2~24-port VoIP Gateway User Guide
Contents
1. Outbound Proxy Server 0 0 00 Figure 63 SIP proxy settings STEP 2 Because the VIP 281GS have registered to IP PBX all the VoIP calls will send to IP PBX so that don t need to set the dial plan settings Outgoing Dial Plan maximun 50 entries maximun length of prefix digits is 16 digit maximun length of number is 20 digit Length iS Outgoing no a oo Prefix no Destination IP DNS estnation Mlle Ta Number g FLAI Ss Outbound Dial Plan E From Tel 1 Figure 64 Outgoing dial sah settings STEP 3 Please browse to the GSM Dial plan menu and set the GSM outgoing number the sample configuration screen is shown 50 Routing Configuration GSM Routing Table Call Service route by GSM network According to the prefix of dialed number on FAS interface you can Route the calls to GSM Network wo aan E L E EB Lo LL el EJ uu Pj Figure 65 GSM Routing settings STEP 4 Repeat the same configuration steps on VIP 281GS B Test the scenario A ext 501 call to GSM 3 1 Ext 501 picks up the telephone and input the trunk code 8 to connect with FXS port of VIP 281GS A 2 Dial the GSM number 09125566 to establish the voice communication wi2. H 323 SIP VoIP GSM Gateway VIP 281GS User s manual Version 1 0 0 Copyright Copyright C 2007 PLANET Technology Corp All rights reserved The products and programs described in this User s Manual are licensed products of PLANET Technology This User s Manual contains proprietary information protected by copyright and this User s Manual and all accompanying hardware software and documentation are copyrighted No part of this User s Manual may be copied photocopied reproduced translated or reduced to any electronic medium or machine readable form by any means by electronic or mechanical Including photocopying recording or information storage and retrieval systems for any purpose other than the purchaser s personal use and without the prior express written permission of PLANET Technology Disclaimer PLANET Technology does not warrant that the hardware will work properly in all environments and applications and makes no warranty and representation either implied or expressed with respect to the quality performance merchantability or fitness for a particular purpose PLANET has made every effort to ensure that this User s Manual is accurate PLANET disclaims liability for any inaccuracies or omissions that may have occurred Information in this User s Manual is subject to change without notice and does not represent a commitment on the part of PLANET PLANET assumes no responsibility for any inaccuracies that may be conta
3. Client Support The router has a built in PPPoE client for establishing a DSL link connection with the ISP There is no need to install a further PPPoE driver on computers e Smart QoS The smart QoS provides stable voice quality while users access internet from private LAN to internet at the same time This device would start suppressing throughput automatically when VoIP call was proceeded and it keep full speed access when there is no VoIP traffic e DDNS Dynamic Domain Name Server DDNS is a service that maps Internet domain names to IP addresses It allows you to provide Internet users with a domain name instead of an IP Address to access Virtual Servers e NAT Traversal The NAT traversal allows gateway to operate behind any NAT Firewall device There is no need to change any configuration of NAT Firewall like setting virtual server VoIP Features e H 323 SIP dual mode communication e SIP 2 0 RFC3261 H 323v4 compliant e Peer to Peer H 323 GK SIP proxy calls e PSTN lifeline support e Voice codec support G 711 A law u law G 729 AB G 723 6 3 Kbps 5 3Kbps e Voice processing Voice Active Detection DIMF detection G 165 G 168 compliant echo canceller silence detection e Built in adaptive buffer that helps to smooth out the variations of delay jitter in voice traffic e Voice channels status display This function displays each port status such as on hook off hook calling number talk duration codec
4. If user does not have Gatekeeper please go to H 323 Dialing Plan Policy for more understandings H 323 Parameter Setting Q 931 Port Adjustment H 323 Call Pass Through NAT Configuration NAT Pass Method Disable Auto Pass Manual Need Key In Public IP STUN Public IP Address 0000 0 0 Figure 25 H 323 parameter setting 25 H 323 Parameters Label Primary Gatekeeper IP Address Secondary Gatekeeper IP Address Primary Gatekeeper Domain Name Secondary Gatekeeper Domain Name H 323 Gatekeeper ID Voice Cap Prefix RAS Port Adjustment Q 931 Port Adjustment Sets the unique name of this Gateway that is communicated as part of H 323 messaging There are two gatekeeper address fields one is primary the other secondary If this gateway does not want to register to any gatekeeper just set value 0 to the primary gatekeeper address If the primary gatekeeper address is not O the gateway will register to the primary gatekeeper If the second gatekeeper is not O the gateway will try to register to the second gatekeeper when failed to register to primary gatekeeper i e if both the primary gatekeeper and second gatekeeper Let user use Domain Name of H 323 Gatekeeper The Gatekeeper ID usually do not need to set this field unless the gatekeeper must need this value Let user set prefix number in RRQ nonstandard voicecap entry In H 323 standard the RAS default port number
5. SIP Protocol SIP number username and Password Setting Please fill out the SIP account including username password from ITSP Note Support digits and character base SIP Account username some SIP Server use character username to login and a number to call number ie VolPBuster if your servers don t support this number Account is the same please input the same username and now only support digits type for SIP number username VoIP Basic Configuration VoIP Protocol Setting F i Port Number Password Setting MAX 20 digit e ee a LJ Ll Use Public Account PORT 1 O Enable 9 Disable Figure 34 Port number setting Port Number Password Setting Input SIP number Username if your server support account and number different input the number else number account are the same username mes tetyoursiacsountregsterSIPSenercdiokWsoplon Input SIP account Username if your server support account and number different input the number else number account are the same username Input Password that ITSP support This allows gateway can use single SIP account for multiple Use Public Account ports User input the only one account in port one field for registering the ITSP Table 21 Network Advance descriptions 33 SIP Hunting Table This allows gateway can answer SIP call from internet by Hunting For example Port 1 and port 2 is hunting for the port 1 SIP account If
6. Disable O Enable 9 Disable Delay Time o Calling Number FKS Battery Reverse O Enable Disable Talking Time Limit 0 mins GoM Frequency 900 1800 850 1900 CLI Presentation Ofisbl Enable 00000 CLI Detection Disable Enable Asterisk Answer Supervision GM Receive Cain l0db db O db 4db 2db Odb OSM Receive Ge Bio e re Ir GSM Transmit Gain S 30db 33db 436db 439db O 42db Figure 13 GSM parameter setting GSM parameter configuration PIN Code Protection Enable PIN Code protection When the calls come to FXS port it will call hot line number to Baby Call GSM automatically The period of talking time when the time ends a beep sound Talking Time limit will come out as a warning sound If disable this option the phone number of SIM card won t be CLI Presentation shown in the callee side Answer Supervision Support Battery Reverse Detection Its able to adjust the GSM Receive Gain range from 10db to GSM Receive Gain M Its able to adjust the GSM Transmit Gain range from 30db to GSM Transmit Gain me Table 9 GSM parameter descriptions PSTN Dialplan PSTN Route Numbers The numbers which are filled in the form will go through the PSTN line unconditionally You can use x as wild card Routing Configuration PSTN Routing Table Call Service route by PSTN network According to the prefix of dialed number on FXS interface you c
