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Firmware Version Mismatch cause one way audio?

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We have bought some refurb IP230s, they have Firmware SEV.3.4.7 and Boot ROM SEV.3.4.7 on them (we are using SEV.3.3.16P SEV.3.3.15). When they boot up and connect to my ftp server they won't accept the older version of the firmware.

Several of them have a one way audio problem. If I swap them for an older phone with the older firmware they work fine. I'm on build 14.41.1108.0.

Remote Call Unpark

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I am in the planning stages for converting a site from a Nortel/Avaya CS1000 to a Shortel/Mitel Connect Onsite.

One feature we use on the CS1000 is "park". Any person with a phone (including the operator consoles) can press "park" during a call and the call is placed on hold on one of several system defined extensions. To pick up the call any person with a phone dials that four digit extension number. Easy peasy.

Our operators will "park" a call then text page people with the extension number. If they are off site they can call an access number and then the four digit extension and be connected to the call. If they are on site they can pick up any phone and dial the extension.

We would like to replicate this on our ShoreTel/Mitel system. We want some way that people can easily pick up a call. We understand that if they are onsite they can use the unpark button or dial *12#### with the extension number. We need something that works when they call the external access number. (Or dial the "park" number on the old system to pick up a call on the new system.)

Any thoughts or experience? If something simple is not available is there any other solution?

chat / IM log

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I did a little bit of research and I remember Communicator did have an option to log chat conversations locally to UNC path.

In Connect, do IM logs get saved on the SA100 Server? If so where and can we disable that?

IP655 and 100 Mbps vs 1000 Mbps

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In two of our offices, we've found an Ethernet run in a conference room where the 655 negotiates at 100 Mbps, powers on, but displays "No Ethernet". A Windows laptop plugged into the same run negotiates at 1000 Mbps. When the 655 is plugged directly into the switch (same port as the aforementioned run), it negotiates at 1000 Mbps and works fine.

Anyone else run into this? Why would the 655 negotiate 100 and normal workstations/laptops 1000?

How To Assign A Sip Phone Or Device To Connect With Basic Settings

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As time goes on I noticed that simple things get lost and swallowed up in the world wide web. So I will periodically create posts that are hopefully simple and give anyone the directions that are searching for.

Some simple steps to adding a Sip phone to Connect:



Requirements:
  • Enable sip resources on a local switch
  • Creating or modifying a system user and giving them a sip password...this is a 6 digit password
  • Adding your values to your sip phone.
  • Confirm the sip device in director
In my example i will use the following data
  • Extension 3005
  • My local switch with sip proxies will be 192.168.1.88
  • The Polycom sip phone will have a local ip of 192.168.1.108

Configuring My User And Sip Password (ext 3005):

Sip Shore User.JPG



Assigning My Sip Resources to My Switch In Connect (192.168.1.88)

swich resources.JPG




Designate My Proxy For My Site To Use (192.168.1.88)

site sip.JPG





Configuring Your Sip Device With The Required Information (192.168.1.108)


Polycom Settings.JPG



Quick And Simple Video Of The Whole Process

Caller ID Name & Number Connect Client

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So, I made the jump to ShoreTel Connect, and over all things went rather well. But I have one complaint from users, When someone calls, the pop up notification does not show the Caller ID Number, only the Name. This isn't an issue with landlines, but for cell phones it only shows Wireless Caller, rather than the number. Is there anyway to change this?

Digging into it, I see it simply calls an HTML file, that in turn calls a .js file, but from there, I would have no clue how to edit it.

Thanks,

Brandon

ShoreTel Phones drop calls at random

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At office of 30 employees, our ShoreTel phone keep dropping connection randomly and it effects most users (at random). Sometimes you don't have to be on the phone and it loses connection with the phone server. I restarted the Shortel switch and it didn't resolve anything. It's on premise which is connected to a netgear switch. I looked at the HQ logs and some noticable erors are:
  • Swithc 10:00, Calls 2, Free Mem 58508720, Max Mem Block 57678720
  • IP Phone Display Server: A called remained in the Unknown state for more that 30 seconds. This usally indicates a failure in a switch, a SIP device, or the Telephone Management
  • Switch: Trunks unavailable to route to *phone number*
Any thoughts?

