I would like to push out the new software with the server name and "use windows credentials" already configured in the .msi, but I don't know where in the .msi these settings are. Any help?
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Edit Mitel Connect .MSI
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Outlook Contacts not syncing to Mitel Connect
We just upgraded to the newer Mitel Connect from ShoreTel 14.2. We are having some users whose Outlook contacts are not syncing to Mitel Connect and I have some users whose Outlook contacts are syning fine. All setttings in Mitel Connect and Outlook are the same for all users. I have looked in the ContactUploadAddIn-180703 Log but not really sure what to look for. Any help would be appreciated. Thanks!
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ECC Agent Status Change Radomly
For nearly 2 years I've been experiencing issues with agent status changing randomly. Some examples include:
Agents will put themselves in a lunch status but return from lunch in a Forced Release status
Agents will not show any status in the status panel in the upper-left corner of the web page even though they have a status (busy, lunch, etc)
Agents will not be able to take their first call through the web page. The call will get delivered but will not show in the web page, only at the phone
Agents will get bumped to a release state in the middle of a call
Agents will be in wrap-up but a new call will be delivered before they end wrap-up
Any thoughts? I'm happy to answer any questions but listing everything we've done to try and rectify it would run pages long.
Thanks in advance.
Agents will put themselves in a lunch status but return from lunch in a Forced Release status
Agents will not show any status in the status panel in the upper-left corner of the web page even though they have a status (busy, lunch, etc)
Agents will not be able to take their first call through the web page. The call will get delivered but will not show in the web page, only at the phone
Agents will get bumped to a release state in the middle of a call
Agents will be in wrap-up but a new call will be delivered before they end wrap-up
Any thoughts? I'm happy to answer any questions but listing everything we've done to try and rectify it would run pages long.
Thanks in advance.
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Contact Center Transfer Button
We just upgraded from ShoreTel v14.2 to the Mitel Connect update that includes the move to the Mitel Contact Center web-based version. Our agents, while in the contact center, are not able to select the transfer button to perform a cold transfer or consultative transfer. It's a button that is greyed out.
Does anyone know if this is a known issue? The recommendation we received was to use the Mitel Connect Client to perform the transfers
Tips or validation? Thanks.
Does anyone know if this is a known issue? The recommendation we received was to use the Mitel Connect Client to perform the transfers
Tips or validation? Thanks.
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WorkGroup Queue - not showing second caller is ringing
Hello,
First time poster but constantly read the forums as it's a great source for solutions! We currently have a shoretel system running 19.44.7900. We have a work group that includes one agent, the receptionist and also has a queue setup. A few weeks ago she says that when she is on the call to someone and someone else calls, the button on the phone would flash allowing her to pick it up (which makes sense), now, the call just goes directly to the queue (as I have test called, it doesn't ring, it just goes straight to the queue) - but she has no idea that someone else is calling. I can see in queue handling that "Allow agents to pickup from the queue" is ticked, it's set to ring 3 times, but honestly, nothing has been changed in the last few weeks (most of the time things just work so we leave it as it is).
Not sure if anyone has any suggestions as to what this might be?
First time poster but constantly read the forums as it's a great source for solutions! We currently have a shoretel system running 19.44.7900. We have a work group that includes one agent, the receptionist and also has a queue setup. A few weeks ago she says that when she is on the call to someone and someone else calls, the button on the phone would flash allowing her to pick it up (which makes sense), now, the call just goes directly to the queue (as I have test called, it doesn't ring, it just goes straight to the queue) - but she has no idea that someone else is calling. I can see in queue handling that "Allow agents to pickup from the queue" is ticked, it's set to ring 3 times, but honestly, nothing has been changed in the last few weeks (most of the time things just work so we leave it as it is).
Not sure if anyone has any suggestions as to what this might be?
