Quantcast
Channel: ShoreTel Forums - ShoreTel Forums
Viewing all 1780 articles
Browse latest View live
↧

Automatic Wrapup not working

December 17, 2019, 1:23 pm
$
0
0
Hi
I have a customer with ShoreTel 14.2 and ECC 8
Recently one Agent has seemingly lost the ability to go into Auto Wrapup (20 seconds before new call presented unless extended by agent)
what happens is, the existing call ends and if there are calls in Queue a new call is presented immediately
The set up for the Agent mirrors all others in ECC\Agents and everything else appears to be working as expected.
Has anyone come across similar issues ?
↧
Search

Return unanswered calls to transferor

December 24, 2019, 10:37 am
$
0
0
Hi

We have a customer who struggling to get their new Connect install to work the way they want for incoming call flows... it is currently setup as follows

External call come into "Reception" workgroup (say ext 500) via a SIP trunk
A reception operator answers
Routes the call and makes an unattended/blind transfer to the extension required (in the customers words "fire and forget")
(In an ideal world the call is answered and this story ends)
If the call isn't answered or the extension is busy they don't want it to go to voicemail box, that is the organisation policy, they want to make sure that every call is handled in a timely manner, be that routing that taking a message and making urgent contact with the recipient, transferring it to someone else or an off system transfer to a mobile phone; in essence they want those call to return to operator to direct them.

From what I can see a call will basically ring indefinitely and doesn't have any kind of automatic "call return/recall" after a timeout period - I have tested it on an extension with forwarding setup and after several minute of the unanswered phone on a blind transfer ringing I resorted to answering the call to shut it up. To get around this the installer when they put the system in set it up with a second "Call Return/Recall" workgroup (say 501) that they then set as a forward on no reply destination on each extension; in principle this sort of does what they want in that the call arrives back to an operator... the problem is it is an operator as opposed to the operator that handled the original call.

Apparently the operators don't know it is a call return/recall call (although I am a bit dubious as they should be able to tell the workgroup the call queue is in) and are thus answering it as if it were a "new" call. My gut feeling is I think what is actually being reported is slightly different perhaps and is perhaps the awkward thing for them is that when the call is represented not only is it ending up going back into a general call queue across all operators but even if they can see it is an unanswered call they don't see who that call was originally transferred too to be able to go "I'm sorry I'm not getting a reply from Mr Smith's line, would you like me to put you through to his assistant, for me to take a message or for me to try his mobile etc" but instead are having to ask as if it were a brand "new call" who the person is trying to reach - and apparently there have been instances where one operator has transferred a call to an extension and it bounce back for one of their colleagues to try to put the call back to the same unanswered extension.

I am a bit at a loss as to what to suggest - other than making sure all calls are attended/announced transfers so the operator remain in the loop until they are sure they can complete the transfer (which is what various of our other clients do so this had never crossed my mind or come up earlier). From from my experience of other phone system I know I have seen "Call forwarded from XYZ extension" come up on the display of the phone and that unanswered call will effectively time out and revert back to the transferor after a wait - and from experience been the calling party I assume this "prior routing" information being given isn't exactly unusual as I have been on the end of "fire and forget" call routing where the call has bounced back to reception having sat in a departmental queue for a bit.

I have to say I thought the original installer had come up with a fairly good workaround originally to get these call to revert back but I can sort of understand the frustration that what they are trying to achieve isn't something it appear you can do out of the box on the Connect platform and was something that existed previously in there old system.

Any bright ideas or suggestions would be welcome before the customer contemplates potentially skipping a 6 month old $50,000 phone system and replacing it with something else as they are that touchy about upsetting callers although to be fair some of whom are potentially spending millions of dollars and they very much competing in a quite specialized market on global basis to offer their services.


↧
↧

Change Phone Conference Button

December 27, 2019, 10:48 am
$
0
0
We're moving away from the SA100 to Microsoft Teams Audio conferencing. Is there a way to change the behaviour of the conference button on the phone to dial a route point rather than the SA100 extension?
↧

Force Release from Call Center when locking Windows

December 31, 2019, 5:03 am
$
0
0
Hi, Does anyone know a way to force a call center agent to logout when they lock their Windows computer? We have one call center with very poor management and agents frequently walk away and end up going into release. Calls can sometimes ring through four agents before being picked up or customer hangs up.
Thanks for any help.

