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| DNS Mode |
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| One from the 3 modes are 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 |
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| If the HT502 has an assigned PSTN telephone number, this field should be set to |
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| Number |
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| “Yes”. Otherwise, set to “No”. If set to “Yes”, a | “user=phone” parameter will be |
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| appended to the “From” header in SIP request. |
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| SIP Registration |
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| Controls whether the HT502 needs to send REGISTER messages to the proxy server. |
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| The default setting is Yes. |
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| Unregister on Reboot |
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| Default is No. If set to Yes, the SIP user’s registration information will be cleared on |
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| reboot. |
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| Outgoing Call without |
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| Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if |
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| Registration |
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| allowed by Internet Telephone Service Provider) but is unable to receive incoming |
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| calls. |
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| Register Expiration |
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| This parameter allows the user to specify the time frequency (in minutes) the HT502 |
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| refreshes its registration with the specified registrar. The default interval is 60 minutes |
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| (or 1 hour). The maximum interval is 65535 minutes (about 45 days). |
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| Registration Retry Wait |
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| Retry registration if the process failed. Default is 30 seconds. |
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| Time |
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| Local SIP port |
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| Defines the local SIP port the HT502 will listen and transmit. The default value for FXS |
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| port 1 is 5060. The default value for FXS port 2 is 5062. |
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| Local RTP port |
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| Defines the local |
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| RTP port for channel 0. When configured, |
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| channel 0 uses this port _value for RTP and the port_value+1 for its RTCP; channel 1 |
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| uses port_value+2 for RTP and port_value+3 for its RTCP. |
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| The default value for FXS port 1 is 5004. The default value for FXS port 2 is 5012. |
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| Use Random Port |
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| This parameter forces the random generation of both the local SIP and RTP ports when |
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| set to Yes. This is usually necessary when multiple HT502 are behind the same NAT. |
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| Refer to Use Target |
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| Default is NO. If set to YES, then for Attended Transfer, the |
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| Contact |
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| transferred target’s Contact header information. |
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| Transfer on Conference |
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| Default is No. In which case if the conference originator hangs up the conference will |
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| Hang up |
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| be terminated. When option YES is chosen, originator will transfer other parties to |
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| each other so that B and C can choose either to continue the conversation or |
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| 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 |
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| Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you |
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| need to dial *23 + second callee number. |
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| Remove OBP from |
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| Default is No. When option YES is chosen, the Out Bound Proxy will be removed from |
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| Route Header |
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| Route 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 |
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| Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the |
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| incoming INVITE |
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| call will be rejected. If this option is enabled, the device will not be able to make direct |
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| IP calls. |
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| Grandstream Networks, Inc. |
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| Page 24 of 32 | |||||
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| Firmware Version 1.0.4.2 | Last Updated: 06/2011 |