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Mitel Connect phone state not syncing with Outlook
I have our users extensions setup to "Availability state synchronized with calendar status" however, our phones never seem to follow Outlook hours. For example, my Outlook hours are from 7am-4pm but my Phone never switches to Out of Office at 4pm. This feature worked for us in our old ShoreTel before we upgraded to Mitel Connect. Thanks!
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Pass-though port on phones active or not
I have a request by a company's IT; to disable the PC pass-through Ethernet port on all phones. I figured out how to do this using the Config files, but now their IT has another request; How can he see which phones have an active Ethernet connection from the phone to a PC, so that they can move the connections prior to disabling the pass-though port.
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Ccir
Hi All,
We are using ccir/cciv for better reporting of our calls. We had an issue over the weekend where the services got stuck and the cdrc2g databse cdr table didn't update.
The services have been restarted and all is working again but there is a "loss" of data for the period that he service was stuck.
Is there anyway to re-process this? or are we stuck with a gap in the data?
Any thoughts?
We are using ccir/cciv for better reporting of our calls. We had an issue over the weekend where the services got stuck and the cdrc2g databse cdr table didn't update.
The services have been restarted and all is working again but there is a "loss" of data for the period that he service was stuck.
Is there anyway to re-process this? or are we stuck with a gap in the data?
Any thoughts?
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Shoreware Director Event 119 Warning
Good evening all,
I am still a fairly new Shoretel/Mitel administrator, so still learning the ropes. One of the things I noticed in the event viewer of my Shoretel server was Event 119 being logged multiple times a day. The description is something like this:
Switch ST100DA: Excessive number of packets lost from 10.11.10.2 (15753 out of 36829).
Obviously the numbers change but the devices in question are always the same, the ST100DA and the host 10.11.10.2, which is our SBC. These are SIP trunks.
So one of the things I have gone through with a fine toothed comb was QoS settings. I'm 100% confident QoS is configured end to end and is working properly. The SBC and ST100DA are on the same switch and the ports are nowhere near capacity. Nobody is reporting any call quality issues on these calls where the lost packets reported are so high, there would have to be some noticeable degradation. But I happened to be on a call earlier, after hours, and when I hung up, I got an notification about excessive packet loss on the call, but the call quality was fine. It then occurred to me that I had been placed on hold during the call, and there was a period of silence where there was no music or anything playing while I was on hold. I was able to duplicate this behavior on another call.
So I guess my observation is that this is something that happens on SIP trunks when we're placed on hold, but my questions are, is this the way its supposed to work? I guess it could make sense to just not send any RTP packets during periods of silence, but it doesn't make sense to me that the Shoretel system would perceive this as packet loss. Is something misconfigured between my Shoretel and my Adtran SBC? Is this a bug/bogus error report and safely ignorable?
Thanks in advance.
I am still a fairly new Shoretel/Mitel administrator, so still learning the ropes. One of the things I noticed in the event viewer of my Shoretel server was Event 119 being logged multiple times a day. The description is something like this:
Switch ST100DA: Excessive number of packets lost from 10.11.10.2 (15753 out of 36829).
Obviously the numbers change but the devices in question are always the same, the ST100DA and the host 10.11.10.2, which is our SBC. These are SIP trunks.
So one of the things I have gone through with a fine toothed comb was QoS settings. I'm 100% confident QoS is configured end to end and is working properly. The SBC and ST100DA are on the same switch and the ports are nowhere near capacity. Nobody is reporting any call quality issues on these calls where the lost packets reported are so high, there would have to be some noticeable degradation. But I happened to be on a call earlier, after hours, and when I hung up, I got an notification about excessive packet loss on the call, but the call quality was fine. It then occurred to me that I had been placed on hold during the call, and there was a period of silence where there was no music or anything playing while I was on hold. I was able to duplicate this behavior on another call.
So I guess my observation is that this is something that happens on SIP trunks when we're placed on hold, but my questions are, is this the way its supposed to work? I guess it could make sense to just not send any RTP packets during periods of silence, but it doesn't make sense to me that the Shoretel system would perceive this as packet loss. Is something misconfigured between my Shoretel and my Adtran SBC? Is this a bug/bogus error report and safely ignorable?
