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GoIP User Manual - VoIP

1. 3 3 2 Network Proper network environment is the key to insure the voice call performance of the device In general Intranet offers a more stable network environment than Internet and it is the preferred network to be used If Internet is going to be used please make sure that the network can offer low packet loss small packet jitter and low packet delay Each voice channel requires less than 90 kbps when A law or u law voice codec is used GolP 8 will require 8 times this bandwidth Therefore it is very important to make that both upsteam and downsteam have enough bandwidth 30 headroom in order to accommodate the data traffics for the device installed when all lines are used simultaneously In order to get external network access the LAN port must be configured according to the network environment to be connected LAN Port 15 There are 3 access methods available to configure the LAN port 1 Static IP This mode applies to both public and private IP network environment In the LAN port configuration shown on the left select Static IP and then fill in the parameters as provided by your network administrator 2 DHCP default setting When the device is installed behind NAT and a DHCP host is available select DHCP to enable LAN Port IP Address Subnet Mask optional Default Route Primary DNS Secondary DN S optional LAN Port GolIP User Manual StaticiP Le lh DHCP the device to obtain
2. GolIP User Manual This section describes how to define custom network tones The Custom selection allows the following tones to be defined as shown on the right 1 Dial Tone When an incoming call is answered this tone is generated to indicate to the caller to dial a number 2 Ring Back Tone When a call is dialed from the device to VoIP and the SIP 183 is not enabled this tone is generated to indicate that the calling is in progress Network Tones Dial Tone Ring Back Tone Busy Tone Indication Tone Customized e fF 3 Busy Tone When a call dialed from the device to VoIP is busy this tone is generated 4 Indication Tone When a call waiting call is presence this tone is generated The syntax for a network tone script is defined as lt nf rpt plon ploff p2on p2off p3on p3off f1 f2 f3 f4 LL 12 13 l4 gt where nf is the number of single frequency tone 1 4 to be generated rpt is the number of times for the tone to be repeated based on the on off pattern defined 0 means infinite plon is the tone on duration for the first frequency tone ms ploff is the tone off duration for the first frequency tone ms p2on is the tone on duration for the second frequency tone ms p2off is the tone off duration for the second frequency tone ms p3on is the tone on duration for the third frequency tone ms p3off is the tone off duration for the third frequency ton
3. a Internet ZINN 2 os a Server Mode GolP Client Mode Goip Client Mode GolP Client Mode GolP Incoming call forward to Client GolP 48 Appendix E CID Call Forward GolIP User Manual For incoming GSM calls the phone number of the caller can be displayed at the called party SIP terminal The device supports the following two methods Unfortunately not all SIP servers support one or both methods Please check with the vendor of the SIP server for more information 1 Remote Party ID This is a parameter in a SIP INVITE message Choose this if both SIP Server and SIP terminal support this parameter Example Caller ID number 13800000000 The Remote Party ID parameter is included in the SIP INVITE Message below Sending Message to 192 168 2 1 5060 INVITE sip 5000 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 180 5060 branch z9hG4bK1645487913 From lt sip 20001 192 168 2 1 5060 gt user phone tag 406202416 To lt sip 5000 192 168 2 1 gt Call ID 847230278 192 168 2 180 CSeq 2 INVITE Contact lt sip 2000 192 168 2 180 5060 gt Max orwards 30 User Agent HBT Remote Party ID 13800000000 lt Sip 13800000000 192 168 2 1 gt party calling screen no privacy off Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 2 USD CID as SIP Caller number This parameter specifies the
