This resource allows an administrator to configure an Aastra 6737i IP Phone.
Version 1.11.0 - Early deployment
- Feature: Implements Aastra 6737i (M10280)
- "Aastra Phone Support" module 2.4.0+
- "Aastra Virtual Phone" resource 1.7.0+ (optional: upgrade if used)
- "PhoneZeroConf" task 1.12.0+ (optional: upgrade if used)
Resource configuration interface
Resource configuration parameters
- Programmable keys: Define the behaviour of the programmable key. The possible behaviours are:
- Default: The behaviour as defined in the factory settings
- Speed dial: Speed dial to the number indicated in the 'value' field
- Speed dial and BLF: Speed dial with busy lamp indication of the extension indicated in the 'value' field
- Local directory: Access to the directory locally stored in the phone
- Callers list: Access the list of all received calls
- XML Application: Enable to access an XML application for which the HTTP url must be filled in in the 'value' field. If the value is set to /aastra/Dir.php, the application will access the corporate directory. This will display data in english by default. If you want to customize your output language or messages, please set the value as following /aastra/Dir.php?lang=[language]. Currently, replace [language] by one of these values: en,fr,nl.
- Services: Access the predefined services
- Display: Select how the phone will display your information on the phone screen. You can select:
- Extension only : to only display your extension,
- Extension + name: to display both name and extension
- name: to only display name.
- Dialplan: String compatible with the digit map feature of MGCP (cfr RFC3435) which indicate when auto-dial should be triggered
- Application URL:
- Idle URL: URL being pushed
- Auto append extension: If you want to append the extension of the phone to the pushed url, put this flag to yes.
- SIP server feature:
- SIP reinvite: Indicate if SIP reinvite should be enabled, disabled or should not overwrite the global SIP reinvite setting
- Qualify SIP devices registration: In order to monitor the SIP device availability, the Telephony server can send SIP OPTIONS messages (every 60s). If the device does not respond within the defined delay, it will be considered as off-line. Possible values are:
- Default: Keep the default setting defined in the Telephony server module (for example Asterisk 1.2.x)
- No: Disable Qualify SIP registration for this device
- Yes: Enable Qualify SIP registration for this device, with a default timeout value of 2000ms
- Suggested number: Enable Qualify SIP registration for this device, with the chosen timeout value (expressed in milliseconds).
- Force specific codec: Indicates if the codec below should be used or if the global codec list should be used
- First codec: Preferred codec to use during call setup
- Second codec: First fallback codec to be used during a call setup
- Third codec: Second fallback codec to be used during a call setup
- Registration refresh period: period after which the phone must refresh its SIP registration
- Use authentication: To define authentication method that should use the phone to register. You can select one of following:
- Default password
- No authentication
- Pin Code
- Custom password
- Password: Defines the password to use to register.
- Primary SIP Proxy: The primary SIP server to register against.
- Secondary SIP Proxy: The secondary SIP server to register against.
- Advanced options:
- Language: The language to be activated on the phone UI
- Automatic Off-Hook Call Placement: Automatic call on off-hook event
- Contact to call: The contact to call on Off-Hook Placement
- Callist-received: Enable or Disable the access to the caller list
- Callist-missed: Enable or disable the missed call indication
- Date format: Specify the date format to be used to display the date on the phone screen
- Hour format: Specify if the 12h time format or the 24h time format should be used
- Time zone: Specify the time zone to be used to display the time on the phone screen. The daylight saving is automatic.
- Voicemail service extension: The extension to call for voicemail.
- Switch configuration:
- Enable VLAN: Enables or disables VLAN tagging on the phone
- Voice VLAN id (802.1q: 1 -> 4094): VLAN id to be indicated in the voice traffic
- Data VLAN id (802.1q: 1 -> 4095): VLAN id to be indicated in the data traffic coming for the PC port of the embedded switch
- Diffserv setting for Voice Media (RTP): Priority to be used to route the RTP traffic inside the network
- Diffserv setting for Voice Signaling (SIP): Priority to be used to route the signaling traffic inside the network
- Disable PC LAN Port: To disable the phone's PC LAN port.
Conditional Idle is a service that can be polled or pushed to the supported phones to display the status for that particular extension.
It also allows you to display a certain variable under certain conditions. For example: you can use this service, to check if you are in forward mode, and if so, display the forward number.
Navigate to: Resources > IP Phone > Select your Aastra phone > Edit
For Aastra Phones, add the following url to your idle url, and set append extension to true.
- Where if contains the parameter to be compared.
- Where equals contains the value to compare the parameter to.
- Where thenshow contains the value to display if the compared values are equal.
- Where withlabel contains the value to prefix the display if the compared values are equal
As of Aastra Phone Support 2.0.1, we allow to show if the extension is logged into a queue. You can use up to 3 queues. You will need SOP Base 1.4.6 installed.
To set up the queues to monitor you have to set the following parameters in the URL:
q1v is the first queue to monitor.
q2v is the second queue to monitor.
q3v is the third queue to monitor.
If you are not logged into any queue, no indicator will be displayed.
Some devices don't have enough space to display the queues if your status and your extension fill up the whole screen.
Queue names are fetched from the profile parameters, you cannot set the name directly. You will have to indicate a var number, as explained below.
Acessing Profile Parameters
Possible values for if, thenshow, q1v, q2v and q3v are var_1 to var_42, they match the parameter order defined in your profile
Navigate to: Communication Flow Studio > Profile
| Parameter XY matches
This service is supported up to 50 phones.
- When both the corporate directory and the Application URL are defined, if the user changes his status or his queues statuses, he will be kicked out of his corporate directory until the changes appears on the phone. After the change has appeared, the corporate directory can be used like usual.