Push Knowledge base: March 2019

Monday, March 18, 2019

ShoreTel: Mobility Trusted Admin APP Set-up

Here are the steps to create the certificates that are used between Mobility and Connect.

Run cd “C:\Program Files (x86)\Shoreline Communications\ShoreWare Director\App\bin” pki.bat -S SMRAdminApp in a Command Prompt window
To generate the certificate for the SMR they are located in the Shoreline Data\keystore\certs directory and copy the contents of the following cert and key files

SMRAdminApp.crt - located in cert folder within the above root directory
SMRAdminApp.key - located in private key folder within above root directory

Complete the following steps to set up trusted server applications for the SMR

Log in to Mitel Connect Director and navigate to System > Security >Trusted Server Application
Click New, and complete the following steps to create a new trusted server application for the

Mobility 9.0 SMR
Specify the Trusted account name. This should be a descriptive name that conveys the location and use of the SMR. This information is for reference only
Browse to Shoreline Data\keystore\certs, and select the SMRAdminApp file
Select Client Application Service in Application Type, and select Enabled
In Property Type, select admin-cas in Available, and then click to move it to Selected
Click Save

Navigate to Configuration > System > Authentication > Directory, and complete the following steps to configure the trusted application settings

Click Add
Select Mitel Directory in Server Type
Specify a Name
Click Apply
Specify the headquarters FQDN or IP address in Server Address
Select Trusted Admin App, and then click the Manage App Certificate link to launch the Directory Server Certificate page
Click Import, and paste the contents of the cert and key files you copied in step 1 of this section
Click Import again, and then cancel the prompt to reboot
Select tls in Security type
Click Apply, and then click Verify

Sync Authenticator Keys on the SMR

Open a browser and navigate to the SMR configuration page with administrator permissions
Navigate to Configuration > System > Authentication > Directory, and select the directory you defined in Configure Trusted Application Settings on the SMR on page 21
Click Sync ABC Keys to sync the authenticator public keys with the headquarters PBX. Mitel recommends you use the Query option to search for a known Mitel directory user name to verify that you can successfully access the Mitel directory

Specify the Authorization Directory Servers

Open a browser and navigate to the SMR configuration page with administrator permissions
Navigate to the Configuration > Groups and Users page
Select the appropriate group, and then select the appropriate directory type and directory in External User Authentication/Authorization
Click Next. Complete configuration as necessary, and then click Apply

ShoreTel: Installing Connect on Server 2016

I have run into a few issues when installing ShoreTel Connect on Server 2016, and the main one is you run into a issue where while installing Connect you get a popup that says "A digitally signed driver is required" and the install fails. This is usually because driver signing and secure boot is enabled. So here is the check list I use when I install Connect on Server 2016.

Prep the new server as per the Build Notes and Install Guide (Check with TacTools Powershell script)
Disable secure boot (BIOS)
Disable Digital driver signing enforcement run ->gpedit.msc User configuration->Administrative Templates->System->Driver Installation->Code signing for drivers set to disable
From Admin CMD run the following command bcdedit.exe -set loadoptions DDISABLE_INTEGRITY_CHECKS
Change DEP to essential windows applications
Disable UAC

HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\Policies\System EnableUA set to 0

Disable Windows firewall
Set Quality Audio experience service to automatic start
Set Simple mail transfer service to automatic start
Check to make sure NO group policies are applied
Check to make sure NO antivirus is installed (disable Defender)
Disable automatic Windows updates
Run ShoreTel compatibility checker

ShoreTel: Order to paste in Certificate keys for ShoreTel Mobility

When importing Certificates into a ShoreTel Mobility router, this is the order you use to paste in the keys, while making sure you do not have any spaces between them and you include the start here and end here line part of the keys.

Paste in Public key, then the Private key, then the Bundle key.