4. RFC 2833 This specifies the QoS method used for SIP signaling Both IP TOS and DiffServe format are supported Select the proper setting that is compatible with your network environment Signaling encryption is employed to offer a more secure environment for SIP communications The following encryption methods are supported Consult your network VoIP administrator for more the proper selection if required RC4 Fast VOS AVS N2C ECM ET263 XOR OOS te Oe EY ae E EES This setting is not required if the target SIP server PBX supports NAT traversal However if your ISP blocks VoIP traffics you could try to use Relay Proxy setting Medial Settings 12 RTP Port Range 13 Packet Length ms 14 Jitter Buffer gt Delay gt Min Delay ms gt Max Delay ms 15 Media QoS 16 Media Encryption GolIP User Manual Depending on how your ISP blocks VoIP traffics the Relay Server method may or may not work in your network environment Two NAT Traversal methods are supported 1 Stun Server An external Stun Server is required This allows the device to obtain the public IP of the network used 2 Relay Proxy This is a proprietary method developed by HYBERTONE Technology HYBERTONE s Relay Proxy server must be used A free copy of the Relay Proxy can be download from HYBERTONE s website www hybertone com Please contact support hybertone com for further assistance if needed This specifi
5. Remark Adding a digit in front of an Whitelist entry enables a special call back function When the caller ID of an incoming call is matched the device first drops the call and then call back the caller automatically to allow the caller to dial a phone number Blacklist This list contains a list of caller numbers that are rejected by the device to make outgoing GSM calls when Call OUT Authentication is set to Blacklist 2 Parameters for Call IN via GSM This section defines how incoming GSM calls are handled on a per Can als Geng Enable Disable channel basis Calls can either be answered by the device or forwarded to VoIP C to the VoIP network connected Dial Plan Ir Hunt Group Mode Disable Please note that once the GSM Group mode of a channel is enabled the 27 Auth WhitelstBlacklist gt SIM Card Parameters gt gt However GSM forwarding must be disabled before a channel is SMS Settings GSM forwarding in the SIM Card settings is disabled automatically configured as Client in GSM Group mode 29 The parameters related to routing GSM incoming calls to VoIP are summarized in the table below GolIP User Manual Default Value 1 3 Call IN via GSM Forwarding to VoIP Number Dial Plan This setting controls the device to accept select Enabled or reject select Disabled incoming calls via the selected GSM channel This parameter defines if an incoming call
6. GSM Channel Shut Down to access the webpage below to shut each GSM module individually Place a check mark M to select the desired channel and then click Save to activate the shut down Remove the check mark and then click Save to turn on the channel again 39 Ss GolIP User Manual GSM Channel Shut Down E Shut Down Channeli E Shut Down Channel E Shut Down Channel3 E Shut Down Channel4 E Shut Down Channels E Shut Down Channel E Shut Down Channel E Shut Down Channels save Backup Restore Configuration The device configuration can be backup or restore via this page Click Backup Resotre Confguration to access the page shown below Password optional Backup Configurations Restore Configurations Load from file Choose File No file chosen passwordtoptiona C To backup the device configuration just click Save in the Backup Configuration section If a password is required when restoring a saved configuration enter a password before the backup To restore a saved configuration choose the configuration file in the Restore Configuration section and then click Restore Enter the password if required Reset Click Reset to reset the device configuration back to the factory default Click OK in the pop up window shown below to confirm this action The page at ww dbltek comials23 says Are you sure to reset to factory default Click OK to reset the device conf
7. GSM module 850M 900M 1800M and 1900M v Internal Antenna optional 1 4 Software Features v LINUX OS v Built in Web Server for device configuration v Built in SIP Proxy Simplified v PPPoE Dial Up v Router function Y DHCP client amp Server v QoS VLAN v VPN PPTP v Online firmware upgrade v Remote Control Mechanism for remote technical support Y Proprietary Auto Provisioning Mechanism v Remote SIM function v Short Messages SMS support standalone and server based v Call Management and Routing GolIP User Manual GolIP User Manual 1 5 Package Content Use care when unpacking the device package in order to avoid damage to the main unit and the packing materials Retain the packing materials in case the unit is to be transported in the future Please inspect the shipping container and the contents for any damages If visible damages are present please contact your vendor Keep the shipping materials for the carrier inspection The package should contain the items listed in the table below GolP 1 Channel 1 x Main Unit Uff GoIP 4 4 Channel dhl GolP 8 8 Channel pe GoIP 16 16 Channel AC DC Power Adapter GoIP1 12V 500mA GoIP4 12V 2A GolP8 12V A Adapter external power cord GolIP User Manual 1 x Ethernet CAT5 Cable 2M 1 6 LED Indicators Channel 4 Channel 2 mn Channel 3 Channel 4 w m a RUN LAN PM co ER ER C E o om E O The
