User manual
 FIRMWARE VERSION 1.0.14.1   HT503 USER MANUAL  Page 42 of 64  
Local SIP port 
This parameter defines the local SIP port the HT503 will listen and transmit. The default 
value for FXS port is 5060. 
Local RTP port 
This parameter defines the local RTP port pair used by the HandyTone ATA. The 
default value for FXS port is 5004. 
Use Random Port 
Default is No. If set to Yes, the device will pick randomly-generated SIP and RTP ports. 
This is usually necessary when multiple HandyTone ATAs are behind the same NAT. 
Hold Target Before Refer 
Default is Yes. Allows user to hold the phone call before refer it. If set to No, the call 
will not be hold before referred. 
Refer to Use Target 
Contact 
Default is No. If set to “Yes”, then for Attended Transfer, the “Refer-To” header uses the 
transferred target’s Contact header information. 
Transfer on conference 
hangup 
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. 
Disable Bellcore Style 3-
Way Conference 
Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you 
need to dial *23 + second callee number. 
Remove OBP from Route 
Header 
Default is No. If set to Yes, the Outbound Proxy will be removed from the route header. 
Support SIP instance ID 
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. 
Validate incoming SIP 
message 
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. 
Check SIP User ID for 
incoming INVITE 
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. 
Authenticate incoming 
INVITE 
Default  is  No.  If  set  to  Yes,  device  will  challenge  the  incoming  INVITE  for  the 
Authenticate ID and Password with 401 Unauthorized. 
Allow Incoming SIP 
Messages from SIP 
Proxy Only 
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. 
Use Privacy Header 
If  set  to  Default,  it  will  only  add  Privacy  or  PPI  header  when  special  feature  is  not 
Telkom SA or CBCOM. 
Use P-Preferred-Identity 
Header 
If  set  to  Default,  it  will  only  add  Privacy  or  PPI  header  when  special  feature  is  not 
Telkom SA or CBCOM. 
SIP T1 Timeout 
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. 
SIP T2 Interval 
Maximum retransmission interval for non-INVITE requests and INVITE responses. 
DTMF Payload Type 
This parameter sets the payload type for DTMF using RFC2833 
Preferred DTMF method 
The HT503 supports up to 3 different DTMF methods including in-audio, via RTP 










