ShoreTel Agent Login/Logoff via phones

We have a small number of users that are part of some groups within our ShoreTel System (Sales/Customer Service).

**Note Following only works if you have ShoreTel Workgroup Agent/Supervisor Access Licenses: else ShoreTel Administrator must manually make changes**

Recently there have been more questions on logging in and out as an agent during scheduled times. Below is a quick write up covering major models.

IP 212k:

• Press Options > [Voice Mail Password] > #
• Select "Agent State" via Custom Button
• Select State [Logged In / Logged Out / Wrap-up] via Custom Button
• Press Menu button to exit

IP 230/230g/265/560/560g/565g/:

• Press Options > [Voice Mail Password] > #
• Select "Agent State" via Up/Down Arrow control or press 5
• Select State via Up/Down Arrow control
• Logged In [1]
• Logged Out [2]
• Wrap-Up [3]
• Press Done Softkey

Feature not available on following phones:
IP 110, IP 115, IP 210, IP 655 , IP 8000

For users on non-supported phones Agent status will need to be updated by ShoreTel Administrator.

ShoreTel v13.1 SIP and BIAMP

At the beginning of the year we expanded and built a new set of training/conference rooms.  We employed GrayBow (http://www.graybow.com/) to handle the Audio/Visuals of the room.

One such feature is the use of Crestron, Revolabs for conference phones.  We have had several issues with programming the system to dial, maintain connections longer then 30 some minutes.  GrayBow has been great sending techs out and working with the vendors to resolve the issue (hoping the new firmware release fixes all the issues).  Below documents how we had to configure our ShoreTel System to acknowledge the BIAMP card in the new system.

ShoreWare Director Configuration

---

1. Access ShoreTel Director via Web Browser (http://[DirectorIP]/shorewaredirector/login.asp)
• Default Login: admin / changeme
2. Create a SIP Profile
1. Navigate to Administration > IP Phones > SIP Profiles
2. Click New
1. Name: [We used BIAMP]
2. User Agent: biamp.*
3. Priority: 100
4. System Parameters:
• OptionsPing=0
• SendEarlyMedia=0
• MWI=none
• 1CodecAnswer=1
• StripVideoCodec=0
5. Custom Parameters:
6. Click Save
3. Create a Codec List (We skipped as already configured for other SIP System)
1. Navigate to Administration > Call Control > Codec Lists
2. Click New
1. Name: [We used BIAMP_codecs]
2. Under Choose Codecs Select: AAC_LC/32000
3. Click Add >>
4. Under Codec List Members Select: AAC_LC/32000
5. Click Move Up >> until at the top
6. Click Save
4. Configure Switch Port (Again we skipped as already configured for other SIP System)
1. Navigate to Administration > Platform Hardware.. > Voice Switches.. > Primary
2. Select desired switch name to configure SIP on
3. Determine desired port > Change Port Type to 100 SIP Proxy
• Check your switch manual for more details
4. Update Description for easy Identification later
5. Click Save
5. Configuring Site (Again we skipped as already configured for other SIP System)
1. Navigate to Administration > Sites > [Desired Site Name]
2. Update Following Sections:
1. Bandwidth
1. Admission Control Bandwidth: 1544 kbps 
2. Intra-Site Calls: LS_Codecs
3. Inter-Site Calls: LS_Codecs
4. FAX and Modem Calls: Fax Codecs - High Bandwidth
2. SIP Proxy
1. Virtual IP Address: [Designate static IP for SIP Proxy]
2. Proxy Switch 1: [Select defined switch]
3. Proxy Switch 2: [Select defined switch]
3. Click Save
6. Create SIP User Extension
1. Navigate to Administration > Users > Individual Users
2. Click Go to the right of Add new user at site [desired site name]
3. Complete Following Fields
1. First Name: [RoomNames]
2. Number: [System may auto-populate otherwise enter desired extension]
3. License Type [Select Extension and Mailbox]
• You can come back after completion to set to Extension-Only
4. Primary Phone Port > IP Phones: Any IP Phone
5. Click Save
4. Scroll to Bottom and update following:
1. Client Username: [Keep things simple same as First Name]
2. SIP Password: [Define password / By default nonassigned]
3. Click Save
5. Click Save

