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| If the NAT Traversal field is set to “Yes” with no specified STUN server, the DP715 will |
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| periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to |
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| the SIP server to keep the “hole” on the NAT open. |
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| Advance Configuration |
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| DNS Mode |
| 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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| Tel URI |
| The default setting is “Disabled”. If the phone has an assigned PSTN |
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| Number, this field should be set to “User=Phone” then a |
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| “User=Phone” parameter will be attached to the “From header” in the SIP request to |
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| indicate the E.164 number. If server supports TEL URI format, then this option needs to |
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| be selected. |
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| SIP Registration |
| Controls whether the DP715 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 |
| 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 |
| Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if |
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| Registration |
| allowed by Internet Telephone Service Provider) but is unable to receive incoming calls. |
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| Register Expiration |
| This parameter allows the user to specify the time frequency (in minutes) the DP715 |
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| refreshes its registration with the specified registrar. The default interval is 60 minutes (or |
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| 1 hour). The maximum interval is 65535 minutes (about 45 days). |
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| SIP Registration Failure |
| Retry registration if the process failed. Default is 20 seconds. |
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| Retry Wait Time |
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| Local SIP port |
| Defines the local SIP port the DP715 will listen and transmit. Default is 5060 for profile 1 & |
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| 6060 for profile 2. |
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| Local RTP port |
| Defines the local RTP port the DP715 will listen and transmit. It is the base RTP port for |
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| channel 0. When configured, |
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| channel 0 uses this port _value for RTP. Default is 5004 for profile 1 & 6004 for profile 2. |
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| Use Random Port |
| Default is No. This parameter forces the random generation of both the local SIP and RTP |
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| ports when set to Yes. This is usually necessary when multiple DP715 are behind the |
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| same NAT. |
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| Default is No. If set to YES, then for Attended Transfer, the |
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| Contact |
| transferred target’s Contact header information. |
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| Transfer on Conference |
| Default is No. In which case if the conference originator hangs up the conference will be |
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| Hang up |
| terminated. When option YES is chosen, originator will transfer other parties to each other |
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| so that B and C can choose to either continue the conversation or hang up. |
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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 need |
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| Way Conference |
| to dial *23 + second callee number. |
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| Remove OBP from |
| Default is No. When option YES is chosen, the Out Bound Proxy will be removed from |
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| Route Header |
| Route header. |
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| Support SIP Instance ID |
| 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 |
| Default is No. If set to yes all incoming SIP messages will be strictly validated according to |
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| message |
| RFC rules. If message will not pass validation process, call will be 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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| incoming INVITE |
| call will be rejected. If this option is enabled, the device will not be able to make direct IP |
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| calls. |
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Firmware version 1.0.0.8 | DP715/DP710 User Manual | Page 42 of 52 |