I'm in a supporting role and still very new. We have ShoreTel 14.2 Build 19.42.8801.0
↧

Change Support Partner Fee?

January 2, 2020, 12:26 pm
$
0
0
We are looking at chaning our support partner to some one a little more local to support our ShorTEL/Mitel setup

I'm told there is a $600.00 processing fee to do so? Is that a bit outrageous for pushing some paperwork around?!?!

Any one have any insight to this?
↧
↧

Default AA

January 2, 2020, 2:22 pm
$
0
0
We have multiple sites and in one of our satellite sites have made some changes that require that we use the main AA in our primary site going forward for all AA functionality. I was able to direct the incoming calls pretty easily but we did discover one setting which appears a bit more difficult. When someone presses 0 while in the VM greeting of a user in the satellite site, it goes to their old AA. I realize this can be manually overridden using the routing for each extension but I'd rather not set each extension or availability state manually.

Where is the setting for the default AA that is used for the "When caller presses '0', transfer to:" setting? I assume it's a site setting but I could not find it in sites.
↧

Silent Monitoring Restrictiona

January 3, 2020, 2:13 pm
$
0
0
I am looking for a way to restrict silent monitoring for select extensions. Where I do NOT want certain extensions to be able to be monitored by anyone. Is there a way to accomplish this?
↧

Automating Mitel Connect Login

January 6, 2020, 8:54 am
$
0
0
I'm trying to figure out a way to automate login to the Mitel connect client on shared computers for my users. It was easy with the old Shoretel Communicator via the registry. It doesn't appear so easy since our Connect upgrade a year ago. Anyone using AutoHotKey, or another scripting program to script this? Each computer will have multiple users who share the same extension. They are crabby when they have to log in to the client with the generic credentials provided to them.
↧
Search

User Voicemail Skipping

January 6, 2020, 9:11 am
$
0
0
Salutations,

Brand new user here. I am having an interesting issue with our Voicemail, and after extensive google searching I have been unable to find anyone having similar. Google did lead me to these forums though so I hope to lean on you fine folks from time to time!

Bit of background, I have inherited an aging Shoretel system with about 50 IP phones of various G and non G models running a Shoregear T1 and an SG-90. I have very little PBX experience and if I am honest, insufficient networking knowledge for what I have been placed in charge of. Recently in an attempt to increase throughput in the office for lower cost I tested a Grandstream GXP2140 on the system using SIP protocols. I had hoped to replace our IP115 and 230's with a cheap G phone. After getting it working for my test user, I realized we had no SIP licenses and I abandoned the project as economically inefficient. Since then we have been having a strange issue where randomly some users experience problems with their voicemail menu skipping ahead. It will begin listing off the options such as "press 1 to hear message, press 2 to ...", but after the attendant says 2 or 3 words she skips to the next option. "Welcome to Shor..., Press 1 to..., Press 2 to..., Press 3..., Press 4..." Users are unable to access voicemail since it does not accept any input while this is happening. After anywhere from 5 minutes to 20 minutes the issue will cease and the user will be able to get their voicemail. Users appear to be randomly affected and this does not affect their ability to place or receive calls or transfers.

I have rebooted the appliances and our resource usage is under 20%. No errors in the logs to indicate an issue. I have gone back an reversed the SIP settings I changed (that I could remember) in case something I did was the cause, but it did not resolve.

Does anyone have any thoughts on whats causing this and how to resolve it? Thanks for your time!
↧
↧

Silent Monitoring Restrictiona

January 3, 2020, 2:13 pm
$
0
0
I am looking for a way to restrict silent monitoring for select extensions. Where I do NOT want certain extensions to be able to be monitored by anyone. Is there a way to accomplish this?
↧

Call delay at beginning of call on 400 series phones onf the Edge Gateway

January 7, 2020, 12:59 pm
$
0
0
Has anybody found a workaround for this, other than replacing all the phones?

https://oneview.mitel.com/s/article/...-series-phones

"Group call events (Hunt Group or WorkGroup calls) that route to a 400-series phone (which is connected to an Edge Gateway) that is NOT a 485g phone may experience a delay(1-4 sec) after the call is established before audio is heard. Additionally, you may experience garbled audio at the beginning of the call during the silence period.