ShoreTel Director - Workgroup configured in a way I just don't comprehend.

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Hey folks!

We have a DID number/Workgroup set up to route voicemails to people's personal cell phone numbers. I need to remove a few people, but can't for the life of me figure out how the voicemails are getting distributed, nor how people's individual cell phone numbers are linked to the DID number.

Environment:
ShoreTel Director build 18.23.2412.0 accessed via Internet Explorer 11.

Workgroup configuration:
  • Nothing is configured under Escalation Profiles and Other Mailbox Options.
    No Escalation Profiles set.
  • No Schedule set.
  • On-Hours/Off-Hours/Holiday/Custom are all the same.
  • Call Handling Distribution Pattern is set to Top Down.
  • Call Forward is set to Always: Extension [voicemail System Parameter System Extension]
  • No Agents are present in the Workgroup Membership.
General notes:
  • Nobody whose number connects to the Workgroup DID has any personal phone numbers under Individual Users.
  • Under Individual Users - Personal Options - Find Me, External Assignment and Additional Phones, Find Me Primary Destination is set to None and External Assignment is not used.
  • Nothing is configured under people's Shoretel Communicator apps.
Any ideas? Any at all. I'm desperate and clueless!

Workgroups and hunt groups have stopped working

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Good afternoon.

You may have to excuse my ignorance but I am fairly new to administering Shoretel (lots of previous expierence with Avaya and others) but I am having an issue with a Shoretel Small Business Edition (Build:21.82.9623.0)

Specifically all of a sudden calls to both work groups and hunt groups are not working, they either just give progress tones or queue but never hit an endpoint (that is an extension assigned as a member)

Checked the users in workgroups are logged in.

Seems strange to lose both workgroups and hunt groups at the same time as the setup consists of a SW server (UC server?) and an ST100DA switch.

Help please!

Cost to upgrade 14.2 to current build and/or move to Connect

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Very new to Shoretel - I recently picked up a client that has Shoretel in-house. The have a contract with a local Shoretel provider to handle maintenance @ about $4600/year. Have roughly 80 seats.

We recently had someone at the client site download a new version of the Shoretel softphone and found it wouldn't work. Reached out to the provider and they said it's because we're on an old build of 14.2. For our discussion, we'll ignore whether that's true or not. Next step was client wanted to find out what it would take to get current so they could use the latest and greatest softphone.

Provider needed access to the ST server and was having problems so asked us to go in and check something. While I was in the system, I noted that Microsoft updates had not been done on the system since February 2014 (Possibly when the system went in.) I asked if this should have been done as part of regular maintenance and they beat around the bush a little bit and finally told us we'd have to open a ticket if we wanted updates done. So that's concern number one.

Second up is the quote we were provided to upgrade the build of 14.2. Now I recently discovered that 14.2 is end of life this year (For the forums, not the provider), so we obviously wouldn't want to upgrade to that regardless. But the quote we received started with the following - "Upgrade the current system in place from build 19.44.4700.0 to the current version of ST14.2 Build 19.49.7200.0."

Then further down the quote is a spec for a new UC server 30 and the licenses for Connect ONSITE. Provider is telling me all of this is required, but it makes no sense to me. So we'll be having a call on Monday. Are we upgrading the ST 14.2 build, or are we putting in a new server and Connect ONSITE, and why are they saying both?

What is jumping out at us though is the price. $1900 for the server, $1200 for project management, and $6000 for engineering labor.

Reading online, many say the upgrade should be included. At $4600/year for maintenance, we generally agree, especially since it appears no maintenance is actually happening (We handle all moves/add/changes.)

