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ShoreTel IP560g No MGC IP Configured

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Hey guys, I'm a ShoreTel noob and found a spare IP560g in my inherited office. One of the IP560g's we have in use is acting a bit strange -- our receptionist is reporting it completely shutting off at times and sometimes being on but not ringing. This phone has had an issue of getting hung up in wrap-up mode rather often, so I figured I would replace it with this extra 560g I have. However, it is telling me there is no MGC IP configured. Shouldn't the phone automatically download the configuration after I manually enter in the FTP server?

I got the phone to get DHCP IP and I assigned the phone to myself to the point that everything looked good, but if I call my DID it goes straight to voicemail.

Any tips appreciated!

Not able to make 800/877/844 calls

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I've got strange issue, Users are experiencing issue with making calls out via 1-877 or 1-844 numbers….i’ve looked on shoretel director and don’t see anything down..

I have attached pic of status of softswitch and server and they are are both in service, though I notice the softswitch has no ip phones, sip trunks or sip proxy connections...

Can't Receive phone calls if PRI 1 is in use on an Out Bound call

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So we have gotten several reports that users are not able to call us. Some times the call connects, some times it just rings and rings.
I have tracked the issue down to the following:

It seems when ever PRI 1 is in use for an outgoing call, no incoming calls can complete. I happened to test calling in today, while trying to fix our work groups, and could not get the call to connect.I looked and it looked like PRI 1 was in use ( VBTrunk Screenshot ), it just rang and rang, finally I got a dial tone, and hung up. VBTrunk showed nothing as an incoming call. I had a coworker test same result.

Any ideas as to why this occurs?

Again, I am still new to using Shoretel, and I may have some of my terminology wrong so please pardon my mistakes, and feel free to correct me.


Pre-announcements

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I have one site that has about 40 DIDs going to about 40 Workgroups. It appears that almost all the DIDs have a DNIS entry that goes to a specific Simultaneous ringing Workgroup. In most cases, each workgroup has the same people in the group. Unanswered calls go to another workgroup.
This customer now wants an outgoing message that pickups up for all DIDs that says that all calls are being recorded. I don't see any other way except to create 40 different Auto Attendants and then change all the DNIS entries to go to the AA before rining the Workgroup.
Could there be another option?

Training class recommendations? (SIP / SBC)

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Hello,

Can anyone recommend some good training class resources for working/troubleshooting with SIP/H.323 and SBC devices?
For someone who has fairly decent server and TCP/IP network background, that is.

Many thanks in advance,

How to change the max MTU in Mitel Connect (1500 is The Default MTU)

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Please be aware these settings are for informational purposes only and change these values at your own risk. The most common reason to change the MTU is the inability to dial between sites. Often site to site connections will restrict the MTU size to reduce overhead. By default the Mitel MTU auto discovery would correct your MTU value if it is detected correctly. However, sometimes the ICMP request can be blocked causing an MTU value above what is possible.

For reference: Mitel Connect has a default MTU of 1500 and it is adjust upon discovery of a lower value if possible.


OBJECTIVE:
  • Step 1 Determine site MTU size
  • Step 2 Check switch MTU
  • Step 3 Change the global MTU value
  • Step 4 Verify MTU changed
Step1: Determining Site MTU
To quickly assess a WAN path’s PMTU, execute a series of ping tests using the following command at a Window’s CDM prompt from the remote end of the WAN tunnel connection; ping <HQ server IP> -f -l XXXX (e.g. ping <HQ server IP> -f -l 1400), where XXXX is the packet size.
ping string.png
Begin increasing or decreasing the packet size from this number in small increments until you find the largest allowable size that does not fragment and successfully pings.

Examples:
1500 Fail
mtu 1500 fail.png

1400 Fail
mtu 1400 fail.png


1393 Fail
mtu 1393 fail.png


1392 Pass (This is our MAX MTU)
mtu 1392 success.png



Step 2: Checking our switch MTU
Now that we know our site MTU can be no more than 1392 let’s check one of our switches for the MTU value.



