Clicking the Advanced Options button will show advanced configuration parameters and fields that were previously hidden from view.
The following options can be used to directly influence how the extension operates, these options vary from changing the way the Voicemail works, all the way to changing the SIP header the device is using to communicate.
By default these options are pre-configured to work out of the box. It is not recommended to change these unless explicitly required or prompted to do so while saving changes. If the wrong options are changed here the outcome may result in your extension not operating how it is expected to.
This field sets the extension number.
By default, this field is automatically populated, but can be changed to any Extension number within the tenant range. We recommend a four digit extension number on all tenants, as some 3 digit codes are linked to emergency services and system feature codes (e.g 700).
(ex. Setting '1008' here will create a new system extension with the same network number.)
Users title like Mrs, Mr, and such
Full name of the person using the Extension. This is the name that displays when internal calls are placed to other extensions and the name that shows on the handset itself.
Email address associated with the extension. This email will be used for the system welcome email, voicemail notifications, the username for the CallSwitch Business app and used for various other system notifications.
(ex. Setting 'email@example.com' here will transfer all Voicemail notifications, Extension PIN and other details to this email)
(ex. Location like Street, City, or State)
Department to which extension will belong to. This is used to assign CallSwitch app privileges and can be used to group extensions depending on which department they belong to.
The best practice is to set up the departments before creating extensions.
See the "Departments" Article for further information.
- User Type
Extensions can be set to make calls only, receive calls only or both make and receive calls
(ex. Select 'User' to make the calls only; 'Peer' to receive the calls only; or 'Friend' for both, to make and receive calls)
- DTMF Mode (Dual Tone Multi-Frequency)
A specific frequency, consisting of two separate tones. Each key has a specific tone assign to it so it can be easily identified by a microprocessor.
This is a sound heard when dialing digits on touch-tone phones. Each phone has different 'DTMF Mode'.
(ex. By default, this field is populated automatically by the UAD, 'rfc2833' is generally the default.)
Every system extension belongs to a certain system context, this is used for back end routing.
Extension status/presence on the network.
Rather than deleting the extension and then recreating it again later on, the extension can be activated/deactivated using this field.
(ex. Setting this field to 'Not Active' will disable all calls to this extension).
Active - Extension is active, it can make and receive calls.
Not Active - Extension is not active and it can't make nor receive calls.
Suspended - Extension is suspended and can't make calls to numbers other than those defined as Emergency Service numbers in Settings -> Servers -> Edit Server -> Locality (section) -> Emergency Services,
- Music on Hold:
On extension's level, Music on Hold is new feature introduced in version 5.
It allows users to choose different Music on Hold class for every extension. Select the MOH (Music On Hold) class name. All sound files belonging to this MOH class will be played to users dialling this extension.
NOTE: When you select MOH class, you need to enter 'm' in Incoming Dial Option field. Entering 'm' will provide Music on Hold to the calling party until the called channel answers.
- Show In Directory:
Whether the extension should be shown in Remote Directory accessed through the deskphones interface.
(ex. Yes, No, N/A)
NOTE: Device must support this feature.
These options are used mainly for registration to the CallSwitch platform.
Username used by the Extension for the registration with the CallSwitch Platform.
By default, this field is the Tenant number and Extension network number and cannot be changed.
Name that can be used for authentication. We utilise extension Username and Secret for authentication but this field will be useful if an extension is being used as a SIP trunk as it changes the SIP header sent.
If you set this field to 12345, for example, the sent SIP header will look like firstname.lastname@example.org
Auth is the optional authorisation user for a SIP server (Not used by default)
Secret/Password used by the Extension device for the registration with the CallSwitch Platform.
By default, this field is automatically populated but can be changed.
NOTE: CallSwitch enforces strong password enforcement for secrets, which means that secret must meet certain criteria in order to be accepted otherwise, CallSwitch will display an error message stating that secret is too weak.