7. GSM Features e SMS Server for SMS sending and receiving e Worldwide GSM network usable 850 900 1800 1900 MHz e Supports GSM PIN code protection Package Content The contents of your product should contain the following items Voice Gateway VIP 281GS unit Power adapter GSM Antenna Quick Installation Guide User s Manual CD RJ 45 cable x 1 VV VV ON WV Physical Details The following figure illustrates the front rear panel of VIP 281GS series Networking amp Communication 3 PLANE GSM Internet Telephony Gateway se VIP 281GS O LNK In Use amp ACT Ringing Figure 1 Front panel of VIP 281GS a pa ce w Phone LINE RESET 12V DC EE U Figure 2 Rear panel of VIP 281GS Front Panel LED Indicators amp Rear Panels On GSM GW is powered ON Off GSM GW is powered Off Line is busy Ring Indication Line is not enabled On GSM Network is found and working properly Flashing Searching GSM Network Table 1 Front panel description of VIP 281GS The Default WAN IP is http 172 16 0 1 Press RESET button L Note on rear panel over 5 seconds will reset the VoIP GSM Gateway to this default LAN WAN IP address and Username Password function Phone Phone port was connected to your telephone sets or Trunk Line of PBX SM mh port which you can Insert SIM Card Connect to the network with an Ethernet cable This port allows your ATA to be connected to an Internet Access device e g rou
8. Please browse to the GSM Setup PSTN Dial plan menu and set the PSTN outgoing number the sample configuration screen is shown 47 Routing Configuration PSTN Routing Table Call Service route by PSTN network According to the prefix of dialed number on FXS interface you can Route the calls to PSTN Network Phone Number Ex l E m B uM Figure 59 PSTN Routing table STEP 4 Please browse to the GSM Dial plan menu and set the GSM outgoing number the sample configuration screen is shown Routing Configuration GSM Routing Table Call Serice route by GSM network According to the prefix of dialed number on FAS interface you can Route the calls to GSM Network k aE ETE Figure 60 GSM Routing table STEP 5 Repeat the same configuration steps on VIP 281GS B Test the scenario A FXS 1 call to GSM 4 1 FXS 1 pick up the telephone 2 Dial the ext 400 to GSM port of VIP 281GS B and get the dial tone 3 Dial the GSM number 09581122 to establish the voice communication with GSM 4 B GSM 3 call to FXS 2 1 GSM 3 dial the GSM number 09127788 to GSM 1 and get the dial tone 2 Dial the ext 300 to establish the voice communication with FXS 2 48 C FXS 1 call to PSTN 1 1 FXS 1 pick up the telephone 2 Dial the PSTN number 10125566 to establish the voice commu
9. not want to register to any gatekeeper just set value 0 0 0 0 to the primary gatekeeper address There is a SIP Proxy Server address fields If this gateway SIP Proxy Server IP does not want to register to any SIP Proxy Server just set addresses value O 0 0 0 to the sip proxy server address Table 7 Gatekeeper SIP proxy descriptions STEP 4 Outgoing Dialing Plan 15 The purpose of Outgoing Direct Call setting is to let user create a proprietary dialing plan when this Gateway is not registered to any H 323 Gatekeeper or any SIP Proxy Server This setting can also assign some dialing plan to local ports including prefix strip prefix addition Through this setting user can directly map a number to a specific gateway IP address Outgoing Dial Plan maximun 50 entries maximun length of prefix digits is 16 digit maximun length of number is 20 digit Length TEST lim Outgoing no of qe Prefix no Destination IP DNS n Number 3 Operation J LE I JL Tol_ DELETE N A Foon Incoming Dial Plan maximun 50 entries maximun length of prefix digits is 16 digit maximun length of number is 20 digit Incoming no Length of Number jme Prefix no Destination Operation El al CJ rxs CJ esu Inbound Dial Plan From To Figure 11 Dial plan setting In the Outgoing Dial Plan settings Leading Number is the lea
10. of VIP 281GS A are ext 100 FXS and ext 200 GSM the VoIP number of VIP 281GS B are ext 300 FXS and ext 400 GSM AN GSM 1 GSM 2 M e Ext 200 Ext 400 409127788 809583344 Am GSM 3 172 16 0 1 172 16 0 2 GSM 4 09125566 vip zsics p 09581122 VIP 281GS A ef PSTN PSTN_1 PSTN_2 10125566 Ext 100 Ext 300 420114433 Figure 56 Peer to Peer GSM topology 46 Machine configuration on the VIP 281GS STEP 1 Please log in VIP 281GS A via web browser browse to the Advance Setup VoIP Basic menu and set the VoIP number as 100 and 200 the sample configuration screen is shown below VoIP Basic Configuration VoIP Protocol Setting SP v Port Number Password Setting MAX 20 digit es N MNZZNNM cw E Figure 57 VoIP basic settings STEP 2 Please browse to the Dial Plan menu and add the outgoing dial plan for calling to VIP 281GS B the sample configuration screen is shown Outgoing Dial Plan maximun 50 entries maximun length of prefix digits is 16 digit maximun length of number is 20 digit Length ORENSE lun Outgoing no of qaem Prefix no Destination IP DNS podes Operation Number gt 300 I 3 Q None 172 16 0 2 5060 B so 3 3 o None 1721602 5060 ee eee NN l E Dur ETE From rol Figure 58 Outgoing dial plan settings STEP 3
11. period in order to VAD Voice Codec option save bandwidth If you use Asterisk please disable Silence Compression it maybe make you call disconnect The Codec is used to compress the voice signal into data packets Each Codec has different bandwidth requirement There are four kinds of Codec G 723 G 729AB G 711 u and G 711_A The default value is G 723 Setting Termination key to speed up VoIP dial Select or 7 to Dial Termination key Termination key The FXO provides wild and complex ac termination impedances for FXO AC Impedance Ring Frequency You can configure how long the Ring Frequency do you want to use mere oegencecse selection 38 Table 28 Telephone Advance descriptions e Network Advance Advance Setting Select Network Advance v Smart OS IS Enable Disable Do Mcr ME Downstream Kbps Bandwidth Control Upstream Kbps 18kbps 12kbps 10kbps 8kbps C743 Bandwidth C799 Bandwidth 40kbps 24kbps 19kbps 16kbps 15kbps 14kbps TOS Enable Disable Figure 42 Network Advance setting SIP Netwrok Advance Configuration If this function is enabled when VoIP call is occurred the other data will be automatically reduced traffic which across the internet in order to guarantee the voice bandwidth Setting G 723 G 729 voice compression size Quality and G 723 G 729 Bandwidth Packet size can adjust by you want Ta
12. your web browser and input http 172 16 0 1 to logon VoIP GSM gateway web configuration page VoIP gateway will prompt for logon username password admin 123 Connect to 192 168 0 1 E i Please input username passward User name Password Remember my password Figure 5 Login prompt of VIP 281GS 12 System Configuration GSM Setup Let you configure your GSM setting Advance Setup Advance Setu n si Let you configure advance features System Administration N System Administration View system information and management system information PLANET Networking amp Communication Figure 6 System configuration VIP 281GS Setup for Quick Start System Configuration After finishing the authentication the Main menu will display 3 parts of configuration please click Advance Setup to enter advance configuration 1 Network Setup WAN Port Type Setup For most users Internet access is the primary application The Gateway support the WAN interface for Internet access and remote access The following sections will explain more details of WAN Port Internet access and broadband access setup When you click WAN Setting from within the Advance Setup the following setup page will be show J PLANET Main Menu Reboot Logout WAN Port Type Configuration Static IP v Select WAN Type Setting iiec P Select IP Address Network Setup Dynamic DNS DNS 72