ShoreTel v13.1 SIP and LifeSize 220

Recently we had issues with our SIP link failing between our LifeSize220 Video Conference System and our ShoreTel VoIP system.  Below are the steps I used to verify and update settings to recreate the connection:

ShoreWare Director Configuration

---

1. Access ShoreTel Director via Web Browser (http://[DirectorIP]/shorewaredirector/login.asp)
• Default Login: admin / changeme
2. Create a SIP Profile
1. Navigate to Administration > IP Phones > SIP Profiles
2. Click New
1. Name: [We used LifeSize220]
2. User Agent: LifeSize.*
3. Priority: 100
4. System Parameters:
• OptionsPing=0
• SendEarlyMedia=0
• MWI=none
• 1CodecAnswer=1
• StripVideoCodec=0
5. Custom Parameters:
• 1CodecAnswer=0
6. Click Save
3. Create a Codec List
1. Navigate to Administration > Call Control > Codec Lists
2. Click New
1. Name: [We used LS_codecs]
2. Under Choose Codecs Select: AAC_LC/32000
3. Click Add >>
4. Under Codec List Members Select: AAC_LC/32000
5. Click Move Up >> until at the top
6. Click Save
4. Configure Switch Port
1. Navigate to Administration > Platform Hardware.. > Voice Switches.. > Primary
2. Select desired switch name to configure SIP on
3. Determine desired port > Change Port Type to 100 SIP Proxy
• Check your switch manual for more details
4. Update Description for easy Identification later
5. Click Save
5. Configuring Site
1. Navigate to Administration > Sites > [Desired Site Name]
2. Update Following Sections:
1. Bandwidth
1. Admission Control Bandwidth: 1544 kbps 
2. Intra-Site Calls: LS_Codecs
3. Inter-Site Calls: LS_Codecs
4. FAX and Modem Calls: Fax Codecs - High Bandwidth
2. SIP Proxy
1. Virtual IP Address: [Designate static IP for SIP Proxy]
2. Proxy Switch 1: [Select defined switch]
3. Proxy Switch 2: [Select defined switch]
3. Click Save
6. Create SIP User Extension
1. Navigate to Administration > Users > Individual Users
2. Click Go to the right of Add new user at site [desired site name]
3. Complete Following Fields
1. First Name: [We used LifeSize_220]
2. Number: [System may auto-populate otherwise enter desired extension]
3. License Type [Select Extension and Mailbox]
• You can come back after completion to set to Extension-Only
4. Primary Phone Port > IP Phones: Any IP Phone
5. Click Save
4. Scroll to Bottom and update following:
1. Client Username: [Keep things simple same as First Name]
2. SIP Password: [Define password / By default nonassigned]
3. Click Save
5. Click Save

LifeSize 220 Configuration

---

1. Access Administration Interface
• IP (Web Browser)
• Open Browser and enter IP address of LifeSize System
• Enter Password (Default 1234)
• System (Remote Controller)
• System Menu (Blue Button)
• Administrator Preferences  > OK
• Communications > SIP
2. Preferences > Communication > SIP
• SIP: Enabled
• SIP Username: [User's First Name]
• Authorization Name:  [User's Extension #]
• Authorization Password: [defined password]
• SIP Server Type: Auto
• SIP Registration: Direct
• SIP Proxy: Disabled
• SIP Registrar: Enabled
• Registrar Hostname: [SIP Proxy: Virtual IP Address:]
• SIP Signaling: Auto
• UDP Signaling Port: 5060
• TCP Signaling Port: 5060
• TLS Signaling Port: 5061

Validate Registration (Update User License Type)

---

1. Access ShoreTel Director via Web Browser (http://[DirectorIP]/shorewaredirector/login.asp)
• Default Login: admin / changeme
2. Navigate to Administration > Users > Individual Users
3. Click on [User's first name created earlier: LifeSize_220 for us]
4. Verify under Current Port you see something like:
• SIP-[String of characters]
• Congrats phone has registered with ShoreTel if not begin troubleshooting (Physical connections and settings)
5. Change License Type: [Set to Extension Only unless Mailbox is needed]

That completes it.  Hope this helps many out there.