From the HQ server open CMD Then Type:
Ipbxctl –telneton <switch IP>
telnet <switch IP>
........Login: anonymous
........Password: ShoreTel

Then type gotoshell
cmd gotoshell.png


Once you see the ->
type ifShow (S must be capitalized)

Now look for the Maximum Transfer Unit size is XXXX
This number is the max MTU for the switch
gotoshell ifShow.png
If you leave this session open during step 3 you will see a notice stating the MTU has been changed.



Step 3: Changing the max MTU
Directly Login into the server and then open the registry aka regedit.exe

From within the registry editor navigate to the following key and edit it:
HKEY_LOCAL_MACHINE\SOFTWARE\Wow6432Node\Shoreline Teleworks\Telephony Management Server\Settings Key Name= SwitchDebug

Add this parameter and the appropriate value to the end of the string ether_mtu_size max mtu here. Use your max value that you found from our previous ping tests.

Default Value= debug_options onhook_dialing_for_IP110 1.
New Value Example= debug_options onhook_dialing_for_IP110 1 ether_mtu_size 1472.

regedit mtu value.png
Saving this key instantly changes the MTU value globally.



Step 4: Verify the MTU Value
Repeat step 2 to see if the value has changed. Also note if you never closed the ipbxctl session from step 3 you would see this notice stating the change.
site mtu change verify.png

3 digit extensions result in dead air calls

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We have observed an interesting issue arise where many of our users will dial our internal 3-digit extensions to only hear dead air when the call is connected. However, if one dials the full extension number (for example 806-350-5570 instead of 570) The call connects fine.
I believe it has something to do with internal vs. external, but we are officially stumped as to what it could be?
Issue has become more pronounced in the last year.

Anyone familiar with using *72 and *73 to redirect calls?

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I have a system I took over from someone else. They have an extension that calls are directed to but is never manned. They use *72 to move the calls to a hunt group and *73 to enable the extension to take calls as the first line. I can not find any documentation on this anywhere. The user merely goes off-hook, dials *72 and it's done. I'm just looking for the setup so that I don't get blindsided down the road when it's not working or needs to be configured again.

SG-50 Switch

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I'm setting up new location, in nearby neighbor state. I have 1gig internet service coming online, but thinking I don't need to order separate phone lines since this is Voip phone system. I will have VPN connecting remote plant site back to corporate IA location.

I will have 5 new users with shoretel phone extensions at new location. .

I was going to order a shoretel SG-50 switch but don't think I'll need this since all 5 lines(have spare licenses with on-prem shoretel system to assign) out of state will just piggyback and will still have 319 area code and not a local out of state phone number.

I would only need this SG-50 switch if I wanted the local area code numbers, right??

Anyone calling someone out state, this call would still be relay through internet and through on-prem shoretel switches system here at corporate ia location?

Live answering

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Hello,

I'm currently on version 19.49.5200.0

I have a project to set up all the executives in my company with live answering, meaning that their line will bounce to the next person until someone live answers the call. Currently, the previous IT employee set up one of the executives with two workgroups and they are pointing to each other creating an endless loop until someone answers.

Now this does not sound like a good solution, especially adding a large number of people in this kind of setup.

I'm wondering if anyone's had experience in setting up this particular kind configuration if you can offer some advice.

Let me know if you have any questions

Tag Connect Client with Voice VLAN

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Hello everyone!

I would like to get some "Best Practice" advice when it comes to setting up the Mitel Connect client on Windows 10.

Usually, when a physical phone is present, it's daisy chained and tagged on the correct VLAN. With no physical phone, all virtual, the Connect client pulls a DHCP from the wired or wireless VLAN. I would like to know if I should keep it like this and just add the wired network to the "IP Phone Address Map" in the Director or do it another way.

Thanks.

