New SIP Extension/Bulk SIP Extensions
I.New SIP Extension/Define SIP Extension:
SIP Extension
- Number
- Username
- Password
- Codecs
- DTMF Mode
- Progress inband
- Call groups/Pickup groups
- Voicemail MWI
- Call Limit
- Do Not Disturb (DND)
- Inbound Dial Timeout
- NAT
- Qualify
- Always Record
- Email Recordings
- Minimum Size (Bytes)
- Host
- Insecure
- Transport
- RTP Encryption
- Outbound Destinations
- Password
- User Name
- Block External Caller ID
- The External CID number
- External CID name
- Emergency CID number
- Area Code
7.Find me/Follow me Configuration
- Active if Checked
- FMFM Number
- FMFM Dial Method
- Request Confirmation
- FMFM Caller ID
- FMFM Caller ID Num and Name Prefix
- FMFM Dial Timeout
- Unconditional
- On No Answer
- On Extension Busy
- On Extension Offline
- Extension Deletion
- Multiple Delete of Extensions
1. Define Extension
The definition of an extension is comprised of multiple sections. The most important one is the general one, where you can define the internal number for the extension and the password.
Number: The number assigned to an extension for a tenant can be the same number assigned to another extension for another tenant. The Hosted PBX is multi tenant, so each tenant configuration is completely independent from others. This is a general rule and applies on every aspect of the configuration.
Name: The name provided will be used as Caller ID for internal calls. This means the Caller ID on the phone will be overwritten with the one specified here. If you don't want to have the Caller ID enforced to the single configured, but rather you want to use the Caller ID coming from the phone (E.g. an extension that isn’t assigned to a single phone, but to another PBX with multiple extensions) you would check/set the “Trunk” checkbox.
The trunk setting will also affect incoming calls to the phone (or PBX). If the “Trunk” checkbox is set, the SIP INVITE sent to the account will include the number dialed.
Username: Usernames are automatically generated by adding the tenant code to the number provided. The format used by default is using the “-”, but some phones has been found to not accept the dash. The joining character (i.e. the dash) can be changed by clicking on the double arrow button at the end of the box. Keep in mind the fact that the usage of “_” is discouraged and needs to be used only if really needed.
Password: can be auto generated clicking on the “Generate” button. It is highly recommended to use long and completely random passwords to keep your system protected.
Codecs: Every extension can use a broad range of codecs.
DTMF Mode: selectable between auto, info, inband and RFC 2833. Please check the phone configuration and the provider support for choosing the right DTMF setting. The most widely accepted format is RFC 2833.
Progress inband: forces the system to generate ringing tones.
Can reinvite: allows two endpoints, like two phones or the phone and the provider, to exchange the RTP (audio) data directly, without routing through the PBX. Usually if one of the parties is behind NAT, you may experience one way audio. Usually set to No.
Call groups/Pickup groups: defines who is permitted to perform a pick-up for which calls. If call group and pickup group matches, then it is possible to pickup using the specified feature code. Note you still need to define the feature code to use.
Voicemail MWI: allows you to assign the MWI (Message Waiting Indicator) on the phone for a voicemail.
Call Limit: sets the max number of channels a phone can use. Setting it to 1 usually doesn't allow the transferring of calls.
Do Not Disturb (DND): sets the extension in DND mode. This is a server assisted DND. It doesn’t affect the phone DND eventually set (when you’re setting DND on the phone).
Inbound Dial Timeout: sets the time in seconds one extension has to ring before going to the “No Answer” additional destination. You can avoid setting a Dial Timeout value and the default values will be used.
2. NAT Control:
NAT: setting is important when the phone is behind a NAT (which is how most phones are set up). Use force_rport, comedia in almost all the cases. If you experience one way audio, then re-check and tweak the NAT setting.
Qualify: allows the PBX to periodically contact the phone to check to see if it is still online. This is beneficial because it keeps an “open” connection to the connection tracking mappings on firewalls/nat routers that can be between the phones and the Internet. Connections are made every second, but if you have slow phones, you can increase the time to wait for an answer.
Keep Alive:
RTP Keep Alive:
3. Outbound Recording:
Always Record: sets the recording preference for the extensions. If set to “Yes”, all the phone calls made by the extensions are recorded. If set to “Yes, but allows stopping” or “No, but allows starting”, then the recording can be turned off or on by using the predefined #0 and #1 DTMF sequence while on the phone. The recorded file will be available for download in the Status/Call History menu.
Email Recordings: Sets an email address to send the recordings once the call is completed.
Minimum Size (Bytes): Set parameters on the recordings sent to your email. The parameters will allow any recordings bigger than the size set, in Bytes, to be sent. Recording takes place only on bridged channels, so IVR prompts or 'Music On Hold' will not be recorded.
4. Security:
Host: can be “dynamic” (accepting registration from any IP), or it can be assigned to a specific IP address. In this way, no registration is needed.
Insecure: allows the peer to be authenticated using the IP address.
