8-Port H323/SIP VoIP Gateway
8-Port H323/SIP VoIP Gateway 8-Port H323/SIP VoIP Gateway
4.2.5.4 RTP Packet Summary /VoIP Setup/Advance Setup/RTP Packet Summary On this page, user will know the RTP package summary about last VoIP call. Line#: number of line Using CODEC: ex.: G.723.1, G.729a Source IP: Remote side IP Source Port: Remote side port Packet Interval: interval time between 2 packets.(ms) Packet Send: number of packets sent. Packet Received: number of packets received. Packet Lost: number of lost packets. 66
4.2.5.5 Flash & Call waiting /VoIP Setup/Advance Setup/Flash & Call waiting On this page, user can define the parameters relative to the FLASH key and Call Waiting function. These functions usually work with PBX Token for flash key on VoIP(!): Define the token for flash key during VoIP protocol ( use “! “ by default). Flash Signal generate length : Define the pause time (ms) for one “,” symbol at /Routing Setting/VoIP Call Out/. This pause till is useful for PBX seize the trunk line from extension line. The default time is 1000ms, Input range from 100 to 3000ms. Flash Signal Detect Threshold: Define the threshold for valid FLASH signal. Only the flash time length between setting between min. to max. is accept by the gateway. Call waiting from PSTN when VoIP talking: Enable/Disable the Call Waiting function from PSTN line when talking by VoIP. Call waiting from VoIP when PSTN talking: Enable/Disable the Call Waiting function from VoIP when talking by PSTN line. 67
- Page 15 and 16: 4.2 Web UI Management The VoIP Gate
- Page 17 and 18: 4.2.2 VoIP Function 4.4.2.1 VoIP Se
- Page 19 and 20: 4.2.2.2 VoIP/Line Configure/ Line S
- Page 21 and 22: 4.2.2.3 Line configure/ Tone Settin
- Page 23 and 24: 4.2.2.4 Line configure/ Line Featur
- Page 25 and 26: Line number Priority: The 1 st line
- Page 27 and 28: 4.2.2.6 Routing Setup/ VoIP Call Ou
- Page 29 and 30: . Remark: Remark for this routing r
- Page 31 and 32: user press the number is 8862123456
- Page 33 and 34: Port Message Display Port Type Stat
- Page 35 and 36: 1 Via_GK2 1 gk2 33Delete 2 GK2_3_1
- Page 37 and 38: 4.2.2.7 VoIP Call In Routing Table
- Page 39 and 40: Index Area Code Strip Prefix Maximu
- Page 41 and 42: Enable: During Talk, you can answer
- Page 43 and 44: 10.1.1.1/104, it mean it will forwa
- Page 45 and 46: A. Time & Digits wait for user The
- Page 47 and 48: 4.2.2.9 VoIP Routing Profile Settin
- Page 49 and 50: Please remember to press the modify
- Page 51 and 52: f. No Answer Sec. Defined the wait
- Page 53 and 54: 1.4.2.11 VoIP Authorization Setting
- Page 55 and 56: 4.2.3.2 Setup the Register Server
- Page 57 and 58: q. Conference ID: Some SIP Server r
- Page 59 and 60: FXO: Analog phone interface for con
- Page 61 and 62: It will indicate the status of link
- Page 63 and 64: 4.2.5.2 Listen Port /VoIP Setup/Adv
- Page 65: Silence Detection / Suppression: En
- Page 69 and 70: 4.2.5.7 QoS /VoIP Setup/Advance Set
- Page 71 and 72: 4.2.5.9 FoIP /VoIP Setup/Advance Se
- Page 73 and 74: Prompt voice for replace dial tone:
- Page 75 and 76: 4.2.6.2 Telnet & SNMP /VoIP Setup/A
- Page 77 and 78: the most update information. Cable/
- Page 79 and 80: LAN MAC Address Short for Media Acc
- Page 81 and 82: 1.2.7.3 Date & Time This page is us
- Page 83 and 84: 1.2.7.5 System Log The system log p
- Page 85 and 86: 4.2.8 Route Function(/System Setup)
- Page 87 and 88: Subnet Mask Set the subnet mask of
- Page 89 and 90: dropped. D. DNS This page sets the
- Page 91 and 92: IP Pool Ending Address The ending a
- Page 93 and 94: 4.2.8.3 NAT Virtual Server This is
- Page 95 and 96: The Port Mapping page does a port r
- Page 97 and 98: DMZ Check this item to enable DMZ s
- Page 99 and 100: scans all the TCP/UDP port of a sta
- Page 101 and 102: URL Filtering Enable URL Filter Che
- Page 103 and 104: 4.2.8.5 Routing Routing Table This
- Page 105 and 106: 4.2.8.6 UPnP Settings UPnP Enable U
- Page 107 and 108: 4.2.8.7 DDNS Enabled Check this ite
- Page 109 and 110: 4.2.9.2 VoIP Module /System Mainten
- Page 111: 4.2.10 Save Modification /Save Modi
4.2.5.4 RTP Packet Summary<br />
/<strong>VoIP</strong> Setup/Advance Setup/RTP Packet Summary<br />
On this page, user will know the RTP package summary about last <strong>VoIP</strong> call.<br />
Line#: number of line<br />
Using CODEC: ex.: G.723.1, G.729a<br />
Source IP: Remote side IP<br />
Source <strong>Port</strong>: Remote side port<br />
Packet Interval: interval time between 2 packets.(ms)<br />
Packet Send: number of packets sent.<br />
Packet Received: number of packets received.<br />
Packet Lost: number of lost packets.<br />
66
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