Difference between revisions of "SIP Devices and Asterisk"
The Wiki of Unify contains information on clients and devices, communications systems and unified communications. - Unify GmbH & Co. KG is a Trademark Licensee of Siemens AG.
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| Call Log | | Call Log | ||
| OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | ||
| − | | Stores a list of missed, dialed, received, and forwarded calls. | + | | Stores a list of missed, dialed, received, and forwarded calls. With V2R1 onwards, the call log can be cleared via the [[WPI]]. |
| − | | | + | | [[Configurable_flag_to_delete_call_log_contents|Configurable flag to delete call log contents]] |
|- | |- | ||
| MWI | | MWI | ||
| Line 188: | Line 188: | ||
|- | |- | ||
| Call Waiting | | Call Waiting | ||
| + | | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | ||
| If allowed, a call from a third party will be indicated acoustically during an active call. | | If allowed, a call from a third party will be indicated acoustically during an active call. | ||
| − | |||
| - | | - | ||
|- | |- | ||
| Line 198: | Line 198: | ||
|- | |- | ||
| Auto Answer | | Auto Answer | ||
| − | | TBD | + | | - TBD - |
| Automatic call answering can be requested from within the incoming call via the SIP Alert-Info header. | | Automatic call answering can be requested from within the incoming call via the SIP Alert-Info header. | ||
| - | | - | ||
|- | |- | ||
| − | | CFBS, CFNR | + | | Call completion (CFBS, CFNR) |
| − | | - | + | | - TBD - |
| − | | | + | | Call completion is a telephony feature which takes action on a failure to complete a call. It allows for notifying the calling user when the called user is available again. CCBS (Call Completion Busy Subscriber) will take effect when the called party is busy; CCNR (Call Completion No Reply) will take effect when the called party does not respond. |
| − | | - | + | | {{File-DL|White_Paper_CC_10090|pdf}} |
|- | |- | ||
| Call Transfer | | Call Transfer | ||
Revision as of 12:29, 23 September 2010
Preparation
Power Supply
Power over Ethernet
The following phone configurations can be operated with PoE, provided that the switch has the appropriate power class:
| Model / Configuration | Power Class |
|---|---|
| OpenStage 15 (includes 1 Key Module 15) | 1 |
| OpenStage 20 E | 1 |
| OpenStage 20 | 1 |
| OpenStage 20 G | 2 |
| OpenStage 40 (includes 1 Key Module) | 2 |
| OpenStage 40 + 2nd Key Module | 2 |
| OpenStage 40 G | 3 |
| OpenStage 40 G + 2nd Key Module | 3 |
| OpenStage 60/80 (includes 1 Key Module + USB-Extension with Acoustic Unit) | 3 |
| OpenStage 60/80 + 2nd Key Module | 3 |
| OpenStage 60/80 G (includes 1 Key Module + USB-Extension with Acoustic Unit) | 3 |
External Power Supply
For an OpenStage 60/80 G with a 2nd Key Module, an external power unit is required. The order no. for the plug-in power supply is region specific:
- EU: C39280-Z4-C510
- UK: C39280-Z4-C512
- USA: C39280-Z4-C511
Energy Saving
OpenStage phones offer an energy saving mode. The display backlight is switched off after a configurable timeout. With OpenStage 40, the main display and key module backlight will be switched off after 90 seconds of inactivity (firmware version V2R0 onwards). Readability even without backlight is ensured by the transflective display. With OpenStage 60 and 80, the timer is configurable by the administrator (Local Functions > Energy saving); the timeout ranges between 2 and 8 hours.
Connecting to an IP Network
802.1x
OpenStage phones support 802.1x EAP-TLS. Certificates for authentication can be downloaded via the WPI.
LLDP-MED
OpenStage SIP phones support the layer 2 protocol LLDP-MED (Link Layer Discovery Protocol-Media Endpoint Discovery). When an OpenStage phone is connected to a switch with LLDP-MED capabilities, the phone is able to
- advertise and receive a VLAN ID,
- advertise and receive QoS parameters,
- advertise the power requirements to the LAN access switch by means of an "Extended Power via MDI" TLV.
LLDEP-MED usage is configured in the administratio menu under Network > IP configuration.
DHCP
The following parameters can be obtained by DHCP:
Basic Configuration
- IP Address
- Subnet Mask (option 1)
Extended Configuration
- Default route (option 3)
- Static IP routing (option 33)
- SNTP server (option 42)
- Timezone offset (option 2)
- Primary/secondary DNS server (option 6)
- DNS domain name (option 15)
- SIP Addresses / SIP Server & Registrar (SIP Server option 120)
- Vendor unique (option 43)
The vendor specific option (code 43), or alternatively, a vendor class, is used to provide the phone with the location of an optional configuration/provisioning service. By this means, full Plug&Play is possible (see the Plug&Play) section. For further information, including an example configuration for dhcp, please refer to the
Administration Manual OpenStage Asterisk.
Plug&Play
A fully automated mass rollout of OpenStage phones can be realized by combining a DHCP server and a provisioning service which uses the WPI. On startup, the phone receives the IP address of the provisioning server from the DHCP server. After that, it contacts the provisioning service. The provisioning service may then request all settings from the phone in order to decide which parameters must be set or updated. When all these parameters have been sent to the phone, it is ready for operation.
For further information, please see the WPI article; for a deeper understanding, refer to the OpenStage Provisioning Interface Developer's Guide.
Using OpenStage Phone with Asterisk
Feature Table
In this table, you find information on all features which are supported by OpenStage phones connected to an Asterisk PBX.
| Phone Applications | |||
|---|---|---|---|
| Feature | Phone with firmware version ≥ | Short Description | Further Information |
| Local Phone Book | OpenStage 60/80 ≥ V1 R5.6.0 OpenStage 40 ≥ V2R1 |
Stores names and call numbers locally on the phone. | OpenStage_Phone_Book_Application |
| LDAP | OpenStage 60/80 ≥ V1 R5.6.0 OpenStage 40 ≥ V2R1 |
The phone can be used to retrieve call numbers or other address data from an LDAP directory. | LDAP on OpenStage |
| Call Log | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | Stores a list of missed, dialed, received, and forwarded calls. With V2R1 onwards, the call log can be cleared via the WPI. | Configurable flag to delete call log contents |
| MWI | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | Message Waiting Indication. The user is notified of new and old voicemail messages. With V2 onwards, the message indication is configurable. | Configuration of MWI count names |
| Help | OpenStage 60/80 ≥ V1 R5.6.0 | Quick reference which is shown on the phone display when the Help mode key is pressed. | - |
| XML Application Platform | OpenStage 60/80 ≥ V1 R5.6.0 | XML-based interface to the phone which allows for developing interactive applications. The phone acts as a front-end for a server-side program. | XML Application Platform at the SEN Community Portal |
| CTI Applications | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | Allows a computer to interact with the phone, e.g. in setting up and terminating calls via a PC application. | OpenStage_CTI_Applications |
| FPK (Free Programmable Keys) | OpenStage 15/40/60/80 ≥ V1 R5.6.0 | These keys can be associated with special phone functions. | OpenStage_Free_Programmable_Keys |
| Call Related Features | |||
| Busy Lamp Field (BLF) | OpenStage 15/40/60/80 ≥ V1 R5.6.0 | This function offers the possibility to monitor another extension, and to pick up calls for the monitored extension. The LED of the key will indicate the state of the extension monitored. | Asterisk Feature Busy Lamp Field (BLF) |
| Phone Based Conference | OpenStage 15/40/60/80 ≥ V1 R5.6.0 | The phone establishes and controls a 3-way conference (3pcc, third party call control). | - |
| Voice Mail | OpenStage 15/40/60/80 ≥ V1 R5.6.0 | The user can contact the Asterisk voice mail service by means of the messages key. | - |
| Call Group | - | - | - |
| Call Pickup | OpenStage 15/40/60/80 ≥ V1 R5.6.0 | The BLF function offers the possibility to pick up calls for a monitored extension. The LED of the key will indicate the state of the extension monitored. | Asterisk Feature Busy Lamp Field (BLF) |
| Call Forward | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | The user can configure local or server based call forwarding to specified destinations. | For information about server based call forwarding, see Fix Forwarding Key to send URL to Server. |
| Call Waiting | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | If allowed, a call from a third party will be indicated acoustically during an active call. | - |
| Do Not Disturb (DND) | OpenStage 15/20/40/60/80 ≥ V1 R5.6.0 | If this feature is activated, incoming calls will not be indicated to the user. | - |
| Auto Answer | - TBD - | Automatic call answering can be requested from within the incoming call via the SIP Alert-Info header. | - |
| Call completion (CFBS, CFNR) | - TBD - | Call completion is a telephony feature which takes action on a failure to complete a call. It allows for notifying the calling user when the called user is available again. CCBS (Call Completion Busy Subscriber) will take effect when the called party is busy; CCNR (Call Completion No Reply) will take effect when the called party does not respond. | White_Paper_CC_10090
|
| Call Transfer | - | - | - |
| CheSe | - | - | - |
| Local three party confernce | - | - | - |
| DTMF | - | - | - |
| Alternate Call | - | - | - |
| Call Hold | - | - | - |
| MWI | - | - | - |
| Mobility | - | - | - |
| Consultation | - | - | - |
| CCBS | - | - | - |
| CCNR | - | - | - |
| CLIP | - | - | - |
| CLIR | - | - | - |
| TAPI | - | - | - |
| Local Music on Hold | - | - | - |