13. 16 0 1 Network Management _ Subnet Mask 25525500 Default Router 172 16 0 254 VolP Setup VoIP Basic lI LLILAA Dialing Plan Advance Setting Hot Line Setting Ee Port Status _ Figure 7 WAN setting 13 Three methods are available for Internet Access If you are a leased line user with a fixed IP address fill out the Fixed IP User following items with the information provided by your ISP IP Address check with your ISP provider Netmsk check with your ISP provider Default Gateway check with your ISP provider Table 4 WAN setting descriptions ADSL Dial Up User PPPoE Enable Some ISPs provide DSL based service and use PPPoE to establish communication link with end users If you are connected to the Internet through a DSL line check with your ISP to see if they use PPPoE If they do you need to select this item WAN Port Type Configuration Lerm m xf PPPoE gw Select Use PPPoE Authentication User Name a MAX 40 SSS characters Password OS mE MAX 40 characters WAN Type Setting m Se Password Get IP Address Get Default Router 172 16 0 1 172 16 0 254 Enter the User Name and Password required by your ISP Apply Figure 8 PPPoE enable se
14. Dial Complete Tone complete tone If you don t want to hear that tone you can disable it Default is enabling FXS Impedance The FXS provides 600 900 OHM impedances for selection You can configure how long the Ring Frequency do you want to Ring Frequency Ring Frequency You can configure how long the Ring Frequency do you want to use Table 19 Telephone Advance descriptions Network Advance Advance Setting Advance Soning Select Nevo Aiae v E emant QOS Enable Disable ene Downstream 512 Kbps Bandwidth Control Upstream Kbps GS Bandwidth IS 18kbps 12kbps 10kbps 8kbps Cr 129 Bandwidth E 40kbps 74kbps 19kbps 16kbps 15kbps 9 14kbps Enable Disable Figure 33 Network Advance setting H 323 Netwrok Advance Configuration If this function is enabled when VoIP call is occurred the other data will be automatically reduced traffic which across the internet in order to guarantee the voice bandwidth G 723 G 729 Setting G 723 G 729 voice compression size Quality and Packet Bandwidth size can adjust by you want Table 20 Network Advance descriptions VoIP Basic Configuration to SIP Protocol Gateway SIP support SIP RFC3261 SDP RFC2327 RFC2833 STUN RFC3489 Symmetric RTP outbound proxy ENUM RFC2916 and RTP RTCP SIP NAT pass through Function can support 80 NAT Firewall that you don t setting DMZ Virtual server in router or Firewall 32 Select
15. UDEIIDPRs ier tere ee teenage E acu ae eiu eer nner 19 GSM DA sinters T 19 S RAS Silio c c aa re ten re 20 TermuinadteBI3e E Tas i sou rue Dora r Put Or oneal quc d eo Coca Det qa a quM DUO d Es t RI OD eet 21 Onnie BIKEN uus circa aed tartigamuniatisita ordo bate ig ee Hue ibat eneoraaaseiiaeens 21 Chapter 5 Advance Setup isiecissccsiccoasvccvenssnacecanicaaeanteaiecssaveccnensnsenenaeecs 22 NetWork SED icio pP rei REI Gee RISE Po I Denn 22 Dynamic JOINS p EM 22 INP WHO KM ana See li eiea dioe atta hi eee f aee on dee det A 23 dll 23 VoIP Basic Configuration to H 323 protocol eese 24 Dialing Plan to 1 523 protOCol wis Sei iere atn ra Ede Hau inte Re Shia pu uU et ens 27 Advance Semne 1011 32 3 Protocol oii ei iei ED E Rp a b uS DIE eia hate aus 20 VoIP Basic Configuration to SIP Protocol eeeessseeeseeeeenereeeeennn 32 Dialis Planto SIP Protocol aei erts a Rete EEUU E MeenE E 35 Ady ance Sette to SIP Protocole a i 37 TOE Tg oe UT T I N 39 PO StI 40 GNADE O sisses Ea 41 System Administralloris iiuscu x coe ed aaa aaia 41 Manase ment E cocsecsenuceaseccdcause sans seen A A A REE 41 SAVE Gro n E CUATRO TODO a OO OO DOT 4 ZACCOSST OIUUL EEE AE EA I AE EE A E AE uh E A uei Ete E dedu 42 Set To Derault CON oUr dO ece ae Cep Dis EE Dupuis eate a ebon 0o dms S ue obe 42 System In
16. an Route the calls to PSTN Network Phone Number pm I Figure 14 PSTN dialplan setting For examples Emergent calls like 911 Zone Numbers like 02x the phone numbers start with 02 GSM Dialplan GSM Numbers The numbers which are filled in the form will go through GSM Network unconditionally You can use x as wild card 19 Routing Configuration GSM Routing Table Call Service route by GSM network According to the prefix of dialed number on FXS interface you can Koute the calls to GSM Network temp Phone Number Length Nu a Ed p Na l e a FIL E JEDEGEEEDE al d E lil Figure 15 GSM dialplan setting For examples 09x All telephone numbers start with 09 0919x All telephone numbers start with 0919 SMS Setup SMS Sending Configuration SMS Sending Table SMS sending Systemr Help User Send Short Message to specific moble number Sending Number SMS Content Figure 16 SMS sending setting SMS sending configuration Sending Number The telephone number which an short message is sent to SMS Content The SMS Content will be sent to the preset telephone number If the SMS text is blan
17. as a signal This selection could force the Gateway to use normal start mode default mode or fast start mode when establishing a VoIP call Many other gateways only support normal start mode enable this selection when it is necessary The default is disabled using fast start mode This selection could force the Gateway to use H 245 Tunneling when H 323 H 245 Tunneling establishing a VoIP call The default is disabled using fast start mode There are 2 choices for this setting Gateway means it will act as the H 323 Registration type l E l VoIP gateway Terminal means it will act as the IP phone terminal This command configures the number of seconds that the gateway should be considered active by the H 323 Gatekeeper The gateway H 323 RRQ TTL transmits this value in the RRQ message to the gatekeeper The default value is O When a VoIP call is incoming the Gateway will ring a specific phone set The H 323 call signaling part could be connected or alerting during this ringing period If this selection is enabled the H 323 signaling part is connected during the ringing period The benefit of this situation is that the remote side could hear the status of the H 323 Autoanswer specific port That is the remote side will hear ring back tone if the Gateway is really ringing the phone set If the phone set is busy the remote side will hear busy tone The disadvantage of this situation is that the H 323 connected time is
18. ateway a wide range of potential applications The VIP 281GS can be installed on a PBX trunk line to enrich its trunks GSM and VoIP routes The PBX is able to have voice communication to either VoIP or GSM environment by the least costs Meanwhile the VIP 281GS is designed for comfort ease of use with a sophisticated and satisfaction to customers The VIP 281GS not only inherits traditions of quality voice communications but the VIP 281GS also eliminates the human resource of VoIP network deployment With optimized H 323 SIP architecture the VIP 281SG is the ideal choices for P2P voice chat and ITSP cost saving solution but also provides network converting feature to translate the packet network into traditional PBX system With built in PPPoE DHCP DDNS clients up to 2 concurrent connections in VIP 281GS voice communications can be established from anywhere around the world The VIP 281GS comes with intuitive user friendly and powerful management interface web telnet that can dramatically reduce IT personnel resource and complete GSM VoIP deployment in a short time Plus remote management capability administrators can monitor machine network status or proceed maintenance trouble shooting service via Internet browser or telnet session Besides it provides voice channels status display and optimized packet voice streaming over managed and public Internet IP networks Network Features e Point to Point Protocol over Ethernet PPPoE
19. ble 29 Network Adavnce descriptions Hot Line Setting You can set hot line When the call incoming the hot line port it will call hot line number automatically The hot line calls the number via VoIP so you setting the hot line number must VoIP number Usually you want to incoming GSM calls transfer to FXS you only setting the GSM hot line to FXS number Port number Input FXS GSM wants to call hot line number The call will via VoIP so the number must be the VoIP number 39 Hot Line Number Setting Hotline Setting Port 2 number Hotline Delay Hotline Delay Time Maz 20 sec Disable Enable 3 a Figure 43 Hot line setting Port Status Each of port show status table You can view all port status Like on off hook caller callee IP duration and packet loss Port Status Display This selection will display concurrent call status of this gateway The status information of each voice channel includes codec dialing number and destination IP address The status is refreshed every 3 seconds Port Status Port Type Status Codec Direction Dial No Ix FXS onhook none none none HEN GSM ionhook none none X none Figure 44 Caller No Dest Source I OUT Duration none none 0 lo none nane 0 lo o Port status 40 Chapter 6 System Administrations Management Management Label You can save configuration and restart the gat