ShoreTel: SIP profile parameters and their usage

DontFwdRefer Usage: DontFwdRefer=[0|1]

When this parameter is set to 1, it inhibits the use of REFER for transfer on the trunk. It also inhibits sending INVITE with Replaces header. Peer must support INVITE without SDP for certain transfer call- flows

SendMacIn911CallSetup Usage: SendMacIn911CallSetup=[0|1]

This parameter is used in conjunction with SIP based emergency gateways, such as those provided by 911 Enable. It appends the MAC address of the IP phone in the From tag of an outgoing emergency call.
From: "Dizzy Gillespie" <sip:+14085551111@10.1.3.55:5060;user=phone>;tag=shorUA_1077733456- 103455277-EPID-001049042E4A

This only applies to ShoreTel IP Phones, excluding the IP-8000 conference room phone

StripVideoCodec Usage: StripVideoCodec=[0|1]

This parameter should be set to 1 if the trunk does not support video properly. When set to 1, it strips video codecs from SDP in INVITE’s being sent to the trunk and properly restores and rejects the video media lines in the 200 response from the trunk. It also strips video codecs from INVITE’s coming from the trunk and properly restores and rejects the video media lines in the 200 response to the trunk

AddG729AnnexB_NO Usage: AddG729AnnexB_NO=[0|1]

This parameter should be set to 1 if the trunk does not support G729 Annex B properly. When this is set, any outgoing INVITE with G729 in the SDP will have the attribute "a=fmtp:18 annexb=no" added to the SDP.

HistoryInfo Usage: HistoryInfo=[none|diversion|history]

This parameter controls how information is presented when an external incoming call is forwarded out this trunk. In this case, the "From" header will indicate the actual caller, which may not be a valid number to present to the trunk. The Diversion or History-Info header will be used to indicate the DID number of the user on who’s behalf the call was forwarded.

If set to 'none' or omitted, then no indication of the forwarding number is presented. If set to 'diversion', the SIP Diversion header is supplied, as dictated by RFC 5806. If set to 'history', the SIP History-Info header is supplied, as dictated by RFC 4244.

EnableP-AssertedIdentity Usage: EnableP-AssertedIdentity=[0|1]

This profile parameter controls how Caller-ID is presented on outbound calls. If it is set to 0 or not pre- sent, then the old style or presenting caller-ID in From header is used when sending outgoing calls. Note that the style of presenting blocked caller-ID has changed in ShoreTel 12.

When set to 1, the Caller-ID is placed in the P-Asserted-Identity header. If privacy is indicated for the call (User dials *67, or trunk group is configured to not send Caller-ID), then a Privacy header is inserted with value “id”, and the From header is anonymous

Port Usage: Port=[5060|1-65535]

This profile parameter changes the remote port used for the SIP trunk. Currently, there is no way to con- figure the port number for SIP trunks in ShoreWare Director. Only port 5060 is supported. This profile parameter allows the port number for a trunk group to be configured

HairPin Usage: HairPin=[0|1]

This profile parameter controls if hairpin is allowed on SIP trunk calls, when enabled and available, features like Barge-in, silent monitoring, whisper-page, whisper-coach, call-record will be supported on the SIP trunks.

OptionsPing Usage: OptionsPing=[0|1]

This profile parameter controls if OPTIONS message should be sent to remote party for detecting connectivity

OptionsPeriod Usage: OptionsPeriod=[180|60-3600]

This profile parameter is used to control the time interval between SIP OPTIONS messages

OverWriteFromUser Usage: OverWriteFromUser=[none|UserID|BTN]

This profile parameter is used to choose either user’s id or billing phone number in the FROM header when making calls

DontAdvertiseUpdate Usage: DontAdvertiseUpdate=[0|1]

This profile parameter is used to decide if UPDATE should be sent in the SUPPORTED header

RFC2543Hold Usage: RFC2543Hold=[0|1]

This profile parameter is used to decide if connection field should be set to 0.0.0.0 in case of sending out- going INVITE for hold

AlwaysSend180 Usage: alwaysSend180=[0|1]

This profile parameter is used to decide if a 180 will be sent out right away after receiving an incoming INVITE

IgnoreEarlyMedia Usage: IgnoreEarlyMedia=[0|1]

This profile parameter is used to decide if early media should be forwarded to the caller, when a SIP de- vice doesn’t wish to accept early media, this parameter should be set to be 1

Register Usage: Register=[0|1]

This profile parameter is used to decide if outgoing REGISTER messages should be sent

RegisterUser Usage: RegisterUser=[BTN|UserID|DID]

This profile parameter is used to decide in what to use in FROM header in the outgoing REGISTER messages

RegisterExpiration Usage: RegisterExpiration=[3600|60-86400]

This profile parameter is used to decide the time interval between outgoing REGISTER messages

1CodecAnswer Usage: 1CodecAnswer=[0|1]

This profile parameter is used to decide if the SDP should contain only 1 codec for an outgoing answer.