Can I get some opinions? Most of our VoIP is hosted, so we're not used to dealing with on-prem equipment and want to make sure the client isn't getting hosed.

changing VOIP service provider

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I just inherited administration of this phone system. Im running Director version 14.2 and i am planning on changing my service provider. Any one have a walk through of all the things i have to change to switch service providers? I believe my current provider is using a PRI trunk with an ISDN profile setup in director? I am switching to a SIP trunk. any help is appreciated!

Workgroup pickup by non-agent

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I have a request from a department head for the following functionality:

He wants people who are either workgroup agents to see incoming calls but not ring their phones, or non-agents to see and pick up calls monitored on their phones.

Are either of these possible ?

Trying to understand call detail report

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Been asked to run a call detail report to when and for how long a certain number calls extensions on our system.

This sounds straight forward, but the reports show that the caller places most of the calls to the Auto-Attendant,
and that these calls last anywhere from 20 seconds to 50 minutes. About 20 calls over the past month.

The caller is a former employee who no longer has a voicemail box on the system.

Why would the report show so many calls to the Auto-Attendant?

Could the caller be accessing someone else’s mailbox?

How to receive ACD call while on Personal call?

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We have an ACD group on Mitel Cloud for our Sales Team.
We have 4 agents assigned to the ACD on a longest idle distribution.
These agents also make personal calls during the day to make sales and we do not want those calls to affect the ACD routing.

We want the agents to be able to receive an ACD call while they are actually on a personal outbound call. This can increase our availability to the ACD volume as a team. right now, it doesn't work because the ACD defines a user as "busy" while on a personal call. How can we re-define the user as idle while they are on a personal call?

If we cannot modify the definition of events for the ACD, are there any other group solutions to accomplish the longest idle distribution by defining idle as "longest since last group call"?

Help needed

ShoreTel Communicator 64bit version

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Is there a 64bit version of ShoreTel Communicator with 64bit Outlook addins? TIA!
I need to know if it will work on Windows 10 too?

Setting up new site to existing Shoretel system (Testing without POTS)

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Don't laugh but this is my first install for my company. I'm using a 40/8 shoregear switch. I set up the networking equipment for the new location before taking it down. We wanted to make sure everything worked correctly before going live from the new locations existing network/ phone system. So I have a SonicWall tz300 connected to each location with a VPN. I have the switch configured to connect back to headquarters. In my connectivity monitor on the shoreware director I'm showing that all sites are connected and working. I plugged in the Shoretel phones and they connected back to the server and we tagged the mac addresses to the right users. The only thing missing that I'm aware of is that we haven't connected the existing POTS lines into the Shoretel switch. I was thinking that I would still be able to make calls within the site and within the connected sites without a POTS line? So to test I made a call to each location, it rings but when the other user picks up neither of us hear anything. But, if the user does not pick up it will go to voicemail.

I would just plug it in and test it but the site is really far away and I want to get some clarification before I get there. Because I will have to go down on the weekend when the office is closed to finish the install.

Licenses configured is incorrect

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System says we have 30 licenses configured and 29 purchased. There are only 29 users and 26 phones. I can't think of anything that actually takes an extension license other than a user profile.

Connect Contact Center - Request a Callback report

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Does anyone know of a way to report on the 'Request a callback feature' of Connect Contact Center? I've been asked to provide information on how much this feature is used but when digging through the report templates and options I don't see anything that looks like it will pull this information.

Salesforce CTI connector

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Hey all,
I have a customer running premise 14.2 and since the latest softphone push from Mitel (updated the branding from Shoretel to Mitel) they have been getting a message saying "Your connection to this site is not secure" and their sofphone won't load.
Anyone else seeing this?

Limit Calls to an ECC Queue???

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Are you able to limit the amount of calls that can enter an ECC service? We have a customer who would like to only allow 50 calls to queue up at max. So the 51st + caller in queue would be redirected to a auto attendant that plays a prompt that all agents are busy and gives the caller detailed info about their website.

I have been working with a logic switch to try and achieve this but I missing something because I can't get it to work. I play a mandatory script when you enter the service but the logic switch is not forwarding the call to the proper location when testing.



Thanks for the help in advance!!
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