20. ding digits of the dialing number Min Length and Max Length is the min max allowed length you can dial Strip Length is the number of digits that will be stripped from beginning of the dialed number Prefix Number is the digits that will be added to the beginning of the dialed number Destination is the IP address of the destination Gateway that owns this phone number STEP 5 Finishing the Wizard Setup After completing configuration setup please press Save Configuration and Reboot hyperlinks to save the configuration and rebooting Gateway After 20 Seconds you could re login the Gateway 16 Chapter 4 GSM Setup In GSM Setup VIP 281GS provides user the major parts GSM function to configure GSM Setup Label GSM Parameter allows you to modify the option of GSM GSM Parameter network Users could apply any dial policy by setting Dial Plan to route the GSM Dialplan Calls to GSM Network Terminate Black List The numbers in the list can not call from VoIP to GSM Network Table 8 GSM setup descriptions GSM Setup GSM Parameter Set GSM parameters PSTN Dialplan Set PSTN Dial Plan GSM Dialplan Terminate Black List Set GSM Dial Plan Originate Black List SMS Setting Send SMS to Mobile Figure 12 GSM setup setting 17 GSM Parameter GSM Parameter Table Configuration GSM Parameter Table lo Enable Disable FIN Enable
21. er The leading digits of the dialing number It has two text fields need filled Min Length and Max Length is Length of Number l l the min max allowed length you can dial The number of digits that will be stripped from beginning of the Delete Length dialed number Prefix no 0 The digits that will be added to the beginning of the dialed number Table 17 Incoming dial plan descriptions Incoming Dial Plan maximun 50 entries maximun length of prefix digits is 16 digit maximun length of number is 20 digit en l Length af Delete 4 Register Be ee Iter Operatie Incoming no Muinber Lenath Prefix no Destination to GK peration rT 4 7171 WW e A A I C FAS O a OC lo n revel Tol Figure 29 Incoming dial plan setting 28 Scenario description Termination call to GSM for one shoot call GSM Port SIM card was connected to GSM Gateway and standby for incoming outgoing calls properly H 323 leading number 081x incoming and delete the first one digit 0 and call to GSM number Note 081x will be registered to H 323 Gatekeeper if Register to GK was enabled show as below E Panini Length of ERES T nenen o ases Register UNS Itera f Chperatic te Incoming na Number Length E Prefix no eee io GK piraton 081x 4 20 1 None esm J L L EC 9 9 n DELETE ere From Tol Figu
22. erver Server U software b A PC or Notebook witch connected to LAN port of gateway c Put the image firmware named FW VIP281GS_vxxx bin at the assigned folder in FTP Server For example FW VIP281GS v305 bin is version 3 0 5L Note Free FTP server 172 16 0 101 username XXXX password Xxxx Environment Architecture Gateway and FTP server are in Internet Lu port VIP Gateway Figure 66 Firmware upgrades topology 2 Upgrading Process a Notebook Telnet GSM GW open DOS mode gt C gt telnet 172 16 0 1 Default WAN port IP b Please insert login password 123 and select 4 Upgrade Software Login Welcome to UIP 281G5 GSM Gateway Cuersion 3 4 55 WAN Status Fixed IP NAT Mode UnIP Status SIP Direct Mode GSM Signal Level 971 dBm IGSM perator Ghunghua Telecom Advanced Setup System Administration Save Current Configurations Upgrade Software Ping Logout GSM Restart Please Select 1 6 Figure 67 Main menu C Please input IP address of FTP server like as 172 16 0 101 username xxxx passswd xxxx and image name FW VIP281GS v305 bin d Upgrade y n y then will write the firmware to flash e After writing flash Please reboot the Gateway f If the new firmware image was most different with the previous version please push the hardware reset bottom to set to default g If the GSM Gateway is in remote site please use WEB configuration to set to default Star
23. eway with the default ave SIONTIHUISHOR configuration or with the current running configuration Access Control Users can sets changes the administrator password Set to Default You can restart the VIP 281GS with the default configuration Display software version WAN Type VoIP status VoIP codec and System Information l l l phone interface and system information SNTP Simple Network Time Protocol configuration for SNTP Setting synchronizing gateway clocks in the global Internet VIP 281GS can send log information to Syslog Server by UDP ports Syslog Setting 514 The VIP 281GS supports packets capture and save the packets to Capture Packets your PC Table 30 Management descriptions 9 PLANET Main Menu Reboot Logout System Administration RM Save Configuration fi l Save current system configuration Access Control Access Control ee Set system administrator username and password eee M E Set to default Set to Default Set to default configuration System Information EGG e Display current system information L Syslog setting SNTP Setting Ca ptur re pa chet SNTP parameter setting Syslog Setting Syslog parameter setting Figure 45 Management setting Save Configuration This page allows you to click Save Configuration and Reboot to save configuration and begin to restart 4 Save and Reboot The system begi
24. ey can t pass NAT or one way talk happen try to open DMZ and virtual server 5060 port in router NAT Pass Setting NAT Pass Method Default use Symmetric RTP pass function Outbound Proxy Support Setting your Outbound Proxy server information Table 23 SIP proxy descriptions Dialing Plan to SIP protocol The Dialing plan needs setting when the user uses the method of Peer to Peer or registering SIP proxy server mode The SIP dialing plan has two kinds of directions Outgoing call out and incoming call in Outgoing Dial Plan Peer to Peer call mode Effective Registering to SIP Proxy Server Mode Effective Table 24 Dialing plan descriptions 35 Outgoing Dial Plan maximun 50 entries maximun length of prefix digits is 16 digit maximun length of number is 20 digit PN tem Outgoing na Pi eh Prefix no Destination IP DNS um Operation pe m ia gt DELETE Qutbound Dial Plan Ben m Figure 38 Outgoing dial plan setting In the Outgoing Dial Plan Configurations settings Maximum Entries 50 Outbound number The leading digits of the call out dialing number The number of digits that will be stripped from beginning of the Delete Length dialed number The IP address Domain Name of the destination gateway that Destination IP DNS owns this phone number Table 25 Outgoing dial plan descriptions Incoming Dial Plan maximun 50 entries maximun length of prefi
25. formation Display FUNCTION ccccccsseesseccceceeseeeeeeecceeceeeaaeseeeeeeeeeeeaas 43 SINTPSettne EUlicttOgr s obo te REED ped im MEO IO e a B p EON DM NEED 43 PY SILOS SOLUTIO iesus Hooded bap MM MEI MM EM A DM MEE M PENA RUE 43 C dpture packets Function Sue carb tela Daniae ote eei Deb obi uta eheu auc dese 44 APPEN A med 45 bioresreD iiie 45 CONCE DIS VOICE POU us netebiqdtoi de E A leid eed ie nd PIeEPd 45 Sample scenario 1 Peer to Peer GSM termination eseeeeeeeeeeeeeeeee 46 Sample scenario 2 Enterprise SIP GSM termination eeeeeeeeeeeeee 49 APPEND qi crue 52 FA O REESE 52 PRD DO INCI qi Oc r 54 Firmware upgrade Requirement and Process Lecce eee eee eee e eee eee eee eee 54 lege dp 56 VIP 238 TGS Specifica NOS em 56 Chapter 1 Introduction Overview With years of Internet telephony and router manufacturing experience PLANET proudly introduces the The PLANET VIP 281GS VoIP GSM Gateway is a signal GSM channel gateway that supports SIP and H 323 VoIP protocol at the same time The VIP 281GS provides a total solution for integrating voice data network and the Global System for Mobile Communications GSM The VIP 281GS is equipped with both FXS and PSTN interfaces which gives the g
26. ined in this User s Manual PLANET makes no commitment to update or keep current the information in this User s Manual and reserves the right to make improvements to this User s Manual and or to the products described in this User s Manual at any time without notice If you find information in this manual that is incorrect misleading or incomplete we would appreciate your comments and suggestions CE mark Warning The is a class B device In a domestic environment this product may cause radio interference in which case the user may be required to take adequate measures WEEE Warning To avoid the potential effects on the environment and human health as a result of the presence of hazardous substances in electrical and electronic equipment end users of electrical and electronic equipment should understand the meaning of the crossed out wheeled bin symbol Do not dispose of WEEE as unsorted municipal waste and have to collect such WEEE separately Trademarks The PLANET logo is a trademark of PLANET Technology This documentation may refer to numerous hardware and software products by their trade names In most if not all cases their respective companies claim these designations as trademarks or registered trademarks Revision User s Manual for PLANET H 323 SIP VoIP GSM Gateway Model VIP 281GS Rev 1 0 October 2007 Part No EM VIP281GSV1 TABLE OF CONTENTS Chapter Eee TIE E aa 6 ij rgerejB eii o mee sec ca
27. is 1719 The VoIP gateway provides user to change RAS port number to meet the network environment Some area carrier blocks or forbidden the default port number In H 323 standard the default Q 931 port number is 1720 The VoIP gateway provides user to change Q 931 port to meet the network environment Some area carrier blocks or forbidden the default port number H 323 Call Pass through NAT Sets the unique name of this Gateway that is communicated as part of H 323 messaging H 323 Pass Through NAT method 1 Disable The Gateway operates in public IP address 2 Auto Detection When the Gateway register to GNU Gatekeeper please select this option 3 Manual Setting When the Gateway registers to H 323 Gatekeeper and operate under NAT enable DMZ please select this option and key in IP address Table 14 H 323 parameter descriptions 26 Dialing Plan to H 323 protocol The Dialing plan needs setting when the user uses the method of Peer to Peer H 323 VoIP call or registering H 323 Gatekeeper mode The H 323 Dialing Plan has two kinds of directions Outgoing call out and Incoming call in Peer to Peer call mode Effective Outgoing Dial Plan Registering to H 323 Gatekeeper mode Effective Table 15 Dial plan descriptions In the Outgoing Dial Plan Configurations settings Maximum Entries 50 Outbound number The leading digits of the call out dialing number The number of digits that will be stri
28. k an empty SMS is sent The Maximum capacity is 40 characters Table 10 SMS sending descriptions 20 Terminate Black List Terminate black list The numbers in the black list will not be able to call from VoIP to GSM network Terminate Black List Setting Terminate Black List Terminate Black List The following number can not call from VoIP to GSM Network I Phone Number ft ________ Originate Black List PSTN Route Numbers The numbers which are filled in the form will go through the PSTN line Figure 17 Terminate Black setting unconditionally You can use x as wild card Originate Black List Setting Originate Black List lOriginate Black List The following number can not call from GSM Network to VolP Item Phone Number EB Lo E I _ T _ Figure 18 Originate Black setting 2 Chapter 5 Advance Setup Network Setup In Network Setup VIP 281GS provides user the major parts Network function to configure Network Setup Advance Setup WAN Setting WAN Setii Set WAN port network pa aters Dynamic DNS DNS NS E aa DDNS Setting SSCL sot bons cover P add
29. matically it is no need to WAM Type Setting configure the DHCP settings The ISP will assign PPPoE username password for Internet access Table 3 WAN port configuration descriptions Please consult your ISP personnel to obtain proper PPPoE IP address related information and input carefully Hint If Internet connection cannot be established please check the physical connection or contact the ISP service staff for support information 11 Chapter 3 Network Service Configurations Configuring and monitoring your VolP Gateway from web browser The VIP 281GS integrates a web based graphical user interface that can cover most configurations and machine status monitoring Via standard web browser you can configure and check machine status from anywhere around the world Overview on the web interface of VoIP GSM Gateway With web graphical user interface you may have More comprehensive setting feels than traditional command line interface Provides user input data fields check boxes and for changing machine configuration settings Displays machine running configuration To start VIP 281GS web configuration you must have one of these web browsers installed on computer for management Microsoft Internet Explorer 6 0 or higher with Java support Manipulation of VoIP GSM Gateway via web browser Log on VoIP GSM Gateway via web browser After TCP IP configurations on your PC you may now open
30. nding the DTMF tone as a RTP payload signal The third one is SIP Info that is sending the DTMF tone as a SIP signal Sending DTMF tone as a signal could tolerate more packet loss caused by the network If this selection is enabled the DTMF tone will be sent as a signal Adjust RFC2833 DTMF payload value range from 96 to 127 RFC2833 Payload default is 101 When your gateway shutdown or something happen that made Watchdog gateway can t work fine Watchdog will reboot your gateway automatically when it can t work Table 27 VoIP Advance descriptions 37 Telephone Advance allene Compression Voice Acmaty Detection Voice Codec Dial Corplete Tone Dial Tenmaination Key IFXE Impedance Phone In Volume Phone Chit Volume Line In Volume Line Cut Volume Ring Frequency L TMP tone power Advance Setting Select Telephone Advance v Select le VAD Enable VAD Disable G T23 1 6 3k O G 729AB G 711 y law G711 a law G Enable O Disable 0000000 CETE G60 O900 0 0 0 0 0 0 0 00m 0 3 db from 9 to 3 3 dbifrom 9 to 3 0 ib trom 9 to 8 4 b fom 9 to 8 Do uz Tdbm amp dbm 3dbm 1dbm Odbm 4 1dbm 3dbm 6dbm Figure 41 Telephone Advance setting SIP Telephone Advance Configuration If this function is enabled when silence is occurred for a period of time Silence Compression no data will be sent across the network during this
31. ne ng see sere cece acegeeecceseanee sev snmeceeaateocecenenenaccaness 6 OVGRVICW De 6 Package Conte Dt 5 icis het OE PUR OE E IRE II E 7 Physical Detalles tee erae eroe ove oor doen eate E E Qovvr er dase iva deve R 8 Front Panel LED Indicators amp Rear Panels tp Er err P Ea Ebene ore Pau ee 8 Chapter 2 Preparations amp Installation 10 Physical Installation Requirement eee eee eee ee e reete aN 10 WAN IP address configuration via web configuration interface 11 Chapter 3 Network Service Configurations 12 Configuring and monitoring your VoIP Gateway from web browser 12 Overview on the web interface of VoIP GSM Gateway eere 12 Manipulation of VoIP GSM Gateway via web browser esses 12 VIP 281GS Setup for Quick Start eee r rte pa eeu eoe iare pee po Nea e Vere a e i eee eaa PE eS PU ee eate PS 13 1 Network Setup WAN Port Type Setup cccccccccccccccsssseeeceeeeeeeeeeeeeeeeeeeeeeeeaas 13 2 NODIPIBOSIC SOUND 3 053 IM HIN INS IMa DIM M EIU M II M LA LARES 15 Chapter 4 GSM Setup iuis utc Orto re Fes vana aocieE Cu ev s ada ri d pr ex dus MESE REUE 17 GSM SEUD oiueinhecasedis E t Pr vite pn iiti 17 GSM Pranit siae rais ER irte i ees eiu oto hel sais ne Diu oho obud cito en ios 18 PSTN