SIP Extension Profile Parameters:

1CodecAnswer Usage: 1CodecAnswer=[0|1]

Some devices do not honor the codec order specified in a 200 OK response to an INVITE. This causes several problems. First, some endpoints in the system do not support asymmetric codecs during a session. Second, any bandwidth calculations based on observing the offer/answer exchange will likely be wrong. When set to 1, only 1 audio codec is sent in a 200 OK response.

AddGracePeriod Usage: AddGracePeriod=[0-1800]

Some SIP devices re-register too close to the expiration time, introducing a race condition where the sys- tem is in the process of deleting the record from the system when the re-register is received. This parameter adds a grace period to the expiration received in the REGISTER request.

AllowedCodecs Usage: AllowedCodecs=[any|[codec[,codec]*]

Valid values are ‘any’ (default) or a comma separated list of codec names. The codec name must be for- matted as shown on the Supported Codecs page (Administration, Call Control, Supported Codecs). For example: 'PCMU/8000'. This should be used if the SIP device cannot follow the normal rules of codec negotiation for all codecs supported in the installation. For example, one particular implementation would rejected requests containing some codecs it didn’t understand.
This only applies to audio codecs. Video codecs and RFC 2833 'telephony-event' is not affected by this parameter.

DelayUnregister Usage: DelayUnregister=[0-20]

Some devices, under certain circumstances, un-register, then immediately register again. This introduces a race condition similar to the one discussed in section 0. Usage of this parameter mitigates this problem.

FakeDeclineAsRedirect Usage: FakeDeclineAsRedirect=[0|1|400-606]

Some SIP devices present an option to decline a call. When invoked, various different response codes have been used by various implementations. If set to 0, only a 3xx class response will cause the call to be diverted to the busy destination. If set to 1, 603 will be sent to busy destination as well. If set to a value from 400 to 606, the selected response code will be used to send the call to the busy destination.

MWI Usage: MWI=[none|subscribe|notify]

This parameter defines how RFC 3842 Message Waiting Indication is handled. When set to "subscribe", an explicit subscription is required. If set to "notify", the NOTIFY messages are sent without requiring a SUBSCRIBE. If set to "none", then MWI is not supported.

OptionsPing Usage: OptionsPing=[0|1]

ShoreGear switches can send a periodic OPTIONS message to SIP devices, and mark them Out-Of- Service if they don’t respond. There are 2 benefits to this: Calls are diverted immediately to the busy destination, and there is logging of the event on the server.
The OPTIONS ping occurs periodically between 3 and 4.5 minutes.

OptionsResponse Usage: OptionsResponse=[200-699]

Some devices reject OPTIONS requests, such as with a 405 "Not Supported" response. This can still be used to determine if the device is alive and on the network by using this parameter. Otherwise, a 405 response would put the device Out-Of-Service.

SendEarlyMedia Usage: SendEarlyMedia=[0|1]

When set to 1, the device will be sent 183 response with SDP for certain call-flows. Currently, this is only used in error conditions when an announcement is played.

StripVideoCodec Usage: StripVideoCodec=[0|1]

This parameter should be set to 1 if the device does not support video properly. When set to 1, it strips video codecs from SDP in INVITE’s being sent to the device and properly restores and rejects the video media lines in the 200 response from the device. It also strips video codecs from INVITE’s coming from the device and properly restores and rejects the video media lines in the 200 response to the device.

XferFailureNotSupported Usage: XferFailureNotSupported=[0|1]

For scalability reasons, there are a few call-flows that use REFER as a means for the caller to hear ringback tone. These call-flows rely on the device’s capability to recover from a transfer failure and keep the original call alive. If the device cannot do this, then this parameter should be set to 1, and an alternative means of providing ringback is used.

ShoreTel: TMSNCC log break down and explanation

Here is some info I found on ShoreTels website a while go that breaks down the TMSNCC log file and how to read it. I am putting here so that I can have easy access to it when I am out in the field and i see something that I don't recognize and i need some reference. There is a lot of data in this log that can give you info as to what is happening on a call. I am not covering all of it just part of it as there is just too much info to cover. Again most of this info was taken from a ShoreTel document that I got on their website.