Thanks in advance.
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Contact Center Director and Web Agent HTTPS capable!
You can enable HTTPS now in Contact Center during the upgrade process.
I had to update the following fields in the "wecc.ini" to FQDN so when I logged into https://<server FQDN>:8433/ccd or :8433/ecc it resolved to FQDN that matched my certificate instead of IP:
ECCServiceURI
CCD2ServiceURI
I had to update the following fields in the "wecc.ini" to FQDN so when I logged into https://<server FQDN>:8433/ccd or :8433/ecc it resolved to FQDN that matched my certificate instead of IP:
ECCServiceURI
CCD2ServiceURI
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Incoming call quality not as clear as outgoing.
I have AT&T IP Flex SIP on a ST100DA switch. My problem is that all of my incoming calls seem to have some static type of artifacts in the voice. It isn't inaudible, but it does make it difficult to sometimes understand the caller, especially if they have an accent. I'm new to this system so bare with me. During my testing I have played with changing codecs. AT&T told me I should be using G711. The default AT&T Codec has PCMU/8000 (G711) first and G729 second. Looking at my TmsNcc logs it seems that incoming calls are coming in as G729 and outgoing is PCMU/8000, which makes sense that outgoing is G711 since outgoing calls are crystal clear. I swapped the codec order and had G729 first, then I had the same static artifacts on both incoming and outgoing calls. If I flip it back then its just incoming again. I can't for the life of my figure out a way to have incoming calls come in on PCMU/8000. I've even created a test codec that only lists PCMU/8000, but looking at the TmsNcc log, it still shows G729 which makes no sense to me. AT&T says that they don't control that and its a PBX issue. I'm a bit lost. Its been doing this since day one (5/22). Both AT&T and Mitel at one point were pointing at QoS, But I would think that my call issue would happen both ways, not just incoming. I've even tried it with only the server, ST100SA and an IP480 phone on the network, no changes. Oh, I am not using an Ingate SIParator, its running native. I'd really prefer not to have to purchase another piece of hardware if I don't have to. Any help would be greatly appreciated
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Police showed up after a spontaneous 911 dial..
Any ideas?
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Upgrade help
We are still running a pretty old version of Shoreware Director (14.41.1108.0) and want to upgrade. It looks like ShoreTel Connect ONSITE is in my upgrade path but I think I'm getting confused between build numbers verses software versions or something along those lines. My vendor is offering "Connect ONSITE Advanced license bundle" and "ShoreTel 14.2 Software (General Release) for SBE100 and Enterprise customers" and "OPERATOR CALL MANAGER (ST 7.5)" to get me up to date. Is this correct?
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One way audio for several seconds in consult transfer
A remote user at a small office receives an external, inbound call on an analog trunk that lands in a hunt group. She she transfers the call using consult, she cannot hear the party she is calling for about 4 seconds, then audio flows normally. Then she transfers the call and all is well.
Site has a SG90, a SG90V. HP 1810 switch interconnects everything. Bit slow but sufficient for 8 users. DVS server is local, connection back to HQ is a VPN running on 100Mb fiber.
HQ is on 14.2 build 19.49.8600.0.
Firmware is up to date throughout the realm.
My thought (that I'll be doing this afternoon) is to throw in a workgroup running on the DVS. The DVS is used for Telstrat's Engage recording server, but should have plenty of CPU to handle a few calls.
Any troubleshooting ideas would be appreciated before I call support.
Thanks!!
Site has a SG90, a SG90V. HP 1810 switch interconnects everything. Bit slow but sufficient for 8 users. DVS server is local, connection back to HQ is a VPN running on 100Mb fiber.
HQ is on 14.2 build 19.49.8600.0.
Firmware is up to date throughout the realm.
My thought (that I'll be doing this afternoon) is to throw in a workgroup running on the DVS. The DVS is used for Telstrat's Engage recording server, but should have plenty of CPU to handle a few calls.