32. nication with PSTN 1 Sample scenario 2 Enterprise SIP GSM termination In the following samples we ll introduce the SIP Proxy and GSM termination applications In this example there are two VIP 281GS the FXS and GSM ports are register to SIP Proxy Server IP PBX The out lines of PBX connect with Phone FXS ports of VIP 281GS The extensions of PBX can make GSM calls via GSM ports of VIP 281GS IP PBX 172 16 0 10 Internet b dts GSM 1 GSM 2 iiy Ext 200 Ext 400 E 4809127788 409583344 GSM 3 172 16 0 1 172 16 0 2 N GSM 4 09125566 a 09581122 S VIP SB y VIP 281GS A i IP 281GS E FXS 2 Phone dnos Ext 300 Trunk code 8 Trunk code 9 Ext 501 Ext 601 Figure 61 Enterprise GSM Routing table Machine configuration on the VIP 281GS STEP 1 Please log in VIP 281GS A via web browser browse to the Advance Setup VoIP Basic menu set the VoIP registration number as 100 200 and the registration server address the sample configuration screen is shown below 49 Port Number Password Setting MAX 20 digit ee es v i E 73 eee Success OK svo ee w JE m E e IE ll SUCCESS OK Figure 62 Port number settings SIP _ SIP Proxy Setting esses Setting BN I5B115QRD S SIP Prozy Server __ _ _ o vn use Net2Phone Service Register Interval seconds 100 ER LIC Enable O Disable
33. not the real voice call connected time So if billing is recorded for this Gateway this function should be disabled Some Gatekeeper register need UA send MAC address to Authentication you need enable this function Default is disable When your gateway shutdown or something happen that made Watchdog gateway can t work fine Watchdog will reboot your gateway automatically when it can t work Table 18 VoIP Advance descriptions 30 Telephone Advance Advance Setting Select Allene Compiessio Ue etn Aon 9 VAD Enable VAD Disable Voice Coder G 723 1 6 3k G 729AB G 711 y law G 711 a law Dial Complete Tone Enable Disable Dial Termination Key amp O0o FS Impedance 600 900 Phone In Volume S kibirom 9 to 3 Phone Out Yolume 3 ldb from 9 to 3 Line In Volume f hbom 918 0 20000 0 0 0 0o Line Cut Volume Fang Frequency B Je OL ALLLL l LIITT DTMF tone power 9 Tdbm 6dbm 3dbm 1dbm Odbm 1dbm 3dbm 6dbm Figure 32 Telephone Advance setting H 323 Telephone Advance Configuration If this function is enabled when silence is occurred for a period of time no data will be sent across the network during this period in Silence Compression VAD order to save bandwidth If you use Asterisk please disable Silence Compression it maybe make you call disconnect When you use the VoIP call you will hear DuDu voice that is dial
34. ns to save and reboot please wait a moment and relogin Figure 46 Save setting Access Control Changing the Administrator Guest Password For security reasons we strongly recommend that you set an administrator password for the router On first setup the router requires no password If you don t set a password the router is open and can be logged into and settings changed by any user from the local network or the Internet Click Access Control Setup the following screen will open Administrator username password admin 123 Guest username password guest guest Access Control Administrator Username and Password Username Confirm Password Username Confirm Password Figure 47 Access control setting Set To Default Configuration If you want to reboot the router using factory default configuration click Apply then reset the router s settings to default values Set to Default All configuration will be set to default setting Figure 48 Set to default setting 42 System Information Display Function Click System Information Display to open the Online Status page In the example on the foll owing page both PPPoE connections is up on the WAN interface H323 SIP Status MAC addr ess Register Status etc System Information Software Version 3 0 5L o WAN Type Fixed IP WAN MAC Address 00 0f fd 48 00 0c VoIP Status SIP Direct Mode VoIP Codec G723 1 GSM Signal Level 9 dBm GSM Operator M
35. o 192 168 0 1 Please input username passward User name v Password Remember my password Figure 3 Login prompt of VIP 281GS 10 Please locate your PC in the same network segment b Note 172 16 0 x of VIP 281GS If you re not familiar with TCP IP please refer to related chapter on user s manual CD or consult your network administrator for proper network configurations WAN IP address configuration via web configuration interface Execute your web browser and insert the IP address default 172 16 0 1 of VIP in the adddress bar After logging on machine with username password default admin 123 browse to WAN Setting configuration menu you will see the configuration screen below WAN Port Type Configuration 178 185 0 1 Subnet Mask e Eb 255 t Default Router 172 16 00 254 Figure 4 WAN port configuration Connection Type Data required StticIP The ISP will assign IP Address and related information Get WAN IP Address auto
36. odel Current system time Chunghwa Telecom GSM VolIP Gateway 0 0 0 00 00 00 Figure 49 System information SNTP Setting Function Click SNTP setting to open the Online Status page In the example on the following page Simple Network Time Protocol SNTP To synchronize Gateway clocks in the Internet Enable Disable MTF Server IP 133 100 3 2 re servers IP 131 107 1 10 192 5 41 209 GMT 08 00 Taipei a NTP Servers IP Time Zone Selecting Figure 50 SNTP setting Use SNTP Setting when checked gateway uses a Simple Network Time Protocol SNTP to set the date and time The gateway synchronizes the gateway s time after you select the time zone Use SNTP Setting select the time zone which gateway was at Syslog setting Use Syslog server to record your VIP 281GS log file To set the Syslog server IP address for this function Kindly please download for this FREE service at http www kiwisyslog com index php for more understandings 43 Syslog Server Configuration Syslog Server Setting yslog is a method to collect messages from devices to a server running a syslog daemon Lagging to a central syslog server helps in aggregation of logs and alerts VolP Gateway devices can send their log messages to a SYSLOG serice The Syslog messages including COR Call Detail Record and system parameters Mote Detault syslog port 514 slog Server Data Syslog Server Port Figu
37. ort you should find that e Phone port The Phone port allows the connection to an end node like telephone or out line of PBX system Phone port is as like your local phone service provider who provides a number to you It is easy to tell that after you have connected an end device to Phone port and you will hear the dial tone from Phone port once the hand set off hook 412 1111 Figure 54 Phone port topology The Phone port is with voltage and current DO NOT connects the port to any PBX extension line or PSTN line This may make the Phone port or your PBX Caution extension port malfunction e GSM port The GSM port allows can be inserted a SIM card that already has a fixed number say 0912 111111 So the only connections for GSM port will be to your local PSTN or GSM network With your GSM connect to GSM network the Internet Voice can then have a GSM call through this line number 0912 111111 Or locally you can have an Internet Call through the line 0912 111111 Your PBX users will need to know this number in the future Figure 55 GSM port topology Sample scenario 1 Peer to Peer GSM termination In the following samples we ll introduce the Peer to Peer GSM termination applications In this example there are two VIP 281GS calling by IP address directly both VIP 281GS have inserted the GSM SIM cards into SIM slots the GSM number are 09127788 GSM 1 and 09583344 GSM 2 The VoIP number
38. pped from beginning of the Delete Length dialed number The IP address Domain Name of the destination gateway that Destination IP DNS owns this phone number Table 16 Outgoing dial plan descriptions Outgoing Dial Plan maximun 50 entries maximun length of prefix digits is 16 digit maximun length of number is 20 digit rem Outgoing na ries wu Prefix no Destination RDNS Operation l oo n Figure 26 Outgoing dial plan setting Scenario description Normally dial 001x leading call out call to destination IP address 172 16 0 100 002x leading call out call to destination domain name h323gw test com 27 rem Outgoing na ace wes Frefix no Destination IP DNS Operation a 001x s n 0 Mone 172160100 P a DD2x am o Mone h323gw testcom MEME 8 mim Duc EE Fom To Figure 27 Outgoing dial plan setting Scenario description Speed dial If user dials 101 the gateway automatically dials 1234567890 to destination IP address 172 16 0 101 If user dials 202 the gateway automatically dials 0987654321 to destination IP address 172 16 0 202 rerl Outgoing na rac pen Prefix no Destination IP DINS Operation E 101 Sous 2 1234567890 72 16 0101 MAE E 202 3 3 3 0887554321 72 000 AME Ii o n DELE A Fom To Figure 28 Outgoing dial plan setting In the Incoming Dial Plan Configurations settings Maximum Entries 50 Inbound numb