Secret has to meet the following criteria in order to be accepted:
- It must be at least 8 characters long
- It must contain at least 1 uppercase
- It must contain at least 1 lowercase
- It must contain at least 1 digit
- It must contain at least 1 special character
- Allowed characters are: a-z, A-Z, 0-9, ! % * _ -
In order to make it easier for our users, we also implemented password generator, that will automatically generate strong password that meets above criteria with a single mouse click on a key icon located on a side of Secret field.
- User Password
Password used for CallSwitch App registration (both Desktop and Mobile App, by default this field will be automatically populated.
The Password is sent in the welcome email of the extension if you press "Save & E-mail" on the extension settings).
TIP - Once the user changes their password this field will show "encrypted" so you cannot see the entry. This is very handy when troubleshooting app login problems, on first sign in to the app the apps force you to change this password, so it allows you to see if the user has logged in and changed their password.
- Show QR Code (Beside User Passwords)
Show QR Code button will display QR code that can be scanned with the CallSwitch mobile application (built in QR reader). This feature will make mobile app setup and registration process as fast and simple as possible.
NOTE: Once you are registered to the Callswitch Platform with your CallSwitch App you will be asked to change the automatically generated User Password in order to log in. Once this procedure is completed User Password will not be visible anymore and the QR Code button will be hidden.
- PIN (Personal Identification Number)
Four digit number used for account authorisation for Voicemail, Agent log in and any PIN authentication set on the Enhanced services features.
NOTE: This number must always be four (4) digits long
(ex. If the PIN for this extension is set to '8474', provide it when asked for it by TelcoSwitch MT when checking your Voice inbox or other 'Enhanced Services')
Billing needs to be disabled across ALL extensions created on CallSwitch - do not touch this section as it is not required.
Permissions - Destinations, Enhanced Services and Editions & Modules:
Destinations - Call barring/restrictions
These options grant/deny certain local/worldwide destinations, conferences, enhanced services, or call monitoring to your edited extension. If the image below is displayed, all destinations are allowed for the user extension. Should extension permissions be changed, click the 'Set destinations manually' button
Manually, destinations are set through the following groups:
- Remote - PSTN/External destinations.
- Local - All destinations within the system/network (Extensions, IVR, Queues, Conferences...).
- Special Routes - Generally not used.
- PIN Required
- Not Authorised
Note - The padlock icon (PIN Required) will set a pin authorisation to be able to access the destination dialled. The PIN will be the extension numbers pin (the same PIN as Voicemail)
Enhanced Services allow users to fully adjust settings like Caller ID, Do not Disturb, Call Pickup, and Call Forwarding etc.
For a breakdown of how to set up Enhanced services, Please see the Enhanced services Article:
For the online self care portal see here:
Editions & Modules:
Here you can set which CallSwitch editions the extension can use. This can be done manually by pressing the edition button or with a template by assigning a department.
See the Departments Article for how to set up Departments.
These options set important network related values regarding NAT, monitoring and security Settings here are inherited from the UAD when the extension is created.
Type of transfer protocol that will be used on the CallSwitch platform. The handset and UAD assigned to the extension needs to support the protocol selected. Some protocols will need specific configuration changes to be able to work with certain protocols - e.g TLS
UDP (User Datagram Protocol) - With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol (IP) network without prior communications to set up special transmission channels or data paths.
TCP (Transmission Control Protocol) - provides reliable, ordered, error-checked delivery of a stream of octets between programs running on computers connected to an intranet or the public Internet.
TLS (Transport Layer Security) - cryptographic protocol that provide communication security over the Internet. They use asymmetric cryptography for authentication of key exchange, symmetric encryption for confidentiality, and message authentication codes for message integrity.
This option enables or disables encryption on CallSwitch transport.
Options: Yes, No, N/A.
- NAT (Network Address Translation)
NAT Always needs to be set to Yes for the extension to work.
- Direct Media
We do not support direct media by default.
- Direct RTP setup:
We do not support direct RTP by default.