Transport: permits the use of a different transport for SIP signaling. Please note that you can use TCP if you’re having issues with your router or Internet service provider (issues mostly because the use SIP ALG). If TLS is selected, a SSL certificate will be needed on the phones.
RTP Encryption: provides encryption to the RTP (audio) part. The key is transmitted over the SIP channel, so it will be useless to use it without setting the transport to TLS (since the key could be “captured” by an “interested party” if the SIP channel is not encrypted via TLS.
Outbound Destinations: allows you to restrict the numbers the extension can dial. In otherwords, the destination allowed can be restricted. For example, the phone placed in the kitchen of the office can be restricted to place international calls. The Outbound destination can be:
All Allowed: Allowing any number
All Prohibited: The phone cannot place outbound calls
Allowed if matches: The call is allowed if the number dialed match the Regex associated
Prohibited if matches: The call is prohibited if the number dialed match the Regex associated
5. Web User Panel:
Enabling the web user panel permits to login to the web interface providing the extension username and the provided web user panel password.
Password: It is not possible to use the SIP password to login. It is highly encouraged to use a generated password to better protect your system
User Profile: defines the user profile to assign to the user connecting to the web user panel.
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6.Outbound calls:
This section allows you to configure how a call is managed when dialing out the local virtualPBX.
Block External Caller ID: Selecting the “Block External Caller ID” checkbox can block use of the caller id. This checkbox can be selected/unselected using a feature code.
The External CID Number: can be chosen among various formats, usually resembling the
E.164 standard. The various options can be enabled or disabled using the Admin/Settings menu. The External CID number can be chosen only from the DIDs assigned to the tenant.
External CID name: allows defining the alphabetic part of the Caller ID.
Emergency CID number: allows you to define the Caller ID number to use when an emergency route is used to dial out. This can be chosen among the DIDs marked as“emergency”. The location of the DID is shown if entered.
Area Code: allows specifying a prefix to add to numbers when the number of digits entered is between the number of digits specified underneath, inclusive. For example, if you area code is 713and your local area numbers are from 4 to 7 digits, you can enter the following data and your number will be automatically completed with the area code. So, if you enter 4531311, automatically the number dialed will be 7134531311.
Routing Profile: allows you to assign this extension a different routing profile than the one assigned to the tenant.
7.Find me/Follow me Configuration:
This feature allows you to define a simple “next hop” for calls when the dialed number is busy or not available. 
Active if Checked: The FMFM configuration needs to be enabled using the “Active if checked” check box. You can also use a feature code to enable or disable it.
FMFM Number: is the number to dial when the extension is busy or not available.
FMFM Dial Method: permits to choose between two dialing methods, “normal” for when the
FMFM number is dialed after the “Inbound dial timeout” for the extension, and “simultaneous” when the FMFM number is dialed at the same time as the extension number.
Request Confirmation: allows asking the called FMFM destination to accept the call, playing the default message or a custom message. If the called FMFM number refuses the call, the call is treated as BUSY (obeying whatever instructions are set in “Additional destinations” for BUSY)
FMFM Caller ID: gives the ability to choose which Caller ID to display to the called number. Two special options are available:
- Use Original will use the caller’s Caller ID
- Use Incoming DID will use the DID receiving the call as Caller ID
FMFM Caller ID Num and Name prefix: defines a prefix to add to the Caller ID chosen when dialing the FMFM number
FMFM Dial Timeout: defines the seconds to dial the FMFM number before going to the
Additional Destination, if defined.
8. Additional Destinations:
This section allows you to specify the destination of the call when the extension is not answered, Busy or Offline.
When defining the Additional Destinations, you may choose a special destination, “Voicemail Same Number”. This destination will automatically create a voicemailwith the same number as the extension and a random PIN. If the voicemail was already available, it will be just assigned to the destination.
Unconditional - allows the redirection of the phone calls to another destination. Every kind of Additional Destination can be enabled or disabled using Feature Codes.
On No Answer: If the line is not picked up by the extension for any reason
On Extension Busy: If the number of channels for a single line exceeds the predetermined amount
On Extension Offline: If the extension isn't available (e.g. due predetermined available hours)
Bulk extension creation
It is possible to create multiple extensions at once by pressing the “Bulk SIP peer” button.
The definition/settings page will be the same as explained above except for the number range requested.
Learn more about setting up an SIP extension
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9. Deleting an Extension:
Extension Deletion:
To delete an extension, you would click the delete button at the end of the extension definition. A message will request confirmation. Deleting the extension will unregister and clean it from the PBX peer cache, denying any operation for the deleted extension.
Multiple Delete of Extensions:
To delete multiple extensions at once you would click on the garbage can icon located in the upper right hand corner of the extensions list. Once clicked, a new column will appear on the left, allowing you to select the extensions you wish to delete. Once you have the extensions you wish to delete selected, you’ll then click the red “Delete Selected” button (located in the upper right hand corner, as shown below).
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