Parameters:
C-CE: Call Create Event- To initiate a call
L-CE: Leg Create Event- Follows a C-CE for every call setup. For internal transfers (blind transfers) C-CE: may be skipped
L-IE: Leg Info Event- Follows a C-CE, L-CE typically. Leg Info provides information on the other parties in the call (peer as well as controller)
C-SE: Call State Event- When call is in progress updates either parties on the state of the call (RingBack, Offering, Established etc)
L-SE: Leg State Event- Typically follows a C-SE to inform the leg state changes
L-DE: Leg Destroy Event- When call is teared down leg is destroyed
C-DE: Call Destroy Event- Call destroyed after user/system hang up
G-MST: Media State Event- For the terminated leg in a call. Every RTP stream (every leg) has media stats

40000001 = Trunk call leg to PSTN
20000023 = Internal call leg
00020000-1aae-4f05-9cce-0010491e1b95 = Call GUID (different for every call)
ncc_media_ctl = PBX system responding to the caller

Example of a Call:

1555-555-5555 Calls in ---> Hunt Group Ext 2110 ---> HG Forward Always ---> Voicemail ---> User Hung Up

07:24:59.656 ( 7616: 4908) C-CE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" ("+15555555555","WIRELESS CALLER",0xC) 00000000,SDP:N,ipCDS:0x00000002,flgs:0x00000000,cd:0x00000000,"7722,TGrp_1,p1,"(704) 940-7722"" "sip:TGrp_1,p1@10.168.98.6:5441"

07:24:59.656 ( 7616: 4908) L-CE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847C(00000000),Req:00000000,1,Flgs:00000000(Null) "sip:TGrp_1,p1@10.168.98.6:5441"

07:24:59.656 ( 7616: 4908) C-CE: 20000023 "00020000-1aae-4f05-9cce-0010491e1b95" ("2110","HQ Main Hunt",0xC) 00000000,SDP:N,ipCDS:0x00000002,flgs:0x00000001,cd:0x00000000,"7722,TGrp_1,p1,"(704) 940-7722"" "sip:2110@10.168.98.6:5441"

07:24:59.656 ( 7616: 4908) L-CE: 20000023 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847E(00000000),Req:00000000,0,Flgs:00000000(Null) "sip:2110@10.168.98.6:5441"

07:24:59.671 ( 7616: 4908) L-IE: 20000023 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847E,rsn:3(Called),1,("+15555555555","WIRELESS CALLER",0xC(NameNumber)),C("","WIRELESS CALLER",0x4(Name)) contact=sip:TGrp_1,p1@10.168.98.6:5441, "sip:2110@10.168.98.6:5441"

07:24:59.671 ( 7616: 4908) L-DE: 20000023 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847E,rsn:13(ForwardAlways),C("2110","HQ Main Hunt",0xC) TrGp=-1 "sip:2110@10.168.98.6:5441"

07:24:59.671 ( 7616: 4908) C-DE: 20000023 "00020000-1aae-4f05-9cce-0010491e1b95" 13,(ForwardAlways) "sip:2110@10.168.98.6:5441"

07:24:59.671 ( 7616: 4908) L-IE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847C,rsn:2(Originate),0,("2110","HQ Main Hunt",0xC(NameNumber)),C("+15555555555","WIRELESS CALLER",0xC(NameNumber)) contact=, "sip:TGrp_1,p1@10.168.98.6:5441"

07:24:59.671 ( 7616: 4908) L-SE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847C,5(Established),0,11:25:00.945 (UTC) "sip:TGrp_1,p1@10.168.98.6:5441"

07:24:59.671 ( 7616: 4944) C-CE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" ("2104","VM-AutoAttendant",0xC) 00000000,SDP:N,ipCDS:0x00000002,flgs:0x00000000,cd:0x00000000,"7722,TGrp_1,p1,"(704) 940-7722"" "sip:2104@10.168.98.5:5441"

07:24:59.671 ( 7616: 4944) L-CE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 010000CC(00000000),Req:00000000,0,Flgs:00000000(Null) "sip:2104@10.168.98.5:5441"

07:24:59.671 ( 7616: 4944) L-IE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 010000CC,rsn:13(ForwardAlways),1,("+15555555555","WIRELESS CALLER",0xC(NameNumber)),C("2110","HQ Main Hunt",0xC(NameNumber)) contact=, "sip:2104@10.168.98.5:5441"