Timing interval in milliseconds at which a 'ping' is sent to the Extension, in order to find out its status(online/offline). Set this option to '2500' to send a ping signal every 2.5 seconds. Navigate to 'Monitor: Extensions' or 'Monitor: Trunks' and check the 'Status' field.
In CallSwitch v5 'Qualify' is set to 8000 by default.
Set the way the Extension registers to CallSwitch. Set this field to 'dynamic' to register the UAD/Phone from any IP address. Alternately, the IP address or hostname can be provided as well.
- Default IP
Default Extension IP address. Even when the 'Host' is set to 'dynamic', this field may be set. This IP address will be used when dynamic registration could not be performed or when it times out.
NOTE: Extension must be on static IP address.
- Max Contacts
Maximum number of contacts per extension. This number is set as default globally.
This can be overwritten by changing max contacts number per Extension.
The caller's name and number displayed here are sent to the party you call and are shown on their Phone display. The information you see here is taken from the extension number and user name. To set different Caller ID information, please go to 'Enhanced services: Caller ID' and set new information there.
- Set Caller ID
Enable 'Caller ID' service
(ex. Set this option to 'Yes' to enable the Caller ID service)
- Caller ID
Extension Number and Name that are displayed on dialled party UAD/Phone display
(ex. These options are read-only. Caller ID information can be changed only through 'Enhanced Services')
- Caller ID Presentation
The way Caller ID is sent by the Extension
If CallSwitch is connected to a third-party software and there are problems with passing the Caller ID information to it, applying different 'Caller ID Presentation' methods may resolve the problem
(ex. Presentation Allowed, Not Screened)
- Hide CallerID for Anonymous calls
When you set this option to Yes, incoming calls with Anonymous as number but have CallerID set, are then formatted as Anonymous <anonymous>.
(ex. Yes, No, N/A)
- Ringtone for Local calls
If you know which phone is registered on this extension, you can set a custom ring tone for local calls. This is set at a UAD level so does not need to be specified here unless another selection is required.
(ex. If your phone is a Yealink T46G you could set Ring2)
- Ringtone for Transferred calls
Ringtone for transferred calls, work same as Ringtone for local calls setting. Depending of your phone manufacturer you can send a string to the phone in order to use different ringtone than the one set on device. Once this string is set in Ringtone for transferred calls field (as well as in your device itself) it will be used for all calls that are transferred to your extension.
We recommend setting the above settings in the UAD script.
- Only Allow Trunk CallerID within DID range
When you assign an extension to a customer and assign some DIDs to it, customer can make calls through that extension with CallerIDs that match its DID numbers. If a customer tries to make a call with a CallerID that doesn't match any of the DIDs assigned to him, the caller ID will be reset to Anonymous.
Defines whether CallSwitch allow Remote-Party-ID header
Enables 'Remote-Party-ID' be added to SIP URI.
- Use Remote-Party-Id - Use the "Remote-Party-ID" header to send the identity of the remote party
- Use P-Asserted-Identity - Use the "P-Asserted-Identity" header to send the identity of the remote party
- Connected Line Updates:
This option is particularly useful as with some providers, if Use PAI is enabled, calls might start dropping short time after update is sent. Setting Connected Line Updates to No will prevent these call drops.
- RPID with SIP UPDATE:
In certain cases, the only method by which a connected line change may be immediately transmitted is with a SIP UPDATE request. If communicating with another Asterisk server, and you wish to be able transmit such UPDATE messages to it, then you must enable this option. Otherwise, we will have to wait until we can send a reinvite to transmit the information.
These options fine-tune incoming/outgoing call settings.
Extension ring time, this is the "Timeout" of the extension. For example, when a call forward on no answer is set the value defined here will be the amount of time it will ring the handset for before overflowing.
Time in seconds that the UAD/Phone will ring before the call is considered unanswered (default: 32).
- Incoming & Outgoing Dial Options
Advanced dial options for all calls.
Please see the "Dial Options" Article for a detailed list of all available dial options (default: tr).
These options define who is allowed to pickup our calls, and whose calls we are allowed to pickup.
- Call Group
This is the Group that the extension you have selected is assigned to. Example: Setting group 1 here will make the extension a member of Pickup group 1.
- Pickup Group
Set which groups the extension is allowed to pickup by dialling '*8' or pressing the pickup key.
Note: Grouping works only within a technology (SIP to SIP or IAX to IAX)
- Call Group = 1
- Pickup Group = 3,4
- Call Group = 2
- Pickup Group = 1
- If A is ringing, B can pickup the ringing call by dialling '*8' or pressing the group pickup key.
- If B is ringing, A cannot pickup the ringing call because B's Call Group = 2, and A can pickup only Call Groups 3 and 4.
These options enable extensions to use custom default trunks for all outgoing calls (this is usually set at a tenant level)
- Primary/Secondary/Tertiary Trunk:
Set the default trunks for all routes dialled from this extension.
If the connection is not established through the primary, the secondary trunk is used, etc.
- Override System LCR
This option tells the systems that when making calls, they should omit checking LCR.
These options set the number of simultaneous incoming and outgoing extension calls.
- Incoming Limit:
Sets the maximum number of simultaneous incoming calls. If an extension receives more incoming calls than set here, they are all redirected to the extensions voicemail box. by default this is generally set to 3.
- Outgoing Limit:
Sets the maximum number of simultaneous outgoing calls. The outgoing call can be placed on hold and another call can be made from the same extension. However, this feature has to be supported by the device being used.
Note: The device being used with this extension must support the limits placed here
- Busy level:
Maximum number of concurent calls until the user/peer is considered busy. This option is not intended for blocking calls, but for displaying user/peer status properly, for example in BLF.
- Apply Busy Level for Incoming Calls:
If "Apply Busy Level for Incoming Calls" is set,and the extension receives an incoming call over the freshhold of the Busy level while the extension is on a call, the incoming call is blocked and redirected to Voicemail or any other option set for extension.(For this to work One needs to set option Busy Level to 1 under group Call Control).
- Play sound on exceeded limit:
If you try to make more calls than allowed in the Outgoing Limit, you will be played a message that the limit has been exceeded.
- Send e-mail on exceeded limit:
Whether or not to send a notification mail on the exceeded limit.
- Notification e-mail:
E-mail address to which notification mail should be sent if the number of calls exceed the limit.
By default, each extension is equipped with a voice mailbox when it is created. standard voicemail administration is done here.
Note: Details on how to Access System Voicemail boxes can be found below:
A description for all the configurable fields can be found below:
Enables/disables the Voicemail service.
When enabled if a call is placed and no one picks up the handset after the specified "Ringtime" (default 32 seconds), the calling party will be transferred to the dialled extensions voicemail box and offered to leave a voice message.
If your Voicemail is turned on, you can set this option to yes to play a greeting and then a busy sound. This is useful for people who want to present a message but do not want people to have the option to leave a message.
- MWI extensions (comma separated):
Message waiting indication enables the extension to notify multiple extensions when a voicemail has been left. Once an extension is added here the system will indicate that a voicemail has been left on the extensions selected here.
(ex. 2002, 2003, 2004, 2005)
Mailbox extension number
(ex. This value is the same as the extension number and cannot be modified)
(Read-only, cannot be modified)
Full name of the user associated with the voice box.
(Read-only, cannot be modified)
- PIN: (Personal Identification Number)
Password used for accessing voicemail. The value of this field is set under 'Authentication: PIN'.
Email address associated with the voice inbox. This email is used for a new voice message notification and audio file attachments
(ex. If 'email@example.com' is set here, once this mailbox receives a new message, a notification and the attached voice message (depending on if this option is enabled) will be sent to this email address)
- Send e-mail
Whether or not to send an e-mail to the address given above.
- Pager e-mail:
Provide the pager e-mail address here
ex. If 'firstname.lastname@example.org' is set here, once this mailbox receives a new message, a notification is sent to this pager email address (no attachment on this email, just a notification)
- Greeting message:
Greeting message played to users before they are transferred to the voice mailbox to leave a message.
(ex. Mailbox user may choose between a 'Busy' and 'Unavailable' message)
- Skip Instructions:
Skip the instructions telling users how to leave a voice message
(ex. Once the caller reaches the voice mailbox, instructions on how to leave voice a message is played. We encouraged to set this option to 'Yes' all the time)
If this is disabled the caller will be transferred straight to the "Beep" sound enabling recording after the greeting.
Whether the voice message should be attached and sent along with the notification email
(ex. A caller leaves a voice message for John. With this option set to 'Yes', the notification email John gets will have the voice message attached to it, so John can listen to it without signing in to his voice mailbox)
- Delete After E-mailing:
Whether the voice message sound file should be deleted from the filesystem after sending it as an attachment to the user's email address
(ex. The caller leaves a voice message to John. With this option set to 'Yes', the voice message will be deleted after sending it as an attachment to John's email address)
- Say Caller ID:
Allow the user to review his voice message before it sends to the voice mailbox
(ex. After a caller leaves a voice message and presses '#', additional review options are allowed: 1 to accept the recording, 2 to re-record your message, etc.
- Allow Review mode:
Allow caller to review the voice message before committing it permanently to the extensions voice box.
The Caller leaves a message on extension 2002's voicemail box, but instead of hanging up, he presses '#'. Three options are offered to the caller:
- Press 1 to accept this recording
- Press 2 to listen to the message
- Press 3 to re-record your message
- Allow Operator:
Allows the calling party to reach an operator from within the voicemail box.
Calling party leaves a message on 2002's voicemail box, but instead of hanging up, the caller presses '#'.
The caller will hear the 'Press 0 to reach an operator' message (Once '0' is pressed, user is offered the following options):
- Press 1 to accept this recording (If selected, 'Your message has been saved. Please hold while I try that extension' is played and operator is dialled)
- Or continue to hold (If the caller holds for a moment, 'Message deleted. Please hold while I try that extension' is played and operator is dialled)
- Operator Extension:
Local extension number that acts as an operator.
(ex. If the extensions voicemail box has the option 'Allow Operator' set to 'Yes', all users dialling '#0' inside the voice box will reach this operator extension).
- Play Envelope Message:
Announces the date and time when the voice message was left in the inbox
(ex. With this option enabled, John will hear 'First message, 11:52, 02 Feb 2007', for example, when checking his voice mailbox)
- Hide from directory:
This option will allow you to hide your extension from the Directory/BLF list.
- Rings to answer:
Number of rings played to the caller before the call enters Voicemail
(ex. Rather than just 'falling' into Voicemail, it is recommended to set the number of ring sounds played to caller)
NOTE: By default, this field is empty which means that there will be no ringing. The caller will go straight into the Voicemail.
- Voicemail Delay:
The time delay in seconds before the Busy/Unavailable message is played to the caller. This solves a previous issue with 'half-played' files. Keep this value between 1-3
(ex. The caller is leaving a voice message to John. It hears '...ot at home right now...'. Adding '1' to this field will add a one second pause before the message is played. So, now the caller will hear the greeting message without the first part being cut off 'I am not at home right now...').
Set the correct date and time format for the message envelope.
(ex. Some countries prefer a time format in the mm-dd-yy or dd-mm-yy format. Select from the available options)
Speakerphone Page Auto-Answer SIP Header:
These options are here to allow the caller to use a UAD in a public announcement system. If the UAD fully supports this service, the call is accepted automatically and put on a loudspeaker. This feature is not currently being utilised but is here in case the requirement comes up.
This section allows certain codecs to be defined for the extension. Codecs are used to convert analog to digital voice signals and vice versa. The Codecs available here are set at a UAD level.
Note: The device being used must support the chosen Codec or issues will be experienced.
Set the codecs extension to allowed to use.
(ex. Only the codecs enabled under the tenant will be available to choose from)
- Video Support
Set this option to Yes to enable SIP video support, This option is for peer to peer video support. a supported video codec will need to be used to support this.
(Yes, No, N/A)
- Force codec on outbound trunk channel
With this option you can force codec use for outbound trunk calls.
- Auto-Framing (RTP Packetization)
If autoframing is turned on, the system will choose the packetization level based on remote ends preferences.
(ex. If the remote end requires RTP packets to be of 30 ms, the CallSwitch platform will automatically send packets of this size if this option is turned on. Default is set to 20 ms and also depends on the codecs minimum frame size like G.729 which has 10 ms as a minimum).
- ITU G.711 ulaw - 64 Kbps, sample-based, used in US
- ITU G.711 alaw - 64 Kbps, sample-based, used in Europe
- ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
- ITU G.726 - 16/24/32/40 Kbps
- ITU G.729 - 8 Kbps, 10ms frame size
- GSM - 13 Kbps (full rate), 20ms frame size
- iLBC - 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
- Speex - 2.15 to 44.2 Kbps
- LPC10 - 2.5 Kbps
- H.261 Video - Used over ISDN lines with resolution of 352x288
- H.263 Video - Low-bit rate encoding solution for video conferencing
- H.263+ Video - Extension of H.263 that provides additional features that improve compression over packet switched networks.
- OPUS - support coming soon.
See Hot-desking article for more information on configuring and using Hot-desking.
- Automatic Logout (hours)
Sets automatic log-out time in hours for devices set up for hot-desking.
- Laws in some countries may require notifying the parties that their call is being recorded.
- Recorded calls, marked with icon, can be accessed from 'Self Care Interface' or 'Reports: CDR' TelcoSwitch' menu.
- Enable call recording service
(ex. Select 'Yes' to enable the service. All incoming/outgoing calls to the group will be recorded. Ensure call recording is enabled on every leg of the call - e.g "DID" > "Ring Group" > "Extension"
Set call recordings should be announced to the parties in a conversation. If Silent=No, calling parties will hear a 'Recorded' or 'This call is recorded' message before their conversation starts.
These options enable CallSwitch to automatically provision extension handsets if supported. Configuration files are downloaded from CallSwitch's provisioning server.
- MAC Address (Media Access Control):
Phone MAC address, this needs to be provided without any formatting, for example: 805EC016845D. If the MAC address is incorrect here the handset will not provision.
- DHCP (Dynamic Hosts Configuration Protocol):
Set whether the Phone is on DHCP or Static IP address. By default this should be always set to "Yes"
(ex. Set DHCP = Yes if the Handset is on dynamic IP or DHCP = No if Handset is on static IP address. If on static IP, you will have to provide more network details in the fields below).
- Static IP:
Static UAD/Phone IP address
(ex. DHCP = No, has to be set. Provide the Handset's static IP address here)
Subnet mask of the Handset.
Gateway IP address
(ex. Local area network gateway IP address)
- DNS Server1 and Server2 (Domain Name Server):
DNS Server IP address
(ex. Local area network DNS IP address (Usually the same as your gateway))
This option simply notifies you of whether device presence is enabled or disabled (if the Handset/UAD supports it)
- Presence Enabled:
Returns the information whether the phone is on call, ringing, or offline (not registered).
(ex. Select 'Yes' to enable presence support, but not every UAD/Phone support this feature)
- Global Presence:
Enables presence like above option but when this option is turned on, it will enable presence on all tenants on the system. (Not required)
User Agent Auto Provisioning Template
This option allows adding of additional settings to auto-provisioning template. Auto-provisioning settings are generally defined in the 'Settings: UAD' and are custom set for each device.
NOTE: Unless absolutely sure, do not change or add to this template.
See the UAD article for more information on this setting.
This option is used for providing additional config parameters for SIP and IAX configuration files. Values provided here will be written into these configuration files.
NOTE: Unless absolutely sure, do not change or add to this template.