39. re 30 Incoming dial plan setting Advance Setting to H 323 protocol In Advanced Setting VIP 281GS provides user three major parts function to configure One is VoIP Advance the other are Telephone Advance Network Advance and Tone Table Setting X Advance Setting Advance Setting Advance Setting Select VoIP Advance Y DTMF Relay for H 323 amp Outband by H 245 Inband by RTP Normal Start Fast Start H 323 H245 tunneling Enable 9 Disable IH 323 Rezistration Type G Gateway Terminal H 323 RRO TTL I kecmis CE REO Polling Peried j 3E REQ Folling Feri 120 iseconds IH 323 Anteans wer On Off MAC Authentication O Enable 9 Disable HETIOURCTUTISEET Enable 9 Disable Watchdog Disable Enable VoIP Encryption 9 Disable Enable VoIP Encryption Port E onn Figure 31 VolP Advance setting 29 H 323 VoIP Advance Configurtion After the VoIP call is connected when you dial a digit this digit is sent to the other side by DTMF tone There are two methods of sending the DTMF tone The first is in band that is sending the DTMF tone DTMF Relay for H 323 in the voice packet The other is out band that is sending the DTMF tone as a signal Sending DTMF tone as a signal could tolerate more packet loss caused by the network If this selection is enabled the DTMF tone will be sent
40. re 51 Syslog setting Free Syslog server Software Listen Port 514 Free Software download Kiwi syslog server aC Daemon Figure 52 Syslog topology Capture packets Function Use Capturer Packets to record VIP 281GS packets Users can start and stop the capture then save the file to PC Use the Ethereal Tool www ethereal com to analyze the packets To troubleshoot what is going on on the network level vou can generate PCAP files on this page These files can be read with Ethereal network tool Press the start button to start recording and press the stop button to stop Please remember that the data is stored in a 15KB buffer and that the recording may Be D jen p Qaem emm amc pe have a negative impact on the phone s performance TOTE ETE OF ARH EB pee peemm Dus gn Click here to save the current pcap trace 0 packets Q octets duration 0 seme Figure 53 Capture packets setting Appendix A Voice communications The chapter shows you the concept and command to help you configure your PLANET VIP 281GS through sample configuration And provide several ways to make calls to desired destination in VIP 281GS In this section we ll lead you step by step to establish your first voice communication via web browsers operations Concepts Voice port There are two type of the voice port Phone FXS Foreign exchange Station on the printing of the RJ 11 port and GSM on the printing of the SIM p
41. ress Network Management Set web server telnet server port Figure 19 Network setup setting Dynamic DNS DDNS is a service that maps Internet domain names to IP addresses DDNS serves a similar purpose to DNS DDNS allows anyone hosting a Web or FTP server to advertise a public name to prospective users Unlike DNS that only works with static IP addresses DDNS works with dynamic IP addresses such as those assigned by an ISP or other DHCP server DDNS is popular with home network who typically receive dynamic frequently changing IP addresses from their service provider To use DDNS one simply signs up with a provider and installs network software on their host to monitor its IP address DDNS Dynamic DNS Service Configuration Dynamic DNS allows you to provide Internet users with a domain name instead of an IP Address to ess your Virtual Servers Register for this FREE service at htfp Awww dyndns org DONS username DONS password DONS domain name DNS Server IP DONS Status DNS OK Figure 20 DDNS date setting 2 Three methods are available for Internet Access Input your DDNS User Name Input your DDNS Password Input you set from your DDNS DNS Server IP Input your DNS Server IP Table 11 DDNS date descriptions Netwrok Management Network Parameter allows you to modify the access port of gateway For example Setting HTTP port 80 and Setting TELNET port 23 Access Service Configuration HTTP Po
42. rt and TELNET Port Configuration Access Port Service Figure 21 Access port service setting VoIP Setup GSM Gateway support 2 VoIP protocol H 323 SIP you can register to H 323 Gatekeeper or SIP proxy server Gateway is not a softswitch it only can use 1 VoIP protocol SIP H 323 at the same time If you don t register GK or Proxy server you can make Peer to Peer call by IP address or domain name Setting Dialing plan In VoIP Setup VIP 281GS provides user the major parts VoIP functions to configure VoIP Setup Label The PLANET series gateway support 2 24 phone line for SIP and VoIP Basic H 323 VoIP call applications You can configure these ports from this menu Users could apply any dial policy by setting Dial Plan including Dialing Plan outgoing dial plan and incoming dial plan VIP 281GS support for silence compression DTMF Relay Codec Advanced Setting Selection FAX mode Option 23 H323 Register Type and H 323 Fast Start Normal Start function FXO AC impedance Volume Adjustment RRQ TTL RFC2833 Payload IP TOS etc Let user can set up hotline to dial the phone number Hot Line Setting l automatically Port Status Display the telephone interface status Table 12 VolP setup descriptions VoIP Basic __VolP Setup 7 Set VoIP basic parameters such as VoIP protocol selection phone number VoIP Basic 9 Dial Plan Dialing Plan Set outbound and inbo
43. st has NAT Pass through problem By SIP there are many NAT Pass through Function can solve 80 NAT Problem You can choose STUN Outbound Proxy oymmetric RTP to Pass through NAT you don t set any other setting DMZ Virtual Server by router side If you use STUN Outbound Proxy you must have a STUN Outbound Proxy Server to support If they can t pass NAT please open the DMZ Virtual Server by Router NAT Firewall Q6 Why does the one way talk happen A Generally one way talk happen when use the different codec between VoIP devices make call Please check and setting the same codec most one way talk will be solved Q7 Why can I call out by Gateway A Please chick your Gateway is registered SIP Proxy Server ITSP and chink your Internet works fine Gateway can t make a call without Internet or SIP Account that from ITSP supply You must have a SIP account or know the other Gateway IP Domain Name and then you can make a VoIP call Q8 Why I use asterisk by G 729 sometimes disconnect happen A In asterisk setting VAD must disable if you open Silence Compression VAD it will make call disconnect happen please disable the option when you use the asterisk Q9 Why can register and use after setting A After setting please save configuration and reboot after reboot you can use new configuration 53 Appendix C Firmware upgrade Requirement and Process 1 Environment Requirement a APC with FTP S
44. ter cable modem ADSL modem through a networking cable with RJ 45 connectors used on 10BaseT and 100BaseTX networks 12V DC Power The supplied power adapter connects here Table 2 Rear panel description of VIP 281GS Incorrectly connecting telephony devices to the RJ11 port Warning on the Telephony Interface can cause permanent damage to the VoIP Gateway Chapter 2 Preparations amp Installation Physical Installation Requirement This chapter illustrates basic installation of VIP 281GS series e Network cables Use standard 10 100Base TX network UTP cables with RJ45 connectors e TCP IP protocol must be installed on all PCs For Internet Access an Internet Access account with an ISP and either of a DSL or Cable modem for WAN port usage Administration Interface PLANET VIP 281GS provides GUI Web based Graphical User Interface for machine management and administration Web configuration access To start VIP 281GS web configuration you must have one of these web browsers installed on computer for management e Microsoft Internet Explorer 6 0 or higher with Java support Default WAN interface IP address of VIP 281GS is 172 16 0 1 You may now open your web browser and insert http 172 16 0 1 in the address bar of your web browser to logon VIP 281GS web configuration page VIP 281GS will prompt for logon username password please enter admin 123 to continue machine administration Connect t
