Asterisk phone Kphone

How to configure Asterisk for KPhone 4.0.1

Asterisk configuration

In this sample configuration, 192.168.0.1 represents the Asterisk server and 192.168.0.2 is the client running Kphone.

Configure Asterisk to accept registration and inbound calls in sip.conf like this:

[general]
…
disallow=all ; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
…

[kphone]
type=friend
host=dynamic
dtmfmode=inband
secret=blahblahblah

Important:
kphone handles DTMF in-band, and this only works if you force a 64 kbps codec, like ulaw or alaw

Configure kphone for the asterix server

File->Identity:
User part of SIP URL: kphone
Host part of SIP URL: 192.168.0.1
Authentication Username: kphone
Auto register: check

KPhone will ask to be restarted. After restarting it should register, and Asterisk’s console should display (if verbosity is high enough):

Registered SIP ‘kphone’ at 192.168.0.2 port 5060 expires 900

Now you can use kphone, complete with the DTMF support.

To dial your asterisk server (e.g. for voicemail) dial the extension of interest once you are registered. For example, “s” for the start extension, or 8500 for voicemail with the default extensions.conf.

Tóth Istvà¡n, 2004-03-13

Please note

This configuration does not take NAT into consideration. To do that, read the Asterisk FAQ on SIP and NAT.

Also, if running kphone and Asterisk on the same machine. Make sure you start Asterisk first or it will have problems binding to port 5060 if kphone is already running.

See Also

• Asterisk SIP channels
• KPhone

Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones