Generic SIP phone (SDX1)
This resource allows to connect a generic SIP phone.
Version 1.06 - General deployment
Version 1.05 - General deployment
- Improvement: Added optional H.263 codec. (M0005447)
Version 1.04 - General deployment
- Improvement: Allow to use the remote-party-id header. (M0004626)
Version 1.03 - General deployment
- Feature: Re-enable the option to use the user's pincode as the secret
Version 1.02 - General deployment
- Added codec selection
- Added support for active-active softphones
Version 1.2.0 - Early deployment
- Added possibility to set no password
- Added possibility to choose the DTMF mode
- Added possibility to choose a password different from the pincode (default: 0000)
Version 1.01 - Deprecated
Version 1.1.0 - Deprecated
- Feature: Generic phone over internet (TS-594)
Version 1.0.12 - Early deployment
- Deprecated: never used in prod
Version 1.0.11 - General deployment
- Improvement: Allow to configure SIP Session timer (M19716)
Version 1.0.10 - General deployment
- Improvement: Added Escaux presence
Version 1.0.9 - General deployment
- Improvement: Added the possibility to disable the mailbox and the SIP notification (M9214)
Version 1.0.8 - Deprecated
- Feature: Added the possibility to add remote party presence information in the pidfnote (M8423)
- Feature: Added the possibility to configure the ptime (M8709)
Version 1.0.7 - General deployment
- Deprecated: Revoked NAT support (M0007329)
Version 1.00 - General deployment
- Improvement: H.261 / H.263 / H.263+ / H.264 video support
Resource configuration interface
Resource configuration parameters
This resource allows and administrator to configure a generic sip phone.
- Use authentication:
- Yes - Custom password: Use authentication. The password will be the one indicated in the field 'Password'. If the 'Password' field isn't set, the default password '0000' will be used.
- No: Do not use authentication. No password will be required/allowed.
- Yes - Pincode: Use authentication. The password will be the pincode of the extension linked to this resource. Note that you should only have one extension which uses this resource.
- DTMF mode:
- RFC2833: Use the DTMF transmission described in RFC2833. (DTMF are sent through RTP packets)
- Inband: Try to detect DTMF in the voice traffic.
- Password: Password used in the SIP authentication when 'use authentication' is set to 'yes'. (Default: '0000')
- SIP server feature: Note: These technical settings if improperly configured can have impact on the availability of the phone
- SIP re-invite: allow the voice media to be transferred directly between SIP endpoints through the SIP re-invite procedure. Possible value are:
- Default: Keep default settings as defined in the Telephony server module (for example Asterisk 1.2.x)
- Yes: Allow the server to perform the re-invite procedure.
- No: Do not allow the server to perform the re-invite procedure.
- 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: If this parameter is set to 'Yes', the SIP server will only accept the 3 codecs defined in this resource with the specified preference.
- First Codec, Second Codec, Third Codec: Ordered list of codec
- Use Remote Party ID header: Set to 'yes' to handle the Remote-Party-ID SIP header. Set to no to ignore it.
- Pidfnote: The pidfnote is the text display next to the presence of a contact. Set to 'Default' to keep default behaviour. Set to 'Add Remote Party Presence' to add put the information concerning the caller and called channel
- Mailbox: Set to 'Enabled' to link the SIP peer with the mailbox of the associated extension. This will be used when sending MWI NOTIFY
- MWI Notify method: Set to 'Automatic' to send MWI-NOTIFY upon the reception of the REGISTER. Set to 'Upon Subscribe' to send the NOTIFY only when the proper SUBSCRIBE is received