We've s number of 186/187 ATAs plugged into fax machines that will work perfectly fine for months then suddenly when dialling the number it just rings constantly, the ATA isn't forwarding the traffic.
The only way to resolve the issue is to power cycle the device, resets don't work.
I did find a line in Cisco's website that ATAs need a periodic power cycle due to environmental conditions which seems bizarre to me.
Has anyone figured out a way to remotely kick them back into life, driving 20 miles to site to turn something on and off again is bonkers
FIX: Restart the publisher, as it had stopped processing SIP over UDP. Since the ATAs use SIP over UDP and other devices use SIP over TCP that explained why only the ATAs stopped registering.
We ran more captures from the publisher and the ATA and saw that the CM never received the UDP packets.
Decided to take u/malchir advice and reboot the cluster. Instantly all the devices registered. Not sure what, if any, role the DNS change played or was it just in the wrong place at the wrong time haha. OR it’s always DNS depending on who you ask 🧐😂
PROBLEM: Looking for some thoughts from the outside on an issue I am facing.
ATA190s are no longer registering with our publisher or subscriber but all other devices continue to work properly, whether they are SIP or SCCP. Logging from the ATAs doesn't seem to give much to go on but a packet capture of one coming online shows it getting its settings from TFTP then attempting to reach out to the subscriber with the SIP protocol but never receiving a response.
Logging from RTMT shows the failure reason being "DirectoryNumberMismatch" but nothing has changed for these devices. This error did not start appearing until after a colleague changed/updated DNS settings in the environment.
CM is version 10.5
I inherited my firm's Cisco telephony in June after the last admin was fired. I handle most of the R&S matter so I guess it naturally landed on me, I rather like it though.
Anyway, what do you use to troubleshoot Finesse and UCCX issues. Where do you start your investigation after having the user do the standard log out, restart, etc steps. Where can I view logs, server and client logs. Can I access there Finesse logs without interfacing with the user, as in through powershell or commandline.
Some examples are I have users complaining of getting kicked of Finesse, of Finesse partially loading but failing to load some of the widgets. Now yes sometimes switching web browsers will resolve the issue or something simple like that but I want to find the actual reason, this environment has just been used but not properly administered nor maintained and I want to figure out how.
Also how do you monitor the servers themselves, ELK or Solarwinds modules, a wallboard of your creation?
Im trying to build a Finesse gadget that would allow access to a web application we host in house. The problem is im using the "WebAppSampleGadget" available in the Finesse dev site and this gadget only supports iFrame websites, but unfortunately the web app i want to show in the gadget does not do iFrame due to security vulnerabilities.
Is there a way to get around the whole iFrame thing? Has anyone done this and have been succesfull embedding a website in Finesse that does not support iFrame? I about ready to got to the call center managment and tell them that "due to security reason is better to open a new tab instead of adding the website to the finesse agent" I just want to make sure i've done my due diligence before i say i cant do what they want.
Hey guys! Hope you are doing well!
I am working as a Windows Server Admin and have a job interview as a System Engineer, Junior level.
However on the job description it mentions:
The prime function of this role is to design, deploy, operate and support Cisco Unified Communications (UC) solutions that will be provided through our data centers and offices infrastructure. As Junior System Engineer working in the Unified Communications team, you will progressively:
Anyone has any ideas where can I find possible interview questions and ideally answers (or if you've had an interview if you could share your experience)?
Thank you for your time and have a wonderful week!
Hey, Can anyone point me towards a good guide or example config for inbound call routing from a SIP ITSP in order to flexibility route calls to CUCM via flexible digits. Would be nice to be able to change number of digits from say 4 to 11 digits for testing.
The ITSP delivers SIP calls in the format 1234568A0@domain.com (sorry can't grab the full format).
This is for a lab setup and have one SIP account via Sipgate and would like to route calls in and out for testing inbound DDI's and potentially sending calls to UCCX and CUC.
Thanks in advance.
Anyone know if you can sign into jabber if you have a primary email address that is different from the domain ? For example if domain is 123.com - and my email is address firstname.lastname@example.org it works fine but if i have a user who logs into the same domain but has a different primary address such as email@example.com it fails. Am i making any sense ? Just wondered if anyone had come across this issue before. Thanks
I'm working on a VoIP deployment to replace some older non-Cisco phones in a building at our university campus. This particular building still has some public use phones wall mounted to small privacy alcoves. I'm looking around for some models that we could use and thought I'd ask what others might have used.
For a basic model I was thinking this:
We can disable speakerphone in the config and there is at least a screen so they can confirm who they've dialed.
The only thing holding me up from just moving forward is that this building originally had the goal of being a showcase of technology on our campus. Our desk phones are all 8945 or 8845 units and it occurred to me it might be nice to include some quick dial buttons for student facing departments on the courtesy phones that would then allow them to have a video conversation (which would mean a video capable sku). While the 8845 datasheet shows they can be wall mounted, it seems like the screen would be at a really awkward angle.
Between KISS design and that I don't know if students would ever actually use the phones to contact other video endpoints on campus, I may end up going with those 3905 phones anyways but figured I'd ask if anyone else had a similar use case that involved a "wow factor" and what you went with.