What version of ShoreTel Connect Director is build 21.84.5543.0

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Hello all,

We are trying to figure out what version of director we are using to make sure it isn't 14.2. I believe this build, 21.84.5543.0, is newer than 14.2, but I can't find any documentation. Is there a handy KB article or anything that correlates build numbers to version numbers? Or is such a thing even possible?

Thanks!

Some users unable to dial local numbers

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I'm relatively new to ShoreTel system

I have a few users that are unable to dial a local number and from what I can dell on director both accounts are set up the same. Anyone know what could be causing this issue?

AutoMatic handover not working

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Hi Chaps, I hope you are all well. I have an issue where on my Iphone i have the "Automotic Handover" selected but this simply does not work. If the wifi signal dereases or if i disconnect the wifi the client always asks to call the access number. Is there a way to make this seamless?

Tag Connect Client with Voice VLAN

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Hello everyone!

I would like to get some "Best Practice" advice when it comes to setting up the Mitel Connect client on Windows 10.

Usually, when a physical phone is present, it's daisy chained and tagged on the correct VLAN. With no physical phone, all virtual, the Connect client pulls a DHCP from the wired or wireless VLAN. I would like to know if I should keep it like this and just add the wired network to the "IP Phone Address Map" in the Director or do it another way.

Thanks.

AD Import tool

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I have the application installed along with the licensing server. I'm looking to update our LDAP with our users DIDs, I know its typically used to update ShoreTel directory from AD but I'm looking to update AD. Is this possible?

vMini Demo Kit

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I'm looking for the custom wav files for our demo kit. My boss wanted ECC deleted off the demo kit never to be used again. Well guess what now he wants it back. I have it all back except for the custom audio files from the director below on the kit. If anyone has them and could make them available to me it would be much appreciated. Thanks in advance.

C:\Program Files (x86)\ShoreTel\ShoreTel Contact Center Server\IVR\0\IVR\Custom IVR\

ShoreTel Web Service SDK - I need Help

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Hi, Hopefully this is the right place to post this. My apologies in advance if it isn't.

I've been tasked with the integration of our ShoreTel system with our custom company CRM.

The goal is ultimately to have a customer call into our ShoreTel system... where their phone number will be looked up and referenced against the CRM... if the number exists it will then be transferred to a specific Route Point.

However, I'm not sure on how to accomplish this since I don't have any documentation or examples on how to use the ShoreTel Web SDK.

I guess my first question is...

Will the Web SDK allow me to accomplish what I'm trying to do?

and, Where can I find documention and/or examples of how to use the SDK?

Any helpful information would be greatly be appreciated.... Thank you.




Auto logout agents end of business day

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Is it possible to have agents logged out of the ECC automaically if they forgot to logout themselves ? For example in the evening at 7 pm or something ?

Thanks in advance

ShoreTel 14.2 -- CDR question

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Hello! Hoping someone can shed some light. I've been asked to pull some CDR data for a user. I typically use a combination of Trunk Activity Detail Reports and User Activity Detail Reports to then dig into the TMSNcc logs for these requests. Sometimes also using the web communicator client to pull routing slips.


I have a weird scenario that I'm unsure about... One of the calls in question does appear on my Trunk Activity Detail Report. It shows an outbound call by the user in question that was 11 seconds in duration and from what I see in the TMSNcc, everything for the associated GUID seems normal... HOWEVER.. this call DOES NOT appear on the same user's User Activity Report for the same date. I can see the call in the web communicator as well.

My question is what is preventing this call from showing up on the User Activity Report?

I had a similar one for this same user and per the TMSNcc, it appears the call didn't actually complete but resulted in a busy tone. I assumed this was the reason that call didn't appear on the User Activity Report. This one though seems like it established properly on both ends so I'm unsure as to why the User Activity Report didn't pick it up... (FWIW, my CDR options are set to *not* display unanswered calls which seemed to fit nicely forthe call with the 'busy' result but not so much for this other one...)

Thanks for any help anyone can provide!
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