Tap here to compare the top VoIP providersTap here to hide the top VoIP Providers
Asterisk phone Kphone
How to configure Asterisk for KPhone 4.0.1
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:
disallow=all ; Disallow all codecs
allow=ulaw ; Allow codecs in order of preference
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
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 István, 2004-03-13
Please noteThis 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.
Asterisk | Asterisk Configuration | Channel Configuration | Configuration for Specific Phones
Please update this page with new information, just login and click on the "Edit" or "Discussion" tab. Get a free login here: Register Thanks! - Find us on Google+