User manual
Table Of Contents
- 1. Introduction
- 2. Pre-Installation
- 3. Programming the VoWiFi Handset
- 4. Installation of VoWiFi Handsets
- 5. Maintenance
- 5.1 Handset
- 5.1.1 Configure Spare Handsets without a Number in Large Systems
- 5.1.2 Upgrade Handset Software
- 5.1.3 Upgrade Software OTA via TFTP
- 5.1.4 Upgrade Software via PDM
- 5.1.5 Upgrade Software Over the Air (OTA) via Centralized Device Management (IMS3)
- 5.1.6 Recapture the Earlier Software
- 5.1.7 Upgrade Handset Functionality using License
- 5.1.8 Perform a Factory reset
- 5.2 Replacement of Handsets
- 5.3 Change Number of a Handset
- 5.4 Update Parameters via IMS3
- 5.5 Perform a Security Upgrade via IMS3
- 5.6 Upgrade the Template
- 5.7 Create a Configuration Backup
- 5.1 Handset
- 6. Handset Configuration
- 6.1 Select Network
- 6.2 IP Address Settings
- 6.3 Network Settings
- 6.4 Security Settings
- 6.5 Handset Settings
- 6.5.1 Automatic key lock
- 6.5.2 Phone lock
- 6.5.3 Audio adjustment
- 6.5.4 Headset Configuration
- 6.5.5 In Charger Behavior
- 6.5.6 Configure Profiles
- 6.5.7 Hide Missed Call Window
- 6.5.8 Battery Warning
- 6.5.9 Shared Phone
- 6.5.10 Prevent Handset Switch off
- 6.5.11 Uploadable Language
- 6.5.12 Select Default Language
- 6.5.13 Short cuts
- 6.5.14 Soft Key Functions During Call
- 6.5.15 Import Contacts
- 6.5.16 Company Phonebook
- 6.5.17 Central Phonebook
- 6.6 Messaging and Alarm
- 6.7 Messaging Settings
- 6.8 Alarm Settings
- 6.9 Telephony
- 6.9.1 Endpoint ID and Endpoint number
- 6.9.2 VoIP Protocol
- 6.9.3 Codec
- 6.9.4 Offer Secure RTP
- 6.9.5 Internal Call Number Length
- 6.9.6 Emergency Number
- 6.9.7 Voice Mail Number
- 6.9.8 Message Centre Number
- 6.9.9 Max number of Call Completions
- 6.9.10 Dial Pause Time
- 6.9.11 Direct off Hook from Charger
- 6.9.12 Replace Call Rejected with User Busy
- 6.9.13 Busy on 1 / Disable call waiting
- 6.9.14 Calling Line Restriction
- 6.10 Regional Settings
- 6.11 Display
- 6.12 Menu Operation
- 6.13 Push-To-Talk (PTT) Group Call
- 6.14 Presence Management
- 6.15 Location
- 7. Use Handset to Verify the VoWiFi System Deployment
- 8. Handset Internal Web Administration Page
- 9. Administration
- 10. Troubleshooting
- 11. Related Documents
- 12. Document History
- Appendix A: Working with Templates
- Index
- Contents
TD 92675EN
20 June 2012 / Ver. E
Configuration Manual
Ascom i62 VoWiFi Handset
51
6. Handset Configuration
• SIP Transport – defines the protocol to use for SIP signaling, either UDP, TCP or TLS. The
setting TLS requires the PBX certificate to be uploaded as root certificate.
• Outbound proxy mode – Set to “Yes” if the handsets ar
e to connect with the SIP proxy
through an outbound proxy. Set to “No” if the handsets are to connect directly with the
SIP proxy (there may be two).
• Primary SIP proxy – defines the primary SIP proxy by
a domain name, a
n IP address or an
IP address together with a port number. The parameter is only visible if the parameter
Outbound Proxy mode is set to “No”.
• Secondary SIP proxy – defines the secondary SIP proxy by a domain nam
e,
an IP address
or an IP address together with a port number. The parameter is only visible if the
parameter Outbound Proxy mode is set to “No”.
• Outbound proxy – defines the primary outbound proxy by a domain name
,
an IP address
or an IP address together with a port number. The parameter is only visible if the
parameter Outbound Proxy mode is set to “Yes”.
• Listening port – the port that the handset l
i
stens to for incoming SIP traffic.
• SIP proxy ID – defines the SIP proxy by a domain name.
NOTE: This parameter is only needed when an
outbound proxy is defined. It may also be
used to specify a domain name when parameters Primary SIP proxy and Secondary SIP
proxy have been assigned IP-addresses.
• SIP proxy password
• Send DTMF using RFC 2833 or SIP INFO – this p
a
rameter defines which path the DTMF
signaling should take. If set to “RFC 2833“, the DTMF signaling will be sent in the RTP
stream, i.e. from handset to handset. If set to “SIP INFO”, the DTMF signaling will be sent
using SIP signaling, i.e. via the PBX.
• Hold type – defines type of hold to send when th
e handset
puts a call on hold. The
selection depends on what type of hold the PBX support. For more information about
what type of hold the PBX support, see the applicable documentation for the PBX.
• Registration identity – defines if the endpoint shall use it
s num
ber or ID for the
registration with the SIP proxy.
• Authentication identity – defines if the endp
oint shall use its number or ID for the
authentication with the SIP proxy.
• Call forward locally – when enabled the call forwarding is handled locally by the handset
instead of updating the PBX.
NOTE: The ha
ndset must be switched on and within coverage to handle this.
• MOH locally – Music on hold is played by the handset i.e. if the PBX does not supply MOH
the han
dset plays a tone when the call is on hold.
• Hold on transfer – puts a second call on hold before transfer, which is required by som
e
SIP proxy servers.
• Direct signaling – defines whether calls originating from other
sources than the
configured SIP Proxy should be accepted or redirected using “USE PROXY“ message.
6.9.3 Codec
A codec encodes a stream or signal for transmission. Codecs are often used in streaming
me
dia
applications. This setting defines how to packetize and compress the sound in a voice
call.
1 Select VoIP > General.