8. GSM number to receive a SMS Alert on the Total Takl Time If this parameter is blank no SMS Alert will be sent SMS Alert Schedule The Remain Time is the Total Talk Time Limit minus the total talk time Remain Talk Time m used When this Remain Time reaches the value set in the parameter a SMS Alert is sent to the SMS Alert Number automatically Hide My GSM Number This parameter determines if the caller party can receive the phone Disable number of the caller or not Enabling this parameter hides My GSM number from the called party This specified if the phone number of the caller is shown at the called party or not GSM Call Forward Mode This section defines the Call Forward conditions for the selected GSM channel This section is not applicable when the GSM Group mode is enabled Please note that the GSM Call Forward defined here is equivalent to the same operation in a cellphone Unconditional Call Forward all incoming calls unconditionally to the number specified Forward Call Forward Busy Forward calls when the GSM Channel is in use Call Forward No Forward the call when an incoming call is not answered Answer Call Forward Forward calls when the GSM channel cannot register to the carrier Unreachable SMS Settings 33 GolIP User Manual 15 SMS Sender This parameter enables the selected channel to register to a SMS server The channel then becomes a SMS gateway for sending and receiving SMS g
9. In general the line status can be classified as follows gt Calls from SIP to GSM or Calls from GSM to SIP i IDLE The SIP line is not engaged in any call activities ii IN USE This status is active when a call connection between SIP and GSM is established gt Calls from GSM to SIP iii READY This status only applies to 2nd dial operation Itis active when a SIP call is answered and a second dial tone is generated to prompt the SIP caller to dial a number out iv DIALING This status is active when the GSM channel is dialing a number out JI NUMBER J0 dialing 10086 v RINGING This status is active when ringback tone occurs and the call is not yet answered vi CONNECTED 7 24 2 The middle column shows information on the hardware and network Hardware a S N This field shows the serial number of the device b Firmware This field shows the current firmware version c Model This field shows the model number of the device Network LOCAL TIME 10 Detail lt lt GolIP User Manual a LAN Port This field shows the IP address of the LAN Port b LAN MAC This field shows the physical hardware address MAC of the LAN port c PC Port This field shows the IP address of the PC Port d PPPoE This field shows the PPPoE dial up status Itis only meaningful when PPPoE is enabled el Gateway This field shows the default gateway IP for data traffics routing f DNS Server This fie
10. Mode for models GoIP GoIP 4 and GoIP 8 In this mode only one SIP registration is used for single or multiple line operation Please make sure that your SIP server supports multiple line operation and the SIP account is configured for the number of lines that matches the number of lines available in the device When receiving a call from the SIP server outgoing call it picks the idle GSM channel that is used the least in terms the number of calls dialed to dial out the call When a GSM channel receives an incoming call the call can either be answered by the device or forwarded to a SIP extension or IVR Please see more details in the Section x x for Call Management Config Mode single Server Mode e Advanced Signaling Settings Phone Number fo Media Settings gt Display Name ee AuthenticationID sd SIP Registrar Server fo Re register Period s Outbound Proxy fo Home Domain ee Backup Server Enable amp Disable Phone Number This is the SIP number used by the device Authentication ID The name of the device used in Caller Identification is defined here Display Name The Authentication ID used for SIP registration is specified here Password The password used for SIP registration is specified here SIP Proxy The address of the SIP Proxy or Server is specified here SIP Registrar Server The address of the SIP Registrar Server is specified here Re register Period Register to the SIP Server at an interval s
