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IP480 Endless Ringing

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Several times a day, a few of our IP480 phones will ring non-stop where voicemail does not pickup. If you pickup the phone, it is just dead air. We are on 19.41.5003.0 . Most calls will go to voicemail, but some simply hang there and ring and ring and ring.

Any ideas?

Factory Reset ShoreGear 90 and T1k

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I used to have a director server along with a T1k and 90. The server is long gone. I'm going to be getting rid of the 90 and T1k soon, but I want to factory reset them first. I'm really struggling with this. I've been able to get into each unit via the management console, but I don't see the option to factory reset. I can change the ip info (ip, mask, gateway) along with server ip, but that's it. The 90 keeps spewing out messages like this:
ERROR: MGCP_SendToNetwork(, 67, <xxx.xxx.xxx.xxx>:2427) errno 65: errno = 0x41
from MGCP_TRANS_ProcessTimeOut 6

I replaced the IP with x's before I posted. This was the IP of an old phone. This message comes up multiple times for all the different phones that used to communicate with it. My biggest question is: if this unit still remembers the old ip addresses, what other information does it still have stored on it?

thanks in advance.

cloud email not deleting

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client uses terminal server in the cloud when they delete messages in outlook does not delete out of inbox
if they use there desktop works fine any ideas using 14.2 19.45.1602 .0

PRI fail over issues

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I'm going to be calling our support vendor tomorrow, but I wanted to reach out to you for any insights. For starters, we're running Shoretel 12.2. We have 20 branch offices and two PRIs. The HQ has a PRI and so does another branch (Branch A). Most of our branches use standards POTS lines; a few others have switched away and are now using the PRIs. All users have access to their local trunks and the PRIs. HQ PRI and Branch A PRI have different area codes.

Yesterday, our HQ PRI went down. This is the first time this has happened to use since we started our POTS lines reduction. Since the HQ PRI was down, we figured the branches that were set to use this PRI (HQ and one other branch, Branch B) would simply use the other PRI. This didn't exactly happen. When users at HQ and Branch B tried to make local calls, the calls would never be placed and users would get the reorder tone. When viewing the VBTrunk tool, we never even saw the calls hit the PRI at Branch A. However, the users could place calls that had a different area code and they went out the Branch A PRI without issue. Only calls with the local area code wouldn't go through. HQ PRI and Branch B both share the same area code.

This evening, we decided to do some testing. We brought down the Branch A PRI (and left the HQ PRI up and functional) and had the users that use that PRI make a bunch of test calls. They called a bunch of phone numbers local to that PRI and they all went though the HQ PRI without issue. We had them successfully call long distance numbers as well. When we had them call the HQ PRI's main phone number, all of them were prompted to enter the area code to complete the call.

So we repeated this test with the HQ PRI. We brought down the HQ PRI and brought the Branch A PRI back up. We had users at HQ and Branch B place local calls, and they simply would not go through. They got the same reorder tone from the outage we had yesterday. When we tried long distance calls. those calls went out the Branch A PRI without issue.

So my question is--what exactly is going on? For our first test, I'm going to assume the caller ID may have something to do with the area code requirement. Since we have a local prefix list for the HQ PRI, the area code is probably being stripped out when Shoretel dials it. When the call hits the provider, it may see the caller ID and expect an area code to be dialed since the area codes do not match.

But, I'm at a loss. And I don't know where to even start with this. Can any shed some light on what is going on? Thanks.

Consult transfer to 930D on SIP trunks fails

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Can anybody test this scenario for me?
I have a couple of places with this problem and I am trying to get to the bottom of it.
Incoming call from a SIP trunk answers fine to any type of phone.
If you do a consultative transfer to a 930D extension then there is no audio heard from the 930D to the external caller.
If you do a blind transfer then it's OK.
If you do a consultative transfer to any other kind of handset then there is two way audio when the transfer is completed.

My setup is ShoreTel 14.2 with an Audiocodes M500 terminating on a 1/2 width SG with ports configured using SIP with Media Proxy.

Issues calling SA Conf Bridge from LifeSize SIP Ext

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Can anyone confirm that they're have this issue or not?

