MAXINCOM MUC1004 Administrator's Manual page 78

DNS SRV Look Up
Maxexpiry
Minexpiry
Defaultexpiry
Qualifyfreq
Qualifygap
Register Timeout
Register Attempts
RTPtimeout
RTPholdtimeout
RTPkeepalive
Notifyringing
Notifyhold
Session -timers
Session-refresher
Session-expires
Session-minse
DTMF mode
Relaxdtmf
Trustrpid
Sendrpid
Contactdeny
Contactpermit
Canreinvite
Audioprefcodec
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MUC1004/2008/2016 Administrator guide
NOT FOUND". It‟s recommended to be enabled for
security.
Please enable this option when your SIP trunk contains
more than one IP address.
Maximum duration (in seconds) of a SIP
registration.Default is 3600 seconds.
Minimum duration (in seconds) of a SIP registration.
Default is 60 seconds.
Default Incoming/Outgoing Registration Time: Default
duration (in seconds) of incoming/outgoing registration.
How ofen to check for the host to be up in seconds and
reported in milliseconds with sip show settings.
Number of milliseconds between each group of peers
being qualified.
Number of seconds to wait for a response from a SIP
registrar before timed out. Default is 20 seconds.
The number of SIP REGISTER messages to send to a
SIP Registrar before giving up. Default is 0 (no limit).
Terminate call if set # seconds of no RTP or RTCP activity
on the audio channel when we‟re not on hold.
Both ends of the call time
Time of packaging
Control whether subscriptions already INUSE get send
RINGING when another call is sent.
Notify subscriptions on HOLD state.(default:no)
Enable session-timer mode, default: yes. If you found
the call is cut off every 15 minutes every time, please
disable this.
Choose session-refresher, the default is Uas
The max refresh interval
The min refresh interval, which mustn't be shorter than
90s.
Set default mode for sending DTMF. Default setting:
rfc2833
Relax dtmf handing
If Remote-Party-ID should be trusted
If Remote-Party-ID should be sent
Use contactpermit and contactdeny to restrict at what IPs
your users may register their phones.
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
Once enabled,When the caller call out via SIP/SPS
trunks,the audio codec of calling channel whould be
selected in preference.
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