11. Server Enable amp Disable DO remove the Gateway Prefix P in the Dial Plan in the Call Divert section for VoIP to PSTN The syntax is Outbound Proxy P P Home Domain Backup Server When a GSM channel receives an incoming call the call can either be answered by the device or forwarded to a SIP extension or IVR Please see More details in the Section x x for Call Management Please note that the parameters defined in this mode are for each line individually Their definitions are similar to those defined in the parameters table for Single Server mode SIP Settings Config Mode 3 Config By Group for models GolP 4 and GolP 8 Config by Group This mode is basically a combination of Single Server mode and Config By Line mode It allows linestobe Group1 Group2 Group3 Group 4 Phone Number split up into groups Each group only uses one SIP registration for all the lines assigned to the group Display Name Authentication ID Each line can be assigned to only one group in the tg Password Doo Grouping section as shown in the right figure Please note the parameters listed in this figure are the SIP Proxy fe same as the parameters defined in the Config By Line mode except that there is no Backup Server option However these parameters are not group properties rather than line properties SIP Registrar Server Po Outbound Proxy Home Domain Grouping
12. below 15 is poor For poor signal reception it is recommended that the GoIP should be re located or an antenna with external cable should be used for better signal reception Signal strength of 99 means that the GSM channel is logout not registered to any Carrier e Login This fields shows whether the GSM channel x is registered to a Carrier or not OK means the GSM channel is ready for making and receiving calls If Fail is displayed please check if corresponding SIM and Signal are showing the correct status Please insert the SIM card to a cell phone to test if it is working properly Contact technical support if the SIM card is valid but registration to the carrier fails f Details Click on this expands the window to show additional channel information i Module This field shows the model number of the GSM module used ii Number This displays the phone number of the SIM card User needs to entered this manually iii Operator This field shows the name of the GSM Service provider registered iv Base Station This field shows the reference number of the base station connected v Remain Time When the Total Talk Time limit of the channel is set this field displays the remaining talk time When it is zero the channel is disabled for all outgoing calls Click on the Reset button reset the Remain Time to the Total Talk Time limit To enable the channel Each channel can be activated to operate for a certain p
13. not equal to the SIP Number defined for this line GoIP dials out the Caller Number according to the Dial Plan defined 3 Dial Plan The Dial Plan specifies rules to modify the Callee Number before dialing it out Each rule is terminated with the delimiter The rule matching begins from the left to the right Once a match is found rule matching terminates and the actions specified in the rule are executed Syntax a b c The portion on the left side a of specifies the prefix for matching starting from the beginning of the Callee Number The right portion b c is the action to be taken Both b and c are optional b means that b is removed from the beginning of the d A Callee Number If b is not found starting from the beginning of the Callee Number c means that c is added to the beginning of the Number generated from the last action In order for this rule to be meaningful b must be the same a or the beginning portion of iw a Example Callee Number 9262124567 Dial Plan 9 9 852 Actual Number dialed 85226124567 28 GolIP User Manual Syntax a b c d e b c specifies the range of a single digit Together with a they form a prefix for rule matching a can be a single or multiple digits d e are the actions to be taken as described in the previous syntax Example 913 5 9 9 86 In this rule Cal
14. one in the LAN port To Intranet Internet Power network 4 The DC port is for power connection Please only use the AC DC adapter provided Adapter with different rating or vendor may damage the device or affect its performance GolIP User Manual 5 The Reset button is recessed inside the GoIP cabinet You need to use a sharp pointer to access the reset button Press it momentarily to reboot the device Press it for 15 seconds or more to reset the device settings including login password to its factory defaults GolIP User Manual 3 Configuration The device can be configured via its built in http web server or via an Auto Provision Server Auto Provision Server is a free utility supporting both Window and Linux OS This utility is developed by HYBERTONE Technology for the sole purpose of automating the configuration of our products It is available in our website for free download This user manual only focuses on the device configuration via its built in http web server Please note that only window based Web browsers such as IE Chrome Firefox Mozilla are supported Do not use Linux based Web browsers If you are having problems in configuring your device with your existing Web browser please try one with lower version or a different Web browser and report the problem to us 3 1 HTTP WEB Server Login There are two methods to access the built in web server 1 Method 1 is to access the built in web server via the LAN port
