Phone

Hello,Currently going through a weird issue with a couple of our sites, feels like we’ve tried everything but nothings fixing this one issue. Every couple days some of our phones randomly stop receiving calls, when you call the line it rings twice then audio from the call comes through for about 1-2 seconds then drops. Nothing fixes this issue, I’ve changed SIP timers TCP & UDP SIP server expire timers, Outbound proxy server expire timers, Keep alive types, Keep alive intervals (set to 20 seconds). Enabling the Rport, changing the audio codec, firmware updates, disabling SIP ALG for the site bridged router and more that I've forgotten about at this point. Its really odd since some of our sites are perfectly fine and then these 4 sites we have with these phones constantly run into this issue. A reboot is the only temp fix for it then it just breaks again. I contacted RingCentral support but so far nothing has come back after sending off packet captures when the phone is showing this

The call queue has an extention why can't I add it to the phone hud?? I want to add call queue extentions to the receptionist phone but I can only add users.

How do I access recordings from inactive users? 

I am trying to eliminate my receptionist AI and also need to double check the caller ID. And making sure that my overall account is where it needs to be for cost. Thanks

Is there a way to limit the Caller ID choices for users based on the site they are assigned to and only show their assigned number, only the site number(s) for the one they are assigned to and any message-only numbers assigned to that site as well (i.e. fax numbers).Currently, under Caller ID restriction settings, choosing “Allow site numbers as caller ID” and “Allow message-only number as caller ID” shows ALL of those numbers across the entire system and doesn’t limit the choices based on what site the user is in.Thanks, 

how do I get an IVR Menu to be the company auto-receptionist default responder for all incoming calls

We have been live on RingCentral EX for over a month now and haven’t had issues until yesterday.  I have tickets open but was curious if anyone else has / had experienced this.We have 125+ Polycom desk phones.. starting yesterday we’re having 2 issues.1-way audio or no audio at all.  The user hangs up and calls back and ALWAYS gets successful 2-way audio on the 2nd attempt. Phones are showing Not Registered then go back to normal after a few moments.  When this happens, the port does not go down on the switch side.  I’m the sole firewall administrator and no changes have been made there.  The only thing I know that’s changed is we got a notice that the Polycom firmware would be updated to 26.2 on May 17th.  

As far as I can tell the new AI Assistant note taking and transcription features are only available if the call is taken on the desktop app or phone app. What about calls that are made or received using a physical desk phone? 

Good morning,Our Vet Clinic phone line is down. We cannot send text messages, and we are not receiving calls or voice messages. 

Our team is having trouble accessing the admin portal this morning. I got a “Bad Gateway” message appear after sometime. The team is also having issues toggling their queue status. When they toggle it on in the desktop app, it will toggle off and present the following error message: 

My RingCentral account and extension are currently tied to the ultimate Super Admin role because RingCentral Professional Services set it up that way and we have yet to change it so it’s not tied to an actual user. I recently deployed our first hot desk phone and have noticed that when users are logging in or out it shows in my softphone’s call logs (*90 and *91 entries). There are also some outgoing calls in my logs that I didn’t make that are related to the group that uses the hot desk phone. It’s only a handful of calls at this point, but I was wondering if that’s expected behavior for hot desk phones or if there’s a configuration setting I’m missing. Appreciate any insight!

I preformed the correct steps to complete a conference call, and it did not work.  The first person got disconnected when calling the 2nd person.  This process has happened on other teammates desk phones.  I cannot complete any conference calls from any desk phone which is a new event.  The phones always let us do conference but in the last couple of weeks we cannot complete.  These are the steps I am taking. Start a Call: Begin by calling the first person you wish to include in the conference. Initiate Conference: Once the first person has answered and you are on an active call, look for the "More" softkey on your phone's screen. Press it, and then select the "Confrnc" or "Conference" softkey. This will put the current call on hold. Dial the Second Person: Dial the phone number or extension of the second person you want to add to the call. Merge the Calls: After the second person answers, press the "Confrnc", "Conference", or "Join" softkey again. All three parties will now be connect

Hello!   The phones in one of the restaurants say “Off Hours Mode” in the top left even during work hours. For a while they were sending everyone to voicemail when they shouldn’t. Our other tech support for the company was able to fix this, but the phones still say “Off Hours Mode”. I checked in the admin portal for the After Hours schedule to see if that was wrong, but it’s not. Phone calls are coming through, but how do I change the phones from “Off Hours Mode”?

