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Refer to Use Target
Contact
Transfer on conference
hangup
Disable Bellcore Style 3-
Way Conference
Remove OBP from Route
Header
Support SIP instance ID
Validate incoming SIP
message
Check SIP User ID for
incoming INVITE
Authenticate incoming
INVITE
Allow Incoming SIP
Messages from SIP
Proxy Only
Use Privacy Header
Use P-Preferred-Identity
Header
SIP T1 Timeout
SIP T2 Interval
DTMF Payload Type
Preferred DTMF method
(in listed order)
Disable DTMF
Negotiation
Send Flash Event
Enable Call Features
Offhook Auto-Dial
FIRMWARE VERSION 1.0.12.4
This is usually necessary when multiple HandyTone ATAs are behind the same NAT.
Default is No. If set to "Yes", then for Attended Transfer, the "Refer-To" header uses the
transferred target's Contact header information.
Default is No. In which case if conference originator hangs up the conference will be
terminated. When option YES is chosen, originator will transfer other parties to each
other so that B and C can choose either to continue the conversation or hang up.
Default is No. you can make a Conference by pressing 'Flash' key. If set to Yes, you
need to dial *23 + second callee number.
Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.
Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP
Instance ID as defined in IETF SIP Outbound draft.
Default is No. If set to yes all incoming SIP messages will be strictly validated
according to RFC rules. If message will not pass validation process, call will be
rejected.
Default is No. Check the incoming SIP User ID in Request URI. If they don't match, the
call will be rejected. If this option is enabled, the device will not be able to make direct
IP calls.
Default is No. If set to Yes, device will challenge the incoming INVITE for the
Authenticate ID and Password with 401 Unauthorized.
Default is No. Check the incoming SIP messages. If they don't come from the SIP
proxy, they will be rejected. If this option is enabled, the device will not be able to make
direct IP calls.
If set to Default, it will only add Privacy or PPI header when special feature is not
Telkom SA or CBCOM.
If set to Default, it will only add Privacy or PPI header when special feature is not
Telkom SA or CBCOM.
T1 is an estimate of the round-trip time between the client and server transactions.
If the network latency is high, select larger value for more reliable usage.
Maximum retransmission interval for non-INVITE requests and INVITE responses.
This parameter sets the payload type for DTMF using RFC2833
The HT503 supports up to 3 different DTMF methods including in-audio, via RTP
(RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.
Default is No. If set to yes, use above DTMF order without negotiation
Default is No. If set to yes, flash will be sent as DTMF event.
Default is Yes. (If Yes, call features using star codes will be supported locally)
This parameter allows users to configure a User ID or extension number to be
HT503 USER MANUAL
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