15. to the selected GSM channel is forwarded immediately to the VoIP network or not If this parameter is blank the device answers an incoming GSM call If IVR in the Preference section is enabled the device generates a voice prompt to ask the caller to dial an extension number otherwise it generates a second dial tone If a phone number is assigned to this parameter a SIP INVITE to the Forward Number is sent to the SIP Server or the SIP trunk address This number must be a number that can be recognized and accepted by the VoIP network registered This means that it could be an extension number in the VoIP network or an E 164 number For 164 number the VoIP network must be setup properly for dialing out via another trunking service Since the device can be used for trunking it is possible to set it up to route an incoming GSM call from one channel and dial out to another party with an E 164 number via another GSM channel Special Feature Conditional forwarding is implemented to forward an incoming GSM call based on its caller ID Syntax a gt b iw a is a complete or portion of a number for matching with the incoming caller ID b is the number to be dialed via the VoIP network It could be an extension number or an E 164 number Example 98765432 gt 108 gt 101 In this example GolP first try to match the incoming GSM caller ID with the number 98765432 If it is a match GolP dial the number 108 If the
16. use of GSM Caller ID instead of its SIP number in the INVITE message when making a call Please make sure that the SIP server supports this type of INVITE message since the call now is not originated from a valid SIP number defined in the server Please note that the Remote Party ID is also included in the INVITE message Sending Message to 192 168 2 1 5060 INVITE sip 5000 192 168 2 1 5060 transportudp SIP 2 0 Via SIP 2 0 UDP 192 168 2 180 5060 branch z9hG4bK 1450498491 From 13800000000 lt sip 13800000000 192 168 2 1 5060 gt tag 232569343 To lt sip 50000 192 168 2 1 gt lt Call ID 18530689860 192 163 2 180 CSeq 2 INVITE Contact lt sip 13800000000 192 168 2 180 5060 gt Max Forwards 30 User Agent HBT Remote Party ID 13800000000 lt sip 13800000000 192 168 2 1 gt party calling screen no privacy offe Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type applications dp Content Length 226 49 GolIP User Manual Appendix F Volume Adjustment The volume adjustment of the device can be accessed via the URL below http lt device address gt en_US gaim html The lt device address gt is the IP address or domain name of the device The volume levels of the audio streams from VoIP to GSM and GSM to VoIP are controlled by the input gain and the output gain respectively An increase in the output gain means that the GSM PSTN party hears a hi
17. will only be used when the device answers an incoming call and the call is not forwarded to a SIP server automatically Predefined Network Tones are classified by country name If the country desired is not found in the list the Custom selection allows users to define the network tones individually Please refer to Appendix B for more information This is a proprietary DDNS service offered by HYBERTONE It allows HYBERTONE s products to identify each other via this DDNS service When this service is activated the domain name of the device is its lt serial number gt com This feature is useful to support peer to peer configuration The default DDNS Address is voipddns net which a free service offered by HYBERTONE Please contact your vendor if you want to install your own DDNS server The default communication port number is 39800 This specifies the interval between registrations to the DDNS This option allows the device to reboot itself at the time defined by Reboot Time This parameter specifies the time to reboot the device and t is defined in 24 Hour format HH MM The device is equipped with a simple voice prompt When this option is enabled and a call is answered the device plays a voice prompt instead of a dial tone to the caller 1920 14 GolIP User Manual Only the GolPs with the serial number xxxx support the Remote SIM feature Enabling this feature allows the SIM Cards to be installed in a