This is a current limitation with the low end 400-series phones when connected through Edge Gateway. If this delay is unacceptable with EGW, please utilize 485G phones instead of the 420 or 480/480g phones."
↧

Etherspeak Ringleader SIP trunking issue with hairpin=1

January 7, 2020, 5:58 pm
$
0
0
Etherspeak has this monstrosity of a SIP setup where the SIP signaling goes over a VPN tunnel and the RTP traffic goes over the public internet.

When the SIP profile for the SIP trunk group has hairpin=1, certain call scenarios lose audio (like call forward always). I can't for the life of me find these missing RTP streams in packet captures. Has anyone worked with this carrier and successfully gotten it to work with hairpinning allowed??
↧

MItel Icon's not Syncing

January 8, 2020, 9:16 am
$
0
0
Greetings,

I have a user who is in a high level management position having a strange error with the Mitel Communicator. I already tried reinstalling the software but figure there is not much that can be done on this. Below is an overview on what is happening. If you have any ideas on things I can do/try please let me know.

Issue: The user has two screens; Monitor 1 and Monitor 2. Monitor 1’s and Monitor 2’s Mitel icon are not syncing. This causes one icon to show orange and the other one not to show that it is orange. It also shows different numbers on each screen. They get notification (popups) from windows but they said they need this functionality working to properly work.

When: This started ever since Mitel was installed on their computer

Why: The user looks at primarily 1 monitor (the one that isn’t flashing). They say that they are missing messages/calls because it isn’t notifying them when someone contacts them.

Monitor 1
[IMG]file:///C:%5CUsers%5Cpoppm%5CAppData%5CLocal%5CTemp%5Cmsoh tmlclip1%5C01%5Cclip_image002.jpg[/IMG]Monitor 1.png
[IMG]file:///C:%5CUsers%5Cpoppm%5CAppData%5CLocal%5CTemp%5Cmsoh tmlclip1%5C01%5Cclip_image002.jpg[/IMG]
Monitor 2
Monitor 2.png
↧
↧

SA100 Mitel

January 8, 2020, 2:36 pm
$
0
0
Version Mitel Connect.

When I login to our SA100 I no longer have the ability to create or edit my conferences. Has anyone run into this? I have tried 4 browsers and a couple of different users, still the same results. I am also looking for the url to create or edit a conference, just to see if I can do it that way.

I know http://xxxxxxxxxxx/index.php?page=MEETINGS displays this page.

The server has been rebooted and re-"burned". Thanks for any help you may be able to provide.

Paul

SA100.png

↧

BRI no Ringback tone

January 15, 2020, 3:59 pm
$
0
0
Hi,

Australia is gradually withdrawing ISDN services and we will soon lose our current PRI circuit.

In anticipation of this, we have recently installed a SIP service with a SIP/BRI converter.

Everything works fine apart from the ringback / call progress tone (when dialling out we don't hear a ringing or busy tone).

Our carrier has looked at the ISDN log I provided, and tell me that the ISDN convertor is sending the progress tone to the PBX.

Does anyone have any thoughts on what is not correctly configured?

Many thanks, Ian

Our BRI ccts connected to the SIP to ISDN convertor (OneAccess) are configured as:

Protocol type:
ISDN User
Central Office Type: Euro ISDN
Signalling: Point to Point
Clock Source: Slave / Medium

ISDN Profile is the default: SystemISDNTrunk




↧
Search

LLDP / QoS / DSCP Settings

January 20, 2020, 1:59 pm
$
0
0
I've been tasked with reviewing and fixing QoS for our ShoreTel phone environment. From my reading I have come across a LOT of material, mostly related to the network itself and not the phones.

However, I wanted to first start by determining exactly what has been setup on the phones regarding QoS. Our environment is as follows:
  • Mostly IP480 phones
  • Each phone has a dedicated ethernet drop and switch port (No PCs are connect to any phone) and the switch ports are configured to be untagged
  • As such, we do not specify layer2tagging or a vlanid in option 156 of our DHCP server (It's nice having the phone only boot once into the correct VLAN)
Questions:
  1. I see that some DSCP options are set in Administration > Features > Call Control > Options. These include three different DiffServ values (which we use the defaults). How, exactly, do these values get sent to the phones? Are these pulled when the phone boots and connects to the tftp server? Are there any other conditionals required to cause these values to be applied or not applied?
  2. How do the values in the area mentioned above actually work? Is it that the phone downloads them, then sends them to the switch when the switch checks LLDP?
  3. Are there any other QoS settings handled in the phone settings, or anywhere else, that I should be aware of?
I'm trying to have 100% understanding of the phones and phone system so I can then move on to how the switches handle QoS.

