server and our users use the Self Care Portal of it. I want to install a SSL so that our users don't get the SSL warning. I know I could trust the cert on all machines but this would also be a pain. We already have a wildcard certificate that we use and pay for on other server so i want to use this wildcard SSL on our CUCM server. The problem i find is that it seems there is no way to upload a certificate that is already signed.. its like it wants you to generate a CSR then sign it and upload that signed certificate. Is there not any way to upload a public and private key for tomcat to use without doing a CSR??