18. your SIM card to a GoIP You can then setup the GolP to forward all incoming calls to another GSM number in the world via a VoIP service provider The charge per call froma VoIP service provider is significantly lower than the roaming charge For office environment GolP offers a quick way to replace the traditional PSTN lines or T1 E1 lines to your IP PBX There is no initial installation reallocation charge and no need to wait for installation Depending on Our usage you can add or remove lines as per your requirement You can even configure the system so that everybody calls the same number regardless the number of lines available 1 2 Protocols Ze a Re ROR FR Se e AR ONS RR RNS e RN TCP IP V4 IP V6 automatic adaptive Dual VoIP protocols ITU T H 323 V4 IETF SIP V2 0 Multiple Codecs ITU T G 711 Alaw ULaw G 729A G 729AB G 723 1 and GSM H 2250 V4 H 245 V7 H 235 MD5 HMAC SHA1 RFC1889 real time digital transmission protocol NAT STUN Network Management Protocol NMP PPPoE Dial Up PPP Authentication Protocol PAP Internet Control Message Protocol ICMP TFTP Hypertext Transfer Protocol HTTP Dynamic Host Configuration Protocol DHCP Domain Name System DNS User Account Authentication via MD5 Proprietary Relay Protocol Avoiding VoIP Blockings 1 3 Hardware Features v ARMSE processor v DSP for voice signal processing v Two 10 100MB Ethernet ports IEEE 802 3 standard with status LEDs v Quadband
19. 8 2 29 IMEI 119706825464670 Ee Signal 99 Login FAIL LAN MAC 00 11 BE 08 7E E6 Login FAIL Status IDLE PC Port INBRIDGEMODE Petail gt gt Channel 3 Line 4 PPPoE DISABLED SIM NOT INSERTED Login FAIL bateway 192 168 2 3 IMEI 355073034572600 BE EEN Signal 99 Status IOLE DNS Server 203 246 252 2 Login FAIL Line 5 Detail gt Channel 4 Login FAIL Status IDLE Line 6 Login FAIL Status IDLE Line 7 SIM NOT INSERTED IMEI 3550 73034572527 signal 99 Login FAIL Detail gt Channel 5 SIM NOT INSERTED IMEI 001262197803351 GolIP User Manual 1 The left hand column shows the VoIP registration and call status by line a SIP Number This shows SIP number of the Line For Single Server mode all lines use the same SIP number For Config By Line mode each line should have its own SIP number For Config By Group mode lines belongs to the same group use the same SIP number For SIP Trunk Gateway mode this field is optional b Login This parameter shows the VoIP registration status The status OK means that the SIP registration for the line is successful FAIL means that the SIP registration fails and no calls can be routed to and from the GSM channels For SIP Trunk Gateway mode the Login status is always OK c Status This shows the operating status of the SIP line It is important to understand how call routing works between SIP and GSM in order to understand the meaning of the status
20. GolIP User Manual GoIP User Manual VoIP GSM Gateways Models GoIP GolP 4 GoIP 8 GoIP 16 HYBERTONE Revision 1 2 2012 6 22 http www hybertone com Sales marketing hybertone com Support support hybertone com GolIP User Manual Content TST EE 2 1 1 Miodu Te pat sce bares REESE BESES E E OO 2 1 2 POG E 3 1 3 FAG Ae PC AOU roe E E E EAE EA AEE ele 3 1 4 Sormare PEES a E T EEE E 3 1 5 Package Content EEN 4 1 6 BE EMC FORE EEN E E E E E EE E 5 SE E e EE 6 SE Seet 0 240 ee e EE 8 3 1 HTIP WEB ret LOS EE 8 a2 E 9 3 3 KEE E 1 0 87 EE 12 3 3 1 PPPS CCG a E E E 13 3 3 2 EE EE 15 3 3 3 TT 18 St AE EE 18 3 3 4 EH E ENEE I bua suhsianiawnsssnntetnatasieabiconaaleiawniatnndauadsseoshebaisuniwasieabasbidonsdubinotiesaiigianvnisasiadanantoubasusey 27 3 3 5 GSM EE EE ee 35 3 3 6 GSM Te Station E ninesini ete science E E a T a E a em dne ble 36 3 4 Reie EE 36 Appendix A Special SMS Commands EE 42 Appendix B SMS le 43 Appendix C Custom Nelwork Ee 47 Appendix D GNEO MOE E 48 Appendix E CID IR 49 Appendix F Ke ten Ee E 50 Appendix G Device WEE ET 51 1 GolIP User Manual General 1 1 Introduction GoIP is the abbreviated from GSM over IP It is a new type of VoIP gateway that allows call terminations from a VoIP network to a GSM network and vice versa Call connections between IP networks and GSM networks are now bridged seamlessly to extend the voice communication coverage significantly As the traditional PSTN lines a
21. IP Message format The To field in the SIP INVITE message contains the phone number of the called party The From field contains the SIP number of the line that is associated with the GSM channel Note The VoIP configuration of the device must be set to Config By Line mode Only the phone number of the called party is passed to the SIP server via a SIP INVITE message This mode is designed to use the GSM channel of the device to complete the Call Back function 43 GolIP User Manual Therefore the Call OUT via GSM parameter of the device must be enabled and the Forward Number associated with this parameter is set to the phone number of the SMS sender To achieve the Call Back function the SIP server calls the called party via its phone network and then calls the SIP number Since a call to this SIP number is set to forward to the phone number of the SMS sender both the called and calling parties can then be connected Example SMS content 8675588228822 Sender s number 861380000000 SIP Server IP 192 168 2 1 SIP Number 20001 The INVITE message sent to the SIP Server is INVITE sip 8675588228822 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 237 5060 branch z9hG4bK363969813 From lt sip 20001 192 168 2 1 5060 gt user phone tag 65248630 To lt sip 8675588228822 192 168 2 1 gt Call ID 117025903 192 168 2 237 CSeq 2 INVITE Contact lt sip 20001 192 168 2 237 5060 gt Max Forwar