Any troubleshooting ideas would be appreciated before I call support.
Thanks!!
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Workgroup filter on Recent? Another way to filter?
It would appear to me that the only way to know if a call came in via a direct extension or via a workgroup is by looking via the call slip information.
Anyone else familiar with a way to filter call history by workgroup to only isolate those calls or quickly identify what are workgroup vs personal?
It looks like the Workgroup Agent license only gets you filtering voicemail by workgroup, not actual call history details.
Anyone else familiar with a way to filter call history by workgroup to only isolate those calls or quickly identify what are workgroup vs personal?
It looks like the Workgroup Agent license only gets you filtering voicemail by workgroup, not actual call history details.
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Filter Directory on Phone to Users in a Site
We're started to deploy our ShoreTel system across multiple sites and I've received requests to filter the directory on the phones to just extensions within the same site. I can't find any documentation that it is even possible. Anyone been able to do that?
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Shoretel Voicemail (User not receiving voicemail)
Hello,
We have a user that has been receiving voicemails since last week but he stopped receiving them this week on his phone and shoretel communicator.
I checked his configuration and it's suppose to go to voicemail after 4 rings and it does. I even left a test voicemail but he never got it on his end. When the user dials his voicemail the auto attendant says he has no voice mail.
Any suggestions?
Thanks!
We have a user that has been receiving voicemails since last week but he stopped receiving them this week on his phone and shoretel communicator.
I checked his configuration and it's suppose to go to voicemail after 4 rings and it does. I even left a test voicemail but he never got it on his end. When the user dials his voicemail the auto attendant says he has no voice mail.
Any suggestions?
Thanks!
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No audio on transfers to Soft phone.
I have a very odd issue. One user with a soft phone on their laptop can make and receive inbound and outbound calls off a sip trunk just fine. Extension to extension is just fine, even across sites. The issue happens when a call comes into a sip trunk and the operator transfers that call to the user with the soft phone. 100% of the time no audio either direction. We have two sites with sip trunks and have tested this from both with the same result. Also tested an internal transfer and the audio works fine. I have checked all the subnets, net masks, gateways and routes are correct. I disabled the firewall on the laptop for testing and no change. I have exhausted my troubleshooting knowledge. Anyone run into this?
User is on windows 10 using connect client build 213.100.2453.0 and thankfully is the only soft phone user in the company.
User is on windows 10 using connect client build 213.100.2453.0 and thankfully is the only soft phone user in the company.
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Workgroup voicemail Message Waiting Indicator on phones
I have Googled until my fingers bled, but I cannot seem to find a definitive answer on whether this is possible or not.
Situation: I need to have a general delivery mailbox that notifies the members of the workgroup that there are voicemail messages waiting and allow them to listen to them and action them accordingly.
The answers to these types of questions I have encountered usually say:
Situation: I need to have a general delivery mailbox that notifies the members of the workgroup that there are voicemail messages waiting and allow them to listen to them and action them accordingly.
The answers to these types of questions I have encountered usually say:
- Use Connect Client. Not an option in this scenario as the phones aren't next to PC's.
- Create VM mailboxes for each phone and then forward a copy. This has some pretty obvious disadvantages as we would never know if a VM had already been listened to and actioned. I can't imagine calling somebody back 3 or 4 times because it went to multiple different mailboxes and the staff not being sure who had done what.
- E-mail a copy of the VM to everyone. Not practical for the same reasons above.
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What's the correct way to log an extension out of a phone?
So this year I wanted our teachers to log themselves out of their phones before they left for the summer. In years past I've manual gone in, which is a pain in the butt and no one on this forum has been able to tell me how you can bulk log out phones, so one by one I set the phone to Any Ip Phone. That logs them out of the phone. This year when I had them press 7, 3, and then 2 to log their extension out of their phone it didn't work. My question is, should I set the phones to soft switch instead of any IP phone? I don't know why the 7, 3, and 2 wouldn't allow them to log their extension out of the phone.
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