Bring Your Own Phone (BYOP) is the ability to supply your own hardware devices for use with Smarsh Hosted Services’s Hosted PBX service. BYOP is available to any SIP enabled telecommunications device that is not listed on Smarsh Hosted Services’s Approved Phone and Equipment list.

Note: A single Anyphone line has a maximum concurrent call capacity of 4 calls. Hardware capacity should also be taken into consideration. This means that 16 line phone will only be able to have 4 concurrent calls, 2 line phone - 2 concurrent calls, 8 line phone - 4 concurrent calls.

Note: Support of TFTP Devices by Smarsh Hosted Services Support is limited to the following:

  • providing SIP credentials
  • verifying registration
  • reviewing call examples
  • troubleshooting basic call features (i.e. making/receiving calls, voicemail, transfer, and hold).

Supported phones include SIP phones with a web user interface (Web UI). In these cases Smarsh Hosted Services will attempt to make the best effort to support basic provisioning and configuration. Support cannot assist with configuring/provisioning a phone in a manner that is not intended to work with our systems including but not limited to:

  • attempting to provision multiple lines/ext to a single device
  • line presence
  • sidecars.

Smarsh Hosted Services is not liable for phones damaged or deemed non-functional as a result of this support.

Activating a phone or SIP device with AnyPhone BYOP

Once you have purchased service from Intermedia to activate your phone(s) or equipment, there are two steps:

  1. Log into the Admin Portal and locate your SIP configuration information
  2. Manually program your SIP credentials into your phone or SIP device

Configuring an AnyPhone Device

  1. Log into CONTROL PANEL
  2. Under the Services section, click on Voice Services
    Voice Services
  3. Select the Numbers & Extensions tab and locate your SIP device by looking for a device type, Extension number or specific DID. AnyPhone devices may be provisioned as either:
  •  Generic SIP Phone – indicates this is a SIP phone with a voicemail box
  •  Generic SIP Paging – indicates this is a paging device without a voicemail box
    Generic SIP Phone

      4. Click on the SIP Configuration tab to locate your SIP credentials

Sip Configuration

Once you have located your device’s credentials,  manually transcribe them into the administrative interface for your SIP device. Different hardware manufacturers use different terminology to describe the same basic SIP credentials:

Smarsh Hosted Services Credential Name Common Manufacturer Equivalent Names
SIP User Name  SIP User ID, SIP ID, Display Name, (Polycom uses ‘Address’)
SIP Authorization ID Auth ID, Authenticate ID, Authorization Name, (Polycom uses ‘User ID’ here)
Password SIP Password, Authentication Password
Domain SIP Server, Registrar Server
Outbound Proxy Proxy Address

 

Save your SIP credentials within your SIP device once you have transcribed them all accurately. You may need to reboot the device to test registration.

Upon successful device registration,  ensure your device is using the same primary and secondary voice codec as what is set in CONTROL PANEL:

  •  G.729
  •  G.711 uLaw

Change either CONTROL PANEL to match your device, or your device to match the CONTROL PANEL primary and secondary codec order. If your device has only one of the codecs above, choose ‘None’ in CONTROL PANEL for the secondary codec.

General Compatibly Information for AnyPhone BYOP

The following tables provides the specific SIP methods and preferences of Smarsh Hosted Services’s VoIP network.

SIP Settings Information

Feature or Setting Recommended Setting or Compatibility
SIP RFC 3261
SIP Authentication All REGISTER and INVITE messages are authenticated
Transport UDP Only
Protocol RTP
RTP Frame Size 20ms
RTCP  Enabled
SIP Record-Route Must be supported by device
Supported Codecs G.729 & G.711 uLaw

General Networking Settings

Feature or Setting Recommended Setting or Compatibility
NAT Traversal 

If device supports keep alive messages such as STUN or binary keep alive:

  • Set registration expiration to 300 seconds
  • Set keep alive interval to 15 seconds

Otherwise, set registration expiration to 30 seconds

Router SIP ALG  Disabled

General Feature Compatibility

Feature or Setting Recommended Setting or Compatibility
DTMF RFC 2833
SIP Transfer RFC 3515 (SIP REFER method)
Call Forwarding 302 Moved Temporarily response
Message Waiting Indicator (MWI) RFC 3842
Paging With Auto Answer on
Un-Parking Uses Remote-Party-ID header
Voicemail  Set device to call own extension or DID

Unsupported Features

Feature or Setting Compatibility
Silence Packets  No, please disable
Comfort Noise No, please disable
G.729 annex b No, please disable

 

Configuring a Cisco Phone via AnyPhone BYOP

The following describes Cisco SPA model SIP phone recommended settings. These settings can be found within the Admin Login web interface for an SPA model SIP Phone.

SPA SIP Phone Setting Location in UI Recommended Setting
NAT Mapping Enable Ext 1, Ext 2, Ext 3, Ext 4 Tabs No
NAT Keep Alive Enable Ext 1, Ext 2, Ext 3, Ext 4 Tabs No
Auth Page Ext 1, Ext 2, Ext 3, Ext 4 Tabs No
Auto Ans Page On Active Call Ext 1, Ext 2, Ext 3, Ext 4 Tabs Yes
SIP 100REL Enable Ext 1, Ext 2, Ext 3, Ext 4 Tabs No (Critical, must be disabled)
G729 annexb Ext 1, Ext 2, Ext 3, Ext 4 Tabs None
Auto Answer Page User tab Yes
Paging Serv User tab  Yes
PSK 1 (Park soft key) User tab  fnc=sd;ext=9000@$PROXY;vid=2;nme=park

 

Configuring a Polycom Phone via AnyPhone BYOP

The following describes Polycom IP and VVX model phone recommended settings. These settings can be found within the Admin web interface for IP and VVX model phones.

IP & VVX Phone Setting Location in UI Recommended Setting
Port Settings -> Lines |Outbound Proxy 6060
Transport  Settings-> Lines |  UDPOnly
Port  Settings -> Lines | Server 1  6060
Transport  Settings -> Lines | Server 1 UDPOnly
Local SIP Port Settings -> SIP 6000 + last 3 digits of phone Ext

Note: if inbound calls on your Polycom phones are displayed as "sip:SIP_User_ID@SIP_Server" or "sip:caller_ID@SIP_Server", complete the following steps:

Configuring a Yealink Desk Phone via AnyPhone BYOP

The settings below have been confirmed to work with Yealink Desk Phones T40P, T41P, T46G and T48P.

Setting name Location in UI Value
Line Active Account -> Register Enabled
SIP Username/Username Account -> Register Unique 9-digit ID located in CONTROL PANEL
SIP Authorization ID/Register Name Account -> Register Unique 9-digit ID located in CONTROL PANEL
SIP Password/Password Account -> Register Unique alpha-numeric password located in CONTROL PANEL
Outbound Proxy/Outbound Proxy Server Account -> Register usbc.telecomsvc.com or UC70.telecomsvc.com
SIP Domain/Server Host Account -> Register usbc.telecomsvc.com or UC70.telecomsvc.com
Keep Alive Type Account -> Advanced Default
Keep Alive Interval (seconds) Account -> Advanced 30
Local SIP Port Account -> Advanced 6xxx (xxx=3-digit ext., e.g. ext. 133 = 6133)
DTMF Type Account -> Advanced RFC2833
DTMF Payload Type Account -> Advanced 101
Voice Mail Account -> Advanced DID of the phone
SIP Registration Retry Timer Account -> Advanced 30
Caller ID Srouce Account -> Advanced RPID-FROM
Voice QoS Network -> Advanced 46
SIP QoS Network -> Advanced 46
Maximum RTP Port Network -> Advanced 50000
Minumum RTP Port Network -> Advanced 30000