On ShoreTel 14.2 19.45.1602.0 and 2x LifeSize Express 220's on 5.0.4(8)

We can call any IP phone perfectly fine, including Auto-Attendants.

We are moving off of the old Converged Conferencing Server and onto the SA-100

Calling either the WG Ext on the Converged Conferencing Server or the Global Conf Ext on the SA-100 results in no audio at all.

I ran a pcap on the Express interface and am seeing from the Express to the ShoreTel switch after the normal messages for setups etc. a 400 Unsupported Method In Refer-To Header message come up then a BYE.

I have tickets open with ShoreTel and LifeSize just curious if someone else can reproduce it or has it working.


Call behavior of Consultative Transfer

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This problem seems to have begun effecting the customer after they upgraded from Shoretel 13.2 recently. They are now on 14.2 and whenever they go to do a consultative transfer and press no thanks or decline it the caller is put on hold, instead of being brought back on the line. The customer does not put them on hold before she transfers. I am unable to find a setting that adjust this behavior and tested this on another system running 14.2 and the user is brought back on the line when I cancel the transfer. This user has an operators license and drags and drops the person in communicator to do this, on the client version, tried the web version as well and it has the same behavior. Also tried it on another user with a personal license in a different user group and still got the same result. Any ideas?

Contact Center Director Login

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Hi

I am the only administrator for our ECC server and I forgot my password for the Contact Center Director Login. Also tried the default "ea\e1996" and it doesn't go through. Is there a way to reset the password for my administrator account? If not, what are my options?

Workgroup Redial

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Hey, guys. I recently setup a new workgroup however agents are unable to use the redial function on their phones. When they hit redial they only see missed calls from an extension equal to the workgroup extension e.g. if a callers calls from 123-321-5678 the redial screen will show 5545 or their workgroup extension. The ring pattern is set to simultaneous. Any ideas??

Dial Lists

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Hello everyone, I have a question about Dial lists, I was using the ECC 7 Admin Guide to get this far, basically I create a DB, uploaded an excel spreadsheet into it to test, the spreadsheet only has 4 columns:
Name Phone1 Phone2 Phone3
and 2 rows with test info.

I created the Connection and tested it with a basic query and it returned that there was 2 rows so I assume that its connected properly:
SELECT * FROM [OutboundCalls].[dbo].[Sheet1$]

gave me: Test Query Result
Number of records received: 2 For Query: select TOP 50 * from [outboundcalls].[dbo].[sheet1$] where Name > '' order by Name

Now however I am a bit stuck and unsure what to do next. I'm not a SQL guy so I am having trouble figuring out what type of SQL query I need to use. I can also change my database to something that makes more sense but I was just testing. Does anyone have some example they can share to give me and idea of what I am supposed to be doing with the SQL Query - Import Dailing Information box. This box is basically pulling in the number to dial and pulling the data into the pop-up box for the agent right?

I assume after I get the Query right i just turn on a schedule for the list, then log into my trianing serive/queue and I can test it.

IP485G phone cannot find config files

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Hi,

I wonder if someone could help me get some IP485G phones working please?
We have a system using version: 11.2 / Build 16.42.1850.0 It is BRI / ISDN based with around 12 extension and I would normally configure phones manually.
It currently has a variety of phones including several IP 655 devices.

I was expecting the IP485G config to be easy – sadly not.
The 485 phone I am working shows an error on services. I have set the config server as the same IP address as the FTP server in the config of the other phones. The errors I am seeing are:

COULD NOT FIND CONFIG FILES
PP Phone event 2702. Could not find country, family, model, or mac level config files.

NO SIP SERVER FOUND IN CONFIG
PPhoneEVent 2702 No SIP Server found in config. No Sip server was found in the local config file and no config files were found to download.

Any thought?

Many thanks, Ian

Call Recorder not recording.

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I've recently installed the ShoreTel recorder software on my HQ server. It seems like everything is set up properly I'm getting the error that says
"ERROR: SYSTEM: Call 67016: ST 210: Call 65820: Failed trying to waveRecordStartEx: cwrRecordFailed"

I'm not sure what to do. I'm certain that I've set up my class of service properly to allow the recorder surrogate user to record the users calls and I've configured the users' class of service to allow their calls to be recorded by the surrogate.

