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Question about services not starting on Windows Server 2016

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Hi everybody,

Last week our vendor did an upgrade to the latest release of Mitel Connect Director Build 22.11.9300.0, and it blew up (TAPI was completely hosed). After hours with Mitel TAC they advised to do a bare metal recovery. So anyway, did that, restored the backup, installed the upgrade to 22.11.9300.0 and it was successful. All the services started up and the board is all green.

However, I noticed today that there are three Windows services that are set to startup automatically but they won't start:

Net.Tcp Listener Adapter
Net.Msmq Listener Adapter
Net.Pipe Listener Adapter

These are apparently related to Windows Process Activation Service. Are these needed by Connect Director? Everything seems to be working fine but just thought I would poll the experts here since I was unable to find anything. Thanks in advance.

Transfer contacts to another user

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We have a receptionist that will be leaving next month. Outside of our company directory, she has a large number of external contacts in ShoreTel Communicator. We have hired her replacement and would like to transfer the contacts to her. Is there a way to export/import those contacts? I can't seem to find anything in the documentation on this. We are running the latest build of 14.2 (19.50.1000.0). Thanks!

Migrate to Google voice

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Hello everyone and I hope everyone is safe and healthy.

My organization is looking to move away from our ancient shoretel system and into google voice.

I was wondering what it would take for us to be able to port all our numbers to it? I believe we pay for a single line and our shoretel system breaks it up into DID numbers? I am not sure how it all works, but I suppose if going to google voice, would it even be possible to port them over? Thank you for your help and time.

Silent Monitor issue, stays active for listening but speaker cannot hear customer

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We are training a new customer service agent by using the Silent Monitor feature so she can listen in to calls. However, we have run into an issue where the caller stops hearing our agent. Meanwhile, the person listening in can still hear the customer. Any thoughts? This appears to only be on this single extension. Thanks!

Implementing PSTN Short Circuit

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Good afternoon all,

I want to implement PSTN Short Circuiting so we stop wasting SIP trunks dialing ourselves. I've found the document that says you need to add ;9d to the custom Site Rules. I did that but it did not do anything different. Here's what I've done so far:
  1. Enter Director Service Mode.
  2. Go to System>Sites>Custom Rules
  3. Entered ;9d in custom. Saved.
Do I have to restart anything for this to take effect? Also, should I have copied the "Generated" contents to Custom and just added ;9d at the end?

Thanks in advance.

Moving users

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I have an employee that moved to a new position. Switched her users to the new phone and had her set the voicemail message. Calling internally, everything works fine but if you can and leave her a voicemail from outside our network, you get the user she replaced voicemail. What am I missing?

Regards,
DJ

Switchboard overview

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Is there a "switchboard overview" in the Mitel system? We were migrated to Mitel from ShoreTel 14.3 a few months back, and we just had a situation where someone didn't hang up from a page call, and both the receptionist and myself couldn't figure out a way to see ALL extensions and their current call status. Is such a thing possible at all, either from the Connect client or from the Director? I'm just trying to support a common business function.

Thanks.

mitel contacts

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I need a way to export everyone's individual contacts. I've poured though the MySQL database and can't seem to locate them anywhere. I seem to have found some from when we ran the original shortel software, but we've since updated to the mitel connect. (not a fan) Does that create it's own data store or something outside of the shortel MySQL data store? I apologize for the lack of specific's.

mitel connect build 213.100.3079.0

Shoretel Communicator Error:The operation failed for unspecified reasons

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Hello all,

I have a user at my company who is receiving the error in the title anytime he uses the "mobile" option in Shoretel Communicator, specifically when he tries to make outbound calls. Please see the attached screenshot. He dials the number in Shoretel communicator and instead of getting the ring-back to his mobile, he gets the error attached and then the call automatically ends. The call never rings back to his mobile for him to answer. Also we are working off of Citrix. Any help is appreciated! Thank you! I tried resetting Shoretel numerous times, had him restart his phone, try it on mobile data/WIFI. Also Reset his Citrix all together. Had him restart his computer. Tried using the web version of Citrix. nothing worked. It also doesnt work for him in A different instance of Citrix so i dont think Citrix is the issue. Please help!


Capture.PNG

Remove extend wrap icon

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I need to remove the extend wrap time in communicator, ECC integration is enabled. I'm not sure where I can remove or disable. Attached Picture below shows where the agents are clicking to extend.

shoretel communicator

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Hi there we are having an issue with the shoretel commincator not loggin in and causing failover mode. Sometimes a reboot will work sometimes a password change will work but nothing consistant. I currently have a user whom i have uninstalled and reinstalled twice and still not good. I also have tried these steps below.

On the client, they are under:
\Program Files\ShoreTel\Contact Center Agent\Agents
There will be a folder for each agent. Delete them, they will get re-built when the agent logs in again.

On the Server, the files are held in :
\ShoreTel Contact Center\Agents\nnnn

Again there will be a folder for each agent, you can delete this if you like, but make sure that the agent is logged out when you do.
When a new agent logs in that doesnt already have a folder on the server they are built from is a file:
\ShoreTel Contact Center\Agents\ETAS.INI

Voice mail integration with Dragon speech to text and audio sample rates

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I have a customer that wants to transcribe the voicemail or call recorded audio to use with Dragon audio to text engine software.

They have a very knowledgeable employee that heads the IT side there and was asking if there was a way that the Connect system could change the sampling rate of the audio coming from voicemails to suit the Dragon software.