I am having a strange issue with music on hold. Randomly, it will play the wrong music. For example, a person calls in, and receives the wrong music. I call in and it plays correctly. Then a few days/weeks later another report that it is wrong, yet when I try the extension, it is correct. I checked all of the Incoming Auto Attendant's and they are set to use the correct stream. Any ideas?
Curious if anyone can clarify what the SIP RFC states about how 2 devices (phones, B2B user agents etc) setting up a SIP call should handle unsupported codecs being advertised in the SDP. My understanding is that if a SIP server/phone receives an INVITE with a SDP advertising various codecs, some not supported by the server/phone but others are then it should simply ignore the codec's it doesn't recognize and proceed to negotiate a codec it does.
I ask as I've been battling Telstra the last few days due to an outbound DTMF issue where our Jabber clients fail to negotiate any DTMF payload with Telstra after I had upgraded our CUCM from 11.0 to 12.0(SU1). Now the cause of this issue is due to Cisco introducing the X-ULPFECUC codec for Jabber audio streams as of CUCM 11.5(SU1) which cannot be turned off in CUCM in combination with my CUBE being configured as per Telstra's integration guide which states that "pass-thru sdp content" should be configured on the CUBE under the sip global settings.
This resulted in outbound calls from Jabber clients negotiating g711alaw (which is fine) but Telstra wouldn't respond with any DTMF payload type in its 183 Session In Progress (normally we settle on RTP-NTE payload 101) if our INVITE contained the X-ULPFECUC codec.
Turning off "pass-thru sdp content" in the global sip settings and using the specified codec groups on the CUBE resolved this issue and Telstra have since emailed me saying they don't support the X-ULPFECUC codec along with the OPUS codec and some others, yet before the upgrade with "pass-thru sdp content" we were sending the OPUS codec to Telstra and they just ignored and sent back g711alaw with RTP-NTE payload 101 and everything hummed along perfectly.
So am I right in thinking that Telstra's not properly handling the call setup as per the SIP RFC or am I just mad?
Have you setup a Unity Connection system to pair with multiple CUCM clusters? How did it go? Were there any gotchas or lessons learned?
We have all our phones on one cluster (UC9) and plan to migrate phones and voicemail to a new cluster (UC11). This migration needs to be phased due to business requirements. Using COBRAS to migrate voicemail users and messages per group is taking a long time. I'm thinking of moving all the voicemail pieces from the current Unity Connection to the new system in one pass and then slowly migrate the phones as we can.
Hello, wondering if anyone is using airwatch and jabber for iPhone ? We seem to have an issue where the app goes into idle mode and gets unregistered in call manager. This then means that calls do not come through to the iPhone. Was wondering if there is anything we can tweak in airwatch to maybe change the app behaviour ?Thanks
I've done alot of searching but can't find any clear information on how to license the UC suite for homelab use. My apologies if this is a bit vague, but the links that people have posted on Cisco's forum don't go anywhere and their new "Promotional Software" site is completely empty. Can anyone offer any more information on this? All I can seem to find is that I need to go through a Cisco partner.
I just finished moving all my CUCM servers to 11.5Su3 from 9.X and installed a couple of IM&P servers. My Unity CXN servers are on 11.0. Then Cisco comes along with this nice bulletin. My CUCM servers now need to upgrade to SU4 so that my IM&P servers can also be SU4. Also, I hate upgrading unity CXN but now I've got to take that to 11.5.1SU4 as well. Oh and that PLM server I just migrated all my licensing to instead of co-hosting it on my CXN server, needs to get upgraded too.
I just got my Friday evenings back from this project and along comes this.
Hello, does anyone know if this can be moved or removed when someone is sharing their screen with another user. As soon as you want to work at the top of the screen the bar appears and then covers anything under neath it.
I was curious if anyone else has encountered an requirement we're seeing.
Security Auditors are advising we need to turn off Web Access to phones. This obviously limits us in a lot of ways, not just troubleshooting. Is there a way to add a password requirement to web into a phone?
I can see ways to do this from the network layer via firewall rules/ACLs, but this would be a daunting task in our environment. Basically our goal is to have only our UC team have web access to phones, but nobody else. Has anyone else had this requirement? Is there a way to password protect web access to phones? Any other ideas, aside from firewall/ACLs? Possibly a URL filtering rule or something?
Is anyone running Exchange Cumulative Update 8 or 9 with Unity 12.x or 11.x? TAC says they support up to Exchange 2016 CU7. I am thinking that CU8 or 9 will work since its only using EWS, but I need confirmation.
I'm an out classed VoIP admin in work as a side job to helpdesk. I'm generally good when it comes to what I have to do on a daily basis but it really shows how clueless I am when a new topic comes up.
We run CUC, CUCM, CUPS,CUAC and expressways. Work flat out aren't willing to send me externally for training. I've watched the CBT CICD vids and have the Cisco press book but found they're good but a bit lacking for work.
I've been recommended INE by one of the voice engineers I know as it goes into CCIE level. Has anyone used INE for collaboration and if so was it worthwhile ?
Edit: sorry I know I said upto IE but meant that in a way they've got plenty of coverage on different levels. Not going specifically for IE vids alone
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 : 220.127.116.1100-10 Active Master Version: 18.104.22.16800-10
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!