Manuals / LevelOne / Communications / IP Phone

LevelOne VOI-7010, VOI-7011 user manual Understanding the SIP

Models: VOI-7011 VOI-7010

1 110

Download 110 pages 51.91 Kb

Page 59

Understanding the SIP

SIP, the Session Initiation Protocol, is a signalling protocol for Internet conferencing, telephony, presence, events notification and instant messaging. SIP was developed within the IETF MMUSIC (Multiparty Multimedia Session Control) working group, with work proceeding since September 1999 in the IETF SIP working group.

SIP enabled PBXes and/or SIP User Agents utilize the Session Initiation Protocol(SIP) to interconnect and to establish voice sessions between each other over an IP Network. SIP Telephony has emerged as a viable alternative to legacy (TDM) and fixed-line circuits for the establishment and transmission of voice communications.

53

Image 59

LevelOne VOI-7010, VOI-7011 user manual Understanding the SIP

Contents

LevelOne Page CE Declaration of conformity SafetyTable of Contents LCD Display and Keypad 103 101Introduction Features Optional Packing ContentsPage LCD Display and Keypads Hardware DescriptionVOI-7011 Front PanelVOI-7010 Memory Card Optional Headset Connection DiagramPower Installation IP Address 192.168.1.100 LAN Default SettingReset to Default Page Web Configuration System Information Phone Book Example Speed Dial Setting Example Phone Setting Phone SettingNo Answer Call ForwardAll Forward Busy ForwardPage Sntp Simple Network Time Protocol SntpVolume Page Ringer DND Do Not Disturb Auto Answer PIN Code EnabledAuto Answer CounterDial Plan Dialling Prefix Example Symbol x equals 0,1,2,3,4,5,6,7,8,9Yes Drop Prefix Replace Rule 007, 5xxx+35xx+21xx Result Dial Now *xx+#xx+11x+xxxxxxxxPage Flash Time Call Waiting Soft-key Hot line Alarm Network NetworkNetwork Status WAN Settings IP, Dhcp Client, and PPPoE modes. This LAN ModelIP Type LAN Settings Ddns Setting Example 802.1Q/TAG Vlan SettingsVlan Packets 802.1P User PriorityDMZ Virtual Server Application Diagram Pptp Password Select On to enable Pptp functionEnter login user name Enter passwordSIP Setting SIP SettingsUnderstanding the SIP Service Domain Register Name Realm 1 ~ ActiveDisplay Name User NamePage Page Codec 723 5.3K Codec PriorityRTP Packet LengthPage Codec ID Other Settings Others OthersAuto Config FXO Port MAC Clone Tones Advanced CID Save Change User PasswordUpdate UpdateUpdate Firmware Page Auto Update Settings Default Setting Reboot LCD Display and Keypad Key Descriptions Keypad DescriptionsKey Name Descriptions Call History LCD MenuPhone Book Call forward Call settingAlarm Setting Date/Time settingAuto Dial Volume and GainRinger DNS Server WAN SetupLAN Setup NetworkService Domain SIP SettingsCodec RTP SettingNAT Transversal AdministratorApplication Example Calling/Answering ConfigurationsPstn Calling ApplicationsSIP-to-SIP Calling Callings SIP-to-PSTN Calling Page PSTN-to-SIP Calling Page Attendant Transfer Way Conference CallingBlind Transfer Call Hold Application Way Conference CallCall Waiting Application Direct IP to Direct IP Calling FreeWorld Dialup FWD SIP Settings Codec Setting Specification 102 Do not hear dial tone? Can not access web page?