I could not find anything in the maintenance guide, admin guide nor on this forum.

He is looking to change the vm/recorded call audio to 16Khz wav files but I am not sure it's possible without first exporting the audio from the server and converting it manually with some other software.

Anyone have any ideas?

Thanks in advance for your time.

Mobility issue with incoming calls

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We have the Mobility Router set up and configured. Users can connect with the app, and we can make outgoing calls without issue. However, incoming calls are not routed to the mobile client. I am not sure what we could be missing.

Using work groups in Director for two tiers of helpdesk calls

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Appreciate any help or advice you might be able to provide this is my first post so please excuse me if there is already a topic on this, but I couldn't locate one.

For our internal helpdesk we use Mitel Director instead of Mitel Contact Centre Director to manage/configure our helpdesk telephone line. We have two work groups to distribute the calls between two levels.

The 1st work group is set to forward calls to the 2nd work group when agents are Busy, No answer or Logged out

(See attachment #1)

The 2nd work group forwards to voicemail if agents are Busy, No answer or Logged out. We can queue calls on the 2nd work group before they reach voicemail but the call will only queue for agents in the 2nd work group if an agent becomes free in the 1st work group the system doesn’t know to forward the call back to the 1st work group to the available agent.

We tested setting the ‘Forward to’ option on ‘Busy’ on the 2nd work group to send it back to the 1st work group but if everyone is engaged on calls the system doesn’t know what to do with the call and cuts it off. We wondered if there is a way to queue the call for both work groups for a period of time before sending it voicemail?

(See attachment #2)

Thank you

IP480G Mute+CLEAR, Static IP, No Service

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Hi All,

I'm working with Mitel TAC on this but from reading forums, see conflicting information. Some people state static assignment works fine for 400 series phones, others have trouble. Here's the situation.

Client is a financial organization.
All IP addressing is static
Phone is at the HQ site on the same subnet. Even tried the same switch.
IP230 MGCP phones work fine. Before migrating 14.2 to Connect this March, IP400 phones seemed to work fine.
Post migration, phones cannot connect to server but work with their cached certificate. Newly cleared phone never work.
Captured packets, examined phone logs, looked through IIS logs, the SSL cert is never downloaded.


I took a phone back to my office and captured packets with DHCP on, then cleared and captured with static IP assignment.

Watching Wireshark during a DHCP boot, after the three-way handshake, the first thing downloaded is the SSL cert.
DHCPCapture.PNG

With static IP assignment after a Mute+CLEAR#, it does not download the SSL cert.
StaticCapture.PNG

As such, when the phone tries to connect to the voice switches, it fails to connect as the root CA is invalid (phone doesn't have one).
SSLCertFail.PNG

TAC noted that in the phone log, there is no FTPServer set. Without DHCP, there is no way to set an FTP server, But...The SSL cert is downloaded via HTTP, so I don't see how they are trying to tie those together.

Any ideas, other than standing up a DHCP server, to get the SSL cert to the phones?

Please note, I've discussed DHCP with the organization's IT management, however they refuse to use DHCP because of "security" concerns. I've explained that I can grab enough network info with a 30 second WireShark capture that I could likely plug a computer in and gain network access, but they don't understand how easy it is after I've explained.

Thanks!
David

App Dialer Login Prompt not opening (Cannot Login)

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We are using the Mitel App Dialer version 10.0.0.1.

We have some users that are unable to login to the app dialer, i.e. the moving bubble splash screen does not open and prompt for credentials to login to App Dialer.

I have tried everything from uninstalling all Mitel apps and reinstalling, cleaning out registry and user profile, clearing credential manager, etc. Same result, cannot login to the app dialer.

Has anyone experienced this?

Instant Messaging on SA100

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Hi all

A few years back we were told that you could install the SA100 virtual machine without any licenses and you would be able to use it as an instant messaging server indefinitely. Does any know whether this is true and can anyone test this for me?

D

Mitel Edge Gateway question

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If you have a Mitel Edge Gateway and its located in your DMZ so that external users can get to it via the internet, what should your Mitel Connect clients point to by default (regardless of whether your computer is inside the network or outside (at home)? Before the Edge Gateway, everybody pointed to the internal Director server. Now i'm being told to point all Connect Clients to the externally available Edge Gateway address. But if i'm inside the network that seems odd. Help!!

BCA Lines issues on IP485 Phones

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Does anyone else have BCA Line issues with IP485 phones? Drop calls on Transfer or the phone reboots? Or lines that show as active after a call has ended? Or lines that can't be picked up (line busy or not permitted)? These happen randomly of course on Executive and Executive Assistants phones. We are on build 22.11.9300.0, but pretty much have had this issue since moving to Connect and the IP485's. Any insight would be appreciated.

Multi-Line ATA on Connect Onsite - HT818 or SPA112

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Has anyone ever used a GrandStream HT818 or a Cisco SPA112 to delivery multiple lines to a remote location? I am trying to send 2 to 4 lines at remote properties. I can get the 1st line to register via SIP back to our onsite Connect server, but lines 2 to 4 fail to register. I get this error in the events logs "The IP-Phone Device: 114950999744, could not be configured since another IP Phone device is using the same IP Address: 192.168.6.25" It seems it doesn't like that multiple lines are being registered under the same IP. Is there a work around for this? Is there a setting I missed on the Connect Server or on the ATA devices?

Thanks
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