ADK-450 Administrator guide
رصعیلو نادیم : سردآ
کلاپ قیاقش نابایخ نایناریا ژاساپ بنج
8
دحاو
1
: نفلت
1811
8888
121
: سکاف
1111
8888
121
WWW.KAVOSHPHONE.COM
AMIR@KAVOSHPHONE.COM
@ADKPBX
63 / 86
registration.Default is 3600 seconds.
Minexpiry
Minimum duration (in seconds) of a SIP registration.
Default is 60 seconds.
Defaultexpiry
Default Incoming/Outgoing Registration Time: Default
duration (in seconds) of incoming/outgoing
registration.
Qualifyfreq
How ofen to check for the host to be up in seconds and
reported in milliseconds with sip show settings.
Qualifygap
Number of milliseconds between each group of peers
being qualified.
Register Timeout
Number of seconds to wait for a response from a SIP
registrar before timed out. Default is 20 seconds.
Register Attempts The number of SIP REGISTER messages to send to a
SIP Registrar before giving up. Default is 0 (no limit).
RTPtimeout
Terminate call if set # seconds of no RTP or RTCP
activity on the audio channel when we’re not on hold.
RTPholdtimeout
Both ends of the call time
RTPkeepalive
Time of packaging
Notifyringing
Control whether subscriptions already INUSE get send
RINGING when another call is sent.
Notifyhold
Notify subscriptions on HOLD state.(default:no)
Session -timers
Enable session-timer mode, default: yes. If you
found
the call is cut off every 15 minutes every
time, please
disable this.
Session-refresher
Choose session-refresher, the default is Uas
Session-expires
The max refresh interval
Session-minse
The min refresh interval, which mustn't be shorter than
90s.
DTMF mode
Set default mode for sending DTMF. Default setting:
rfc2833
Relaxdtmf
Relax dtmf handing
Trustrpid
If Remote-Party-ID should be trusted
Sendrpid
If Remote-Party-ID should be sent
Contactdeny
Contactpermit
Use contactpermit and contactdeny to restrict at what
IPs your users may register their phones.
Canreinvite
Asterisk by default tries to redirect the RTP media
stream to go directly from the caller to the callee.Some
devices do not support this (especially if one of them is
behind a NAT). The default setting is YES
Audioprefcodec
Once enabled,When the caller call out via SIP/SPS
trunks,the audio codec of calling channel whould be
selected in preference.
usereqphone
This provider requires,User=phone on URI
User agent
To change the user agent parameter of asterisk,