Any input or high-level overview is greatly appreciated! Thanks in advance.
↧

SIP Codec List

January 21, 2020, 10:49 am
$
0
0
Have an issue with Codecs being sent in SDP on a SIP trunk. Site codec for inter and intra both contain only G729 and G711. However, in SIP calls the SDP invite we see a whole lot more-- BV32/16000, BV16/8000, G729, PCMU/8000, PCMA/8000. Opened a ticket with TAC and the tech tried telling me that "Not necessarily, it might send any codecs for negotiation." . I asked what is the point of having a codec list if it just sends "any" codec for negotiation? He responds with "To set the priority if multiple codecs are there.",, this again makes no sense as it is not even doing this. I have pulled pcaps of other systems and what is in the list is what is being sent out. This tech even went to confirm what he said with his team lead and said they agreed.
Am I missing something here??
↧
↧

Mitel Connect Client Port 5448 TLS 1 Disable

January 23, 2020, 7:17 am
$
0
0
Hello, We are running version Build 22.10.7600.0 of Shoretel/Mitel Connect Director on-site. Our connect client software appears to be connecting to our Windows 2016 server at port 5448 and is showing CAS. How do we disable TLS 1.0 on this port 5448? We've disabled TLS 1.0 on every other aspect of IIS and the Mitel nginx web config but cannot find anything about disabling TLS 1.0 for port 5448.

Also we need to strengthen the ciphers for this port as it's showing DES as allowed when we scan with nessus.

Thanks


Negotiated cipher suite: AES256-SHA|TLSv1|Kx=RSA|Au=RSA|Enc=AES-CBC(256)|Mac=SHA1

Hosts
5448 / tcp / www




Medium Strength Ciphers (> 64-bit and < 112-bit key, or 3DES) DES-CBC3-SHA Kx=RSA Au=RSA Enc=3DES-CBC(168) Mac=SHA1 The fields above are : {OpenSSL ciphername} Kx={key exchange} Au={authentication} Enc={symmetric encryption method} Mac={message authentication code} {export flag}
5448 / tcp / www
↧

Phone time is off by 10 days and 11 hours

January 23, 2020, 12:01 pm
$
0
0
Good morning,

Our phones were off by about four minutes so I looked on this forum to see what I needed to do to fix that. Per a post I read here I added ShoreTel Option 4 to my DHCP server and pointed it to my Domain time server. After power-cycling the phone I'm now off by quite a bit more than the original four minutes. I removed the Option 4 but after power-cycling the phone I'm still looking at the horrendously wrong time. I should probably say too that my ShoreTel server has the correct time on it and all of my computers and servers have the correct time on them. How can I correct this issue?

I now have a different perspective on the phrase "Leave well enough alone."

Joe B
↧

How to Turn your Mitel Server into an NTP Time Server

January 23, 2020, 1:03 pm
$
0
0
Click Start, click Run, type regedit, and then click OK.

Locate and then click the following registry entry: HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Servic es\W32Time\Config\
  • In the right pane, right-click AnnounceFlags, and then click Modify.
  • In the Edit DWORD Value dialog box, under Value data, type 5, and then click OK.
ntp key.JPG

Enable NTPServer.
  • Locate the following registry subkey: HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Servic es\W32Time\TimeProviders\NtpServer\
  • In the right pane, right-click Enabled, and then click Modify.
  • In the Edit DWORD Value dialog box, type 1 under Value data, and then click OK.
​​​​​​​ ntp key2.JPG


Exit Registry Editor.

open the command prompt as an admin, type the following command to restart the Windows Time service, and then press ENTER
net stop w32time
net start w32time ntp cmd.JPG
​​​​​​​
↧
Viewing all 1780 articles
Browse latest View live
Search
<script src="https://jsc.adskeeper.com/r/s/rssing.com.1596347.js" async> </script>