45. th GSM 3 B ext 501 call to GSM 4 1 Ext 501 picks up the telephone and input the trunk code 8 to connect with FXS port of VIP 281GS A Dial the ext 400 to GSM port of VIP 281GS B and get the dial tone Dial the GSM number 09581122 to establish the voice communication with GSM 4 51 Appendix B FAQ Q1 What is the default administrator password to login to the gateway A By default your default username is admin default password is 123 to login to the router For security you should modify the password to protect your gateway against hacker attacks Note Default guest login username password guest guest Q2 forgot the administrator password What should I do A Press the Reset button on the rear panel for over 5 seconds to reset all settings to default values Q3 What is the default IP address of the router A The default WAN IP address is 172 16 0 1 with subnet mask 255 255 0 0 Q4 What is different set to default and Factory set to default A Factory set to default you must push RST button until 5 second and gateway will clear all your setting and let gateway Wan port become the factory default 172 16 0 1 When you use setting to default by Web or telnet it will clear all your setting but the wan port setting will be saved If you remote the gateway after set to default you can login gateway again No reset the gateway wan port again A VoIP product almo
46. the port 1 is incoming call the other one SIP call from internet will ring port 2 SIP Hunting Table cm M IET Port 1 Port 2 ENS o0 Port 1 Port Z Figure 35 SIP hunting table setting SIP Proxy Setting ENCTEN SIP Prozy Server l use Net2Phone Serv Net2 Phone service 20 Register Interval seconds n c SIP Authentication Authentication SIP Authentication 9 Enable Disable Outbound Prozy Server oom 00 00 Figure 36 SIP proxy setting SIP Proxy Server Setting Domain Realm Enter the SIP realm in this field This field sets how long an entry remains registered with the Register Interval SIP register server The register server can use a different seconds time period The gateway sends another registration request after half of this configured time period has expired The outbound proxy method is just very like the proxy server built in NAT pass through solution except that the packets need to pass through the outbound proxy server Table 22 SIP proxy descriptions 34 NAT Pass Setti ng NAT Pass Method STUN Symmetric ETF STUN Server IP Address el 559 75 21 EB I Local Settin g Local SIP Pot Figure 37 NAT pass setting If your gateway under the NAT Firewall you should setting different NAT Pass function if you setting STUN Outbound Proxy you should have a STUN Outbound proxy server If th
47. ting the file transfer HHHHHHHHRHHHHHHHREHAHHHHHHHHHHEEE RHR RRA HEH 1311648 bytes received in 2304 ms 569 29Kbytes sec transfer succeeded 5 Socket closed 226 File sent ok 3 Socket closed Upgrade y n y Writing Image size 1311648 Written size 1311648 Write successfully Don t forget to restart the system Figure 68 Upgrade firmware procedures 55 Appendix D VIP 281GS Specifications Product H 323 SIP VoIP GSM Gateway VIP 281GS Hardware WAN 1 x 10 100Mbps RJ 45 port FXS 1 x RJ 11 connection PSTN 1 x RJ 11 connection GSM 1 x SIM connection Protocols and Standard H 323 v2 v3 v4 and SIP RFC 3261 SDP RFC 2327 symmetric RTP Standard STUN RFC3489 ENUM RFC 2916 RTP Payload for DTMF Digits RFC2833 Outbound Proxy Support G 711 A law u law G 729 AB G 723 6 3 Kbps 5 3Kbps Voice activity detection VAD Comfort noise generation CNG G 165 G 168 Echo cancellation Dynamic Jitter Buffer SIP 2 0 RFC 3261 H 323 TCP IP UDP RTP RTCP HTTP ICMP ARP PPPoE DNS Smart QoS IP TOS IP Precedence DiffServ System 1 PWR WAN 1 LNK ACT Line 1 In Use Ringing Phone 1 In Use Ringing GSM 1 In Use Standby SMS 1 Transmission Dimension WxDxH 180x110x25mm Operating Environment 0 40 degree C 0 90 humidity Power Requirement 12V DC EMC EMI CE FCC Class B Voice Standard Protocols LED Indications
48. tting Three methods are available for Internet Access Enter User Name provided by your ISP Password Enter Password provided by your ISP Confirm Password Enter Password to confirm again Table 5 PPPoE enable descriptions DHCP Client Dynamic IP Get WAN IP Address automatically IP Address If you are connected to the Internet through a Cable modem line then a dynamic IP address will be assigned 14 WAM Port Type Configuration um WAN Type Setting DHCP jw Select 21504 Subnet Mask 255 255 0 0 172160254 Apply Figure 9 DHCP setting 2 VoIP Basic Setup STEP1 Configure VoIP Call Signal Protocols User could select H 323 or SIP Protocol and click select VoIP Basic Configuration VoIP Protocol Setting SP vi Port Number Password P AX 20 digit a nn once a Figure 10 FXS GSM number setting STEP2 Configure the numbering with Phone FXS GSM ports The representation number is the phone number of the telephone FXS Number that is connected to Phone port GSM Number The representation number is the phone number of SIM CARD Table 6 FXS GSM number descriptions STEP3 Let GW Register to Gatekeeper SIP Proxy Server If user does not have Gatekeeper SIP Proxy Server Please go to STEP 4 Outgoing Dialing Plan There is a gatekeeper address fields If this gateway does Gatekeeper IP address
49. und dial plan Set advance parameters such as codec voice volume Auto Dial Setting Set auto dial number Hot Line Setting Port Status Port Status Display current telephone port status Figure 22 VoIP setup setting VoIP Basic Configuration to H 323 protocol Gateway H 323 protocol support H 323 v2 v3 v4 H 225 Q 931 H 245 and RTP RTCP Don t support H 235 security can t use H 235 security Authentication Username Password H 323 protocol is not good at pass NAT Firewall the best way is installed gateway on Public IP Address when it uses H 323 Configure the numbering with FXS GSM ports VoIP Basic Configuration VoIP Protocol Setting 1 223 w Select E 164 Number Setting MAX 20 digit Port LFXS E 164 Number none Port 2 GSM E 164 Number none i Figure 23 E 164 number setting E 164 number setting The representation number is the phone number of the TOSQHINDIIIDST telephone that is connected to FXS port The representation number is the phone number of SIM CARD Table 13 E 164 number descriptions 24 Configure the ANI Answer Number Indication Caller ID of the FXS GSM ports ITSP needs ANI for authorization when gateway calls Off Net call to PSTN number or mobile phone number Caller ID ANI Setting for Off Met Call Setting MAX 20 digit cos Port GSM Caller ID ANI Ng none Figure 24 Caller ID setting Register to H 323 Gatekeeper Note
50. x digits i is 16 digit maximun length of number is 20 digit Lae Length of Delete Reiste NUM Tren n Operatic Be Incoming na Muinbes Length Prefix na Destination Destination to CK peration SS n oa From To Figure 39 Incoming dial plan setting In the Incoming Dial Plan Configurations settings Maximum Entries 50 Inbound number The leading digits of the dialing number Emm The number of digits that will be stripped from beginning of the Delete Length dialed number 36 Table 26 Incoming dial plan descriptions Advance Setting to SIP protocol In Advanced Setting VIP 281GS provides user three major parts function to configure One is VoIP Advance the other one is Telephone Advance Network Advance and Tone Table Setting VoIP Advance Advance Setting Advance Setting Select VoIP Advance in DTMF Relay for SIP Inband 9 RFC2833 SIP Info EPC2Z833 Payload 101 from 96 to 127 Watchdog Disable 9 Enable VoIP Encryption e Disable C Enable VoIP Encryption Port 8888 MWI 9 Disable Enable Figure 40 VoIP Advance setting SIP VoIP Advance Configurtion After the VoIP call is connected when you dial a digit this digit is sent to the other side by DTMF tone There are three methods of sending the DTMF tone The first one is in band that is sending the DTMF tone in the voice packet The second one is DTMF Relay for SIP RFC2833 that is se
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