07:24:59.671 ( 7616: 4944) L-SE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 010000CC,3(Ringback),0,11:24:59.672 (UTC) "sip:2104@10.168.98.5:5441"

07:24:59.671 ( 7616: 4908) L-IE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847C,rsn:2(Originate),0,("2104","VM-AutoAttendant",0xC(NameNumber)),C("+15555555555","WIRELESS CALLER",0xC(NameNumber)) contact=, "sip:TGrp_1,p1@10.168.98.6:5441"

07:24:59.671 ( 7616: 4908) C-SE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 3(Ringback),sd:0,11:25:00.951 (UTC) "sip:TGrp_1,p1@10.168.98.6:5441"

07:24:59.671 ( 7616: 4944) C-SE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 2(Offering),sd:0,11:24:59.672 (UTC) "sip:2104@10.168.98.5:5441"

07:24:59.687 ( 7616: 6716) ncc_media_ctl (0x20000005, StopOnDTMF, "0123456789*#", id=0x000000B8, "00020000-1aae-4f05-9cce-0010491e1b95")

07:24:59.718 ( 7616: 6716) ncc_answer_call (0x20000005, 00020000-1aae-4f05-9cce-0010491e1b95, mode=None, id=0x000000B9)

07:24:59.718 ( 7616: 4944) C-SE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 5(Established),sd:0,11:24:59.719 (UTC) "sip:2104@10.168.98.5:5441"

07:24:59.718 ( 7616: 4944) L-SE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 010000CC,5(Established),0,11:24:59.719 (UTC) "sip:2104@10.168.98.5:5441"

07:24:59.718 ( 7616: 6716) ncc_media_ctl2 (0x20000005, StopRecord, id=0x000000BA, "00020000-1aae-4f05-9cce-0010491e1b95")

07:24:59.718 ( 7616: 4908) C-SE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 5(Established),sd:0,11:25:01.003 (UTC) "sip:TGrp_1,p1@10.168.98.6:5441"

07:24:59.718 ( 7616: 4908) L-SE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847C,5(Established),0,11:25:01.003 (UTC) "sip:TGrp_1,p1@10.168.98.6:5441"

07:25:01.109 ( 7616: 6716) ncc_set_call_data (0x20000005, 00020000-1aae-4f05-9cce-0010491e1b95,calldata=0x00000001,calldatamask=0x00000001, id=0x000000BB)

07:25:01.109 ( 7616: 4944) C-DAE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 0x00000001 "sip:2104@10.168.98.5:5441"

07:25:01.109 ( 7616: 6716) ncc_media_ctl2 (0x20000005, StartPlay, id=0x000000BC, "00020000-1aae-4f05-9cce-0010491e1b95")

07:25:01.125 ( 7616: 4908) C-DAE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 0x00000001 "sip:TGrp_1,p1@10.168.98.6:5441"

07:25:08.687 ( 7616: 4944) L-DE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 010000CC,rsn:4(HangUp),C("+15555555555","WIRELESS CALLER",0xC) TrGp=1 "sip:2104@10.168.98.5:5441"

07:25:08.687 ( 7616: 4944) C-DE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 4,(HangUp) "sip:2104@10.168.98.5:5441"

07:25:08.687 ( 7616: 4908) L-DE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 0100847C,rsn:4(HangUp),C("+15555555555","WIRELESS CALLER",0xC) TrGp=-1 "sip:TGrp_1,p1@10.168.98.6:5441"

07:25:08.687 ( 7616: 4908) G-MST: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" ("10.168.98.6","10.168.98.5"),2(ULaw),rsn:1,11:25:01.003 (UTC),pl:20,(s:1, r:378, l:0),(j:0,u:0,o:0) flgs:0x00000000 "sip:TGrp_1,p1@10.168.98.6:5441",vpn:0

07:25:08.687 ( 7616: 4908) C-DE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" 4,(HangUp) "sip:TGrp_1,p1@10.168.98.6:5441"

Here is some other useful information that is found in the logs:

The time of the call this shows up in 24 hour time format

07:25:08.687 ( 7616: 4944) L-DE: 20000005 "00020000-1aae-4f05-9cce-0010491e1b95" 010000CC,rsn:4(HangUp),C("+15555555555","WIRELESS CALLER",0xC) TrGp=1 "sip:2104@10.168.98.5:5441"