Any ideas of what this could be?
Also, I've attached a snippet from the log file.

Is there a code to jump the queue for workgroups?

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User A, B, C and D are in a workgroup in the order A, B, C, D. User A B and C are in a discussion but have not logged out of the queue. Is there a number or * code that user D can press to instantly pick up a call that comes through, rather than letting the phone ring through A, B and C first?

Thanks
Michael

Make number private

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We have lots of users have "Make Number Private" checked under their account. Do someone know where to check in SQL database to find out which users has this field checked.

Thanks

Voicemail Issue

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Has anyone heard about following issue:

two users who sometimes get a message when trying to check there voicemail "Your mailbox is being used by another user" .


max call duration

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I have a user that has been on a training call the last 2 days and when he has been on the call for 8 hours the call disconnects
where can I set that value higher

How can I dial 933 trhough ShoreTel system

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One of our client; a school district has change the service provider. The school district has some 700 DID numbers. Each DID number has particular address attached to it. The school district wants to test random 911 calls to see if 911 dispatcher gets correct address. The service provider has a testing number 933 to test 911 calls. Can someone tell me how can I manipulate Shoretel dial plan so that I can dial 933.

Thanks

Dialing issue

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Dialing a certain local number in any fashion from one of our sites results in a pause for a fraction of a second, then a dial tone. Adding the prefix to the trunk group prefix list results in a tone and message stating the call can't be completed as dialed. Playing around with the trunk group settings, I found that if I unchecked "Remove leading 1 for Local Area Codes" and keep the prefix in the list, then the call can go through as dialed. However, this fix obviously causes dialing other numbers to fail. This is not a problem at other sites/other trunk groups. Is there a fix for this without affecting numbers that are already able to be dialed? Thank you!

TMS not connecting to remote switches

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We have recently reconfigured our system. we have 4 locations. Previously they each had an internet connection, and all of the Shoretel stuff was connected by VPN.
We consolidated all of our internet connections to one office and the other sites are connected by point to points.

After some trial and error (mostly error) I have everything working phones work and auto attendant. All sites can pass traffic and communicate with each other.

The one issue that I can't seem to resolve is that the switches at the remote sites are not connecting correctly with the home server. When you go to the switch connectivity page it shows that "TMS can't connect to the switch". the switches and server seem to be able to talk back and forth ok, so I don't know what the issue is.

I have logged into the switch and am getting the following error messages.

NEC_CONNECT from 192.168.10.151:4522, prior was 192.168.10.151:4492, connect new sock 117, up failed
tmsd_svc_tms_connection_down(EXT, "expired_timers THU JAN 01 00:04:59 1970", -1): conn 1, rp# 0
NEC_RELEASE_CCO, from external TMS, ncc_fd= 109
tmsd_svc_tms_connection_down(EXT, "tmsd_svc_run (tmsd_rpc_error: 1) returned", 1): conn 0, rp# 0
ERROR: src=TMSD, eventid=0xc100006c , str=(TMSD: error in function tmsd_svc_tms_connection_up(), failed to create rpc client)
NEC_CONNECT from 192.168.10.151:4557, prior was 192.168.10.151:4522, connect new sock 117, up failed
tmsd_svc_tms_connection_down(EXT, "expired_timers THU JAN 01 00:06:05 1970", -1): conn 1, rp# 0
NEC_RELEASE_CCO, from external TMS, ncc_fd= 109
tmsd_svc_tms_connection_down(EXT, "tmsd_svc_run (tmsd_rpc_error: 1) returned", 1): conn 0, rp# 0
ERROR: src=TMSD, eventid=0xc100006c , str=(TMSD: error in function tmsd_svc_tms_connection_up(), failed to create rpc client)


Any thoughts on the issue would be appreciated.

Scott

SA-100 Virtualization

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Can I use the ST14.2 dowloadable OVA file to install a virtual SA-100 for a site that is on release 13.0?
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