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| Local SIP port |
| Defines the local SIP port the GXW40XX will listen and transmit. The default value for |
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| Profile 1 is 5060 and 6060 for Profile 2. |
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| Local RTP Port |
| Defines the local |
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| base RTP port for channel 0. When configured, channel 0 will use this port _value for |
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| RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP, |
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| port_value+3 for its RTCP and so on. The default value for Profile 1 is 5004 and 6004 for |
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| Profile 2. |
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| Use random port |
| Default is No. If set to Yes, the device will pick |
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| This is usually necessary when multiple GXW40XX/HT50X 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 |
| transferred target’s Contact header information. |
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| Transfer on |
| Default is No. In which case if conference originator hangs up the conference will be |
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| Conference |
| terminated. When option YES is chosen, originator will transfer other parties to each |
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| Hang up |
| other so that B and C can choose either to continue the conversation or hang up. |
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| Enable |
| 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 |
| 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 |
| Default is No. If set to Yes, the Outbound Proxy will be removed from the route header. |
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| Route Header: |
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| Support SIP Instance |
| Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP |
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| ID |
| 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 |
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| message |
| to RFC rules. If message does not pass validation process, call will be rejected. |
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| Check SIP user ID for |
| Default is No. Check the SIP User ID in Request URI. If they don’t match, the call will |
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| incoming INVITE |
| be rejected. |
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| Allow Incoming SIP |
| Default is No. If incoming SIP message does not match with SIP Server, it will be |
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| Messages from SIP |
| rejected. |
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| Proxy Only |
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| SIP T1 Timeout |
| T1 is an estimate of the |
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| If the network latency is high, select larger value for more reliable usage. |
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| SIP T2 Interval |
| Maximum retransmission interval for |
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| DTMF Payload Type |
| Sets the payload type for DTMF using RFC2833. |
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| Preferred DTMF |
| The GXW40xx supports up to 3 different DTMF methods including |
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| method (in listed |
| (RFC2833) and via Sip Info. The user can configure DTMF method in a priority list. |
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| order) |
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| Disable DTMF |
| Default is No. If set to yes, use above DTMF order without negotiation |
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| Negotiation |
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| Send Hook Flash |
| Default is No. If set to yes, flash will be sent as a DTMF event. |
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| Event |
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| Enable Call Features |
| Default is Yes. (If Yes, call features using star codes will be supported locally) |
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| Proxy Require |
| SIP Extension to notify SIP server that the unit is behind the NAT/Firewall. |
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| Grandstream Networks, Inc. | GXW40XX User Manual | Page 30 of 38 | ||||
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| Firmware 1.0.4.2 | Last Updated: 06/2011 |