This call came in on Trunk Group 1 port 1

07:24:59.656 ( 7616: 4908) C-CE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" ("+15555555555","WIRELESS CALLER",0xC) 00000000,SDP:N,ipCDS:0x00000002,flgs:0x00000000,cd:0x00000000,"7722,TGrp_1,p1,"(704) 940-7722"" "sip:TGrp_1,p1@10.168.98.6:5441"

One of the last lines in the call gives you the G-MST which is the Global Mean Statistics for the call and shows you packets sent (s:), packets recieved (r:), packets lost (l:), jutter (j:), under-runs (u:) and over-runs (o:)

07:25:08.687 ( 7616: 4908) G-MST: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" ("10.168.98.6","10.168.98.5"),2(ULaw),rsn:1,11:25:01.003 (UTC),pl:20,(s:1, r:378, l:0),(j:0,u:0,o:0) flgs:0x00000000 "sip:TGrp_1,p1@10.168.98.6:5441",vpn:0

This line does not only show the calling party's phone caller ID but the phone number also includes the number of digits that are being sent to ShoreTel. This is important as sometimes the carrier makes mistakes and sends to many or not enough digits and the call does not get routed properly. This is where you could look for that info if needed. In this example the carrier is sending us all 11 digits.

07:24:59.656 ( 7616: 4908) C-CE: 40000001 "00020000-1aae-4f05-9cce-0010491e1b95" ("+15555555555","WIRELESS CALLER",0xC) 00000000,SDP:N,ipCDS:0x00000002,flgs:0x00000000,cd:0x00000000,"7722,TGrp_1,p1,"(704) 940-7722"" "sip:TGrp_1,p1@10.168.98.6:5441

This line in the call log shows that the server started to play a greeting, either a voice mail greeting or an auto attendant greeting. This is seen by the StartPlay and then once the caller has recorded their message you will see a StopRecord on the line. There is not one in the example that I can show you.

07:25:01.109 ( 7616: 6716) ncc_media_ctl2 (0x20000005, StartPlay, id=0x000000BC, "00020000-1aae-4f05-9cce-0010491e1b95")

ShoreTel: How to statically assign a ShoreTel phone with IP info

Statically assigning a ShoreTel phone is not something I usually recommend doing but it some situations it is needed. So because of this here are a few steps on how to statically assign a ShoreTel phone

MGCP Phone (IP110, 230, 565, ect)

On a ShoreTel phone push the mute button and they push 73738#
It will ask you to Reset Phone press # for yes
It will then reset the phone
It will then ask for a password enter 1234 then push #
It will say Clear All Values press * for yes
It will ask to turn DHCP on or off press the * key to turn it off
Press the # key for OK
Enter the new IP address use the * key for the decimal and the speaker phone for back space
Press the # key for OK
Enter the subnet mask
Press the # key for OK
Enter the gateway
Press the # key for OK
Enter the FTP IP address (this is the SWD server)
Press the # key for OK
Press # for MGCP
Press # for SNMP
Turn on or off 802.1Q *to turn it on # for OK
Enter VLAN ID
Press # for OK
Ethernet1 auto press # for OK
Ethernet2 auto press # for OK
Country # for OK (to change press mute then enter the country number and then press # for OK)
Language # for OK (to change press mute then enter the language number and then press # for OK)
DNS enter the DNS server IP address
Press # for OK
VPN GWY press # for OK
VPN Port press # for OK
VPN=OFF press # for OK
VPN User Prompt=OFF press # for OK
VPN Password Prompt=OFF press # for OK
802.1x Enable=On press # for OK
LLDP Enable=On press # for OK
Save All Changes press # for Yes
The phone will Reboot and should come up with the statically assigned info

ShoreTel IP 400 Series phone

On a ShoreTel phone push the mute button and they push 73738#
It will ask you to Reset Phone press the Reset soft key
It will then reset the phone
When the phone displays "Press any key to enter setup" press a key
It will then ask for a password enter 1234 then push #
Select Network Policy and press open
Toggle LLDP-MED to OFF
Toggle 801.1Q to on/off
Set VLAN ID
Select Back
Scroll down to Internet Protocol and press open
Toggle DHCP to off
Fill in the IP address use the * key for the decimal and the speaker phone for back space
Fill in the subnet mask
Fill in the gateway
Select Back
Fill in the DNS server
Select Back
Fill in the SNTP (Time) (usally the ShoreTel server IP address)
Select back
Select Back
You should no be on the main menu again
Select Apply
The phone will Reboot and should come up with the statically assigned info.

