| SIP User ID |
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| User account information, provided by VoIP service provider (ITSP), usually has the |
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| form of digit similar to phone number or actually a phone number. This field contains |
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| the user part of the SIP address for this phone. e.g., if the SIP address is |
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| sip:my_user_id@my_provider.com, then the SIP User ID is: my_user_id. |
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| Do NOT include the preceding “sip:” scheme or the host portion of the SIP address in |
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| this field. |
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| Authenticate ID |
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| ID used for authentication, usually same as SIP user ID, but could be different and |
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| decided by ITSP. |
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| Authentication Password |
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| Password for ATA to register to (SIP) servers of ITSP. Purposely left blank once saved |
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| for security. Maximum length is 25. |
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| Name |
| SIP service subscriber’s name which will be used for Caller ID display |
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| DNS mode |
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| One from the 3 modes available for “DNS Mode” configuration: |
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| One mode can be chosen for the client to look up server. |
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| The default value is “A Record” |
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| User ID is Phone Number |
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| If “Yes” is set, a “user=phone” parameter will be attached to the “From” header in SIP |
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| request |
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| SIP Registration |
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| This parameter controls whether the HT503 needs to send REGISTER messages to |
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| the proxy server. The default setting is “Yes”. |
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| Unregister on Reboot |
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| Default is No. If set to yes, the device will first send registration request to remove all |
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| previous bindings. Use only if proxy supports this remove bindings request. |
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| Outgoing Call w/o |
| This parameter allows users place outgoing calls even when not registered (if allowed |
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| Registration |
| by ITSP) but it’s unable to receive incoming calls. |
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| Register Expiration |
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| This parameter allows the user to specify the time frequency (in minutes) the |
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| HandyTone ATA refreshes its registration with the specified registrar. The default |
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| interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 |
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| days). |
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| Local SIP port |
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| This parameter defines the local SIP port the HT503 will listen and transmit. The default |
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| value for FXS port is 5060. |
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| Local RTP port |
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| This parameter defines the local |
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| is the base RTP port for channel 0. |
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| When configured, the FXS port will use this port _value for RTP and the port_value+1 |
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| for its RTCP. |
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| The default value for FXS port is 5004. |
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| Use Random Port |
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| Default is No. If set to Yes, the device will pick |
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| This is usually necessary when multiple HandyTone ATAs are behind the same NAT. |
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| Refer to Use Target |
| Default is No. If set to “Yes”, then for Attended Transfer, the |
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| Contact |
| the transferred target’s Contact header information. |
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| Transfer on conference |
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| Default is No. In which case if conference originator hangs up the conference will be |
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| hangup |
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| terminated. When option YES is chosen, originator will transfer other parties to each |
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| other so that B and C can choose either to continue the conversation or hang up. |
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| Enable |
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| Default is No, this will create a |
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| transfer the call upon receiving ring back tone. |
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| Disable Bellcore Style 3- |
| Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you |
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| Way Conference |
| need to dial *23 + second callee number. |
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| Remove OBP from Route |
| Default is No. If set to Yes, the Outbound Proxy will be removed from the route header. |
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| Header |
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| Support SIP instance ID |
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| Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP |
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| Instance ID as defined in IETF SIP Outbound draft. |
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| Validate incoming SIP |
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| Default is No. If set to yes all incoming SIP messages will be strictly validated |
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| message |
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| according to RFC rules. If message will not pass validation process, call will be |
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| rejected. |
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| Check SIP User ID for |
| Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the |
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| Grandstream Networks, Inc. |
| HT503 User Manual | Page 25 of 38 |
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| Firmware 1.0.4.2 | Last Updated: 06/2011 |
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