I found an old thread but did not want to revive it.  I need to either remove or reduce the volume of the sound that plays at the beginning of the page.  Using Polycom VVX model phones.  Page tone/beep too loud | Community

We are using a RingCentral recommended CyberData 011146 as the SIP paging server to integrate with our legacy paging system and amplifier.When someone initiates a page by using the hard phone key or dialing *84, we are having volume issues. It seems that RingCentral adds a "beep" tone at the start of the page. The beep tone that RingCentral adds is significantly louder than the speaking volume of the person announcing the page. We have struggled with normalizing the volume between speaking and the RingCentral beep through volume adjustments on the amp and in the CyberData web UI. We have not had good results.The CyberData SIP paging server has the ability to play its own page noise or beep when a call is connected, but it does not block the incoming RingCentral beep. It is the case that we can have both the CyberData and RingCentral beep play at the same time, or we can have the RingCentral beep alone. There appears to be no way to suppress sound for the first 0.25s of a ca

HiI am new to voip and purchased 3 phones through ring central 2 cordless w76p and one desk phone t54w.However i am unable to park calls on my cordless p hones despite technical support trying many ways.tried ## 803   but immediately get reply” this call group does not accept calls”I created 3 park slots which are easily visible on display adjacent to  hard keys .(801, 802, 803).If i park call on desk phone- i can easily view on display o f cordless: “blf numbers “ with park slot lable and caller id.Anyone have any suggestions? Have gone through 8 technical help people( all  helpful but no solution)

Hello,I use an oldish but fantastic Plantronics headset C052A connected with a APU-76 USB adapter, on my W11 PC. I also use a Jabra desktop speaker on the same PC, USB connected.Both work 100% perfectly in Teams and Zoom. I can switch between them during a call and they both work reliably, and the headset never skips a beat.In Ringcentral (Desktop: 23.2.21.7380 Windows (x64) 2023-05-18 07:03:19) the Jabra works 100% OK. The Plantronics headset keeps cutting out. This is random and generally lasts for a few seconds, then sound is re-established. In a 5 min call this happens 4 or 5 times. When it cuts out, the headset "clicks" and both incoming and outgoing sound are muted. There isn't a repeatable pattern to this. This also happens when the Jabra is disconnected.I have tried downloading and using the Plantronics hub software (https://www.poly.com/gb/en/support/downloads-apps/hub-desktop) but it's no better with or without this running and allowing Ringcentral to manage

It would be helpful to send text messages at a specific time so it's more acceptable and professional, especially when it's late and we can only check it at a later time, we can at least draft a response and send it on a more acceptable time of the day.,It would be helpful to schedule text messages so we can send them at a later time when it's more acceptable by professional means.

I am able to manually provision the ata. Im able to make a call but have no voice.I have other cisco ata’s on this network so I dont think its the sip alg. Is there a setup guide ?

I see that the Cisco ATA 191 has 2 phone ports on it but only 1 seems to work.  Can they be configured to support 2 phones being plugged in.  They could share the same phone number.  This would be important if I had both fax and phone in the same location.

Hi i’m building n AI supervisor app for call monitoring and trying to get some clarity on the supervise limits. Can my supervisor extension only support 5 concurrent calls? Or is that limit by device? My use case is i want to be able to monitor at 20 concurrent calls. Do I need 1 user extension and 4 devices, or do i need 4 extensions with 1 device each?

How to provision a Mitel 480G/485G with Shoretel firmware (v802/803) to ringcentral These phones are now supported on RC with autoprovision phone per (https://support.ringcentral.com/article/certified-supported-deskphones-ringcentral.html#mitel). However, the instructions given by the RC admin portal wizard and also on https://support.ringcentral.com/article/assisted-provisioning-mitel-deskphones.html will not work if your phones are on older shoretel firmware. I spoke with Mitel SIP phone support and here are the steps that worked. Sharing for anyone who is trying to convert these phones to RC.Make sure the phone is added as a device in the RC admin portal using the mac address and selecting the correct model - see steps 1-7 only on https://support.ringcentral.com/article/assisted-provisioning-mitel-deskphones.html if you aren't familiar with adding devices to RC admin.Factory reset your 480G/485G by pressing Mute then 25327# while on the idle screen. It will pop up asking if you&

I need to know what my contract terms are?

We have discovered a peculiar problem that has begun occuring after upgrading our VDI environment to Horizon 2512. We have previously been using Horizon Clients 2503, and 2506 without any incident. However, since the upgrade to 2512 and when using RTAV for our audio devices (NOT THE RINGCENTRAL PROVIDED VDI PLUGINS), both RingEX and RingCX crash the VDI connection when calls are established, or the agent changes their status in CX. We have no problems with any other audio applications. And we also don’t have any issues for users using Windows or MacOS clients.To be clear, this issue definitely sounds like a problem with Omnissa’s Horizon Client 2512 for Linux. However, I can’t ignore the fact that this behavior only occurs when we are using RingCentral apps.I am looking to see if there are any other Horizon 2512 customers who are using the Linux client with their Ring Central environment?