ShoreTel: DHCP config options need for IP phones

I sometimes forget the DHCP configuration options needed for the ShoreTel IP phones to boot and get the correct info. So I figured I would post it here. If the phones use a different VLAN than your normal data traffic and you are using tagging then I put these options in the Default VLAN as well as the Voice VLAN that the phones will be sitting in. ShoreTel phones use DHVP option 156 to get the IP address of their configuration and FTP servers, it can tell them to enable layer 2 tagging and tell them what VLAN ID to use as well as tell them what country and language to use. Below is a image of how i have configured DHCP on one of our servers. It shows the default data network (192.168.1.0) where the phones first boot into and then the Voice network where the phone VLAN hop over to and will actually reside.

ftpservers=ip_address, country=n, language=n, layer2tagging=n, vlanid=x

ip_address is the IP address of the Headquarters server
n in county=n corresponds to the country number found in the ShoreTel Planning and Install Guide
n in language=n corresponds to the language number found in the ShoreTel Planning and Install Guide
n in layer2tagging=n is 0 (to disable 802.1Q) or 1 (to enable 802.1Q) the default is 0
x in vlanid=x corresponds to a VLAN ID number between 0 and 4094 when 802.1Q is enabled the default is0

A filled out option 156 string looks like this:

configServers=192.168.1.10, ftpservers=192.168.1.10, country=1, language=1, layer2tagging=1, vlanid=10

It is possible to add two FTP servers for option 156, just place a comma between them:

ftpservers=192.168.1.10, 192.168.1.15

Default Data network DHCP

Voice Network DHCP

We sometimes have users that work from home that run voice across their VPN back to the office. So here is what the DHCP configuration looks like on a Cisco router. Here we do not have separate networks for data and voice so it is just a single DHCP scope.

ip dhcp pool LAN
import all
network 192.168.9.0 255.255.255.0
default-router 192.168.9.1
domain-name ABC.local
option 4 ip 192.168.1.5
dns-server 192.168.1.12 8.8.8.8
option 156 ascii "ftpservers=192.168.1.10"
option 66 ip 192.168.1.10
lease 0 8

Sometimes you need to add Option 156 to your DHCP server. Here are the steps to do that.

Open DHCP manager on your DHCP server
Right-click the DHCP server and select set pre-defined options
Click add
Set Name to IP Phone Boot Server
Set Data Type to String
Set Code to 156 and add a description if you like
Navigate to the scope options and add option 156

Thursday, March 14, 2019

ShoreTel: How does ShoreTel decide how to route calls?

By default, ShoreTel will follow the decision tree below, unless custom dial strings are introduced. To make the routing decision, the algorithm poses the following questions. For the number dialed, is there:

A trunk at the originating site for which the call is local?
A trunk at the proxy site for which the call is local?
A trunk at any other site for which the call is local?
A trunk at the originating site for which the call is considered nearby?
A trunk at the proxy site for which the call is considered nearby?
A trunk trunk at any other site for which the call is considered nearby?
A trunk at the originating site designated for long distance?
A trunk at a proxy site designated for long distance?
A trunk at any other site designated for long distance?
Any remaining trunk available at originating site?
Any remaining trunk available at the proxy site?

ShoreTel: Auto Attendant call transfer to voice mail and there is no sound on SIP trunks

I ran in to an interesting issue a while back where we installed SIP trunks for a customer using an Ingate SIParator and hardware SIP trunks terminating on a ShoreTel SG-90 where when a call would come in to an Auto Attendant at the same site as the SIP trunks but then transfer to another site and the call was not answered the caller would not hear the voice mail greeting. This happens because a ShoreTel 24A is intercepting the call and the voice mail server is not receiving it. You can add the registry entry below to resolve this

HKEY_LOCAL_MACHINE -> Software -> WOW6432node -> ShoreLineTeleworks -> TelephonyManagementServer -> Settings -> switchdebug and add dont_use_Media_proxy 1