22. LAN IP address and other network information automatically 3 PPPoE ADSL and Cable modems very often use PPPoE dial up to obtan network IPs If this is the case select PPPoE and then enter the information as provided by your ISP PC Port LAN Port User Name Password PPPoE H The PC Port allows other network devices to be attached to the device in order get network connection It offers both Router and Bridge modes to meet your requirements 1 Static IP default setting This mode enables the device PC Port to create another network segment and it then functions as a router gateway for this new network P Address segment Select Static IP for this new segment and Subnet Mask then enter the PC port IP address and Subnet Mask DHCP Server accordingly It also has a built in DHCP server to assign IPs to the Starting Address Ending Address static IP 192 168 8 1 255 255 255 0 Enable Disable 192 168 868 100 192 168 8 120 devices attached to this network segment Enable it Static DNS optional ts i zsCY and then enter the Starting Addrss Ending Address and Static DNS as required As a factory default the PC port is set to Static IP Router mode with IP Address set to 192 168 8 1 and Subnet Mask to 255 255 255 0 2 Bridge Mode Select this mode if your network topology pe por requires the network devices attached to the PC por
23. Line1 OLine OLine Line 4 Line5 Lines OLine7 Lines In Group Group Lei GolIP User Manual 4 Trunk Gateway Mode for models GoIP GoIP 4 and GolP 8 This mode offers a seamless interface with the SIP Trunk Cal Settings configuration in a SIP server In general SIP registration is Config Mode Trunk Gateway Mode SIP Trunk Gatewayt SIP Trunk Gateway2 O SIP Trunk Gateway3 sid not required in this mode this is an advantage for a SIP system that requires a license per registration If SIP server and the device are not in the same network segment it is recommended that both SIP server and the Phone Number RegisterExpinis o 0 co ol device are using public IPs for reliable operation Installing either one or both SIP server and the device Authentication ID behind NAT may or may not work properly depending on entication Password the SIP Server and the routers on each side as well In this case SIP server must support NAT Some routers may map an internal port to a different external port number VolP calls may fail to establish properly in this case The device accepts calls from up to 3 IP addresses SIP Trunk Gatewayl SIP Trunk Gateway2 SIP Trunk Gateway3 specified and then dial out the call via an Idle channel that is used the least in terms of the number of calls dialed The last part of the SIP Trunk Gateway IP addresses can be specified as X or x to represent that
24. SIM Bank rather than in the on board SIM slots GolP can either register to a SIM Bank or a SIM Server Please refer to the SIM Bank User Manual for more information gt Server This specifies the IP address of the SIM Bank or the SIM Server gt ID This specifies the name to be appeared in the SIM Bank or the SIM gt Password Server This specifies the login password to the SIM Bank or the SIM Server DTMF Tone This parameter specifies the maximum dropout time for a DTMF tone Dropout 200 When making a call from SIP to GSM ES Dropout 400 or from GSM to SIP by using the foot 0 second dial method the device needs to detect the dialing digits DTMF tone for one digit from the DTMF tones received via the voice data stream Depending on the network conditions short dropouts may occur due to packet jitter loss Therefore DTMF digit may be detected more than once if these dropouts are not taken into account Consequently the call is dialed to an incorrect number To avoid this problem a dropout window is used to avoid false detection when dropouts occur During this window the same DTMF digit is not recognized more than once The range of the dropout window is specified in terms of packet timestamp value The smaller the value is the smaller the dropout window is This increase the chance of detecting the same digit twice or more However if the value is set too large there is a possibility that the next digit is missed
25. SMS to a SIP terminal and a GSM number SMS Mode a IS eee HE Number ae KEE Number gt SMS Forward Number This parameter is valid in SMS Relay mode When an incoming SMS is received via a GSM channel it is forwarded automatically to the SMS Forward Number gt SMS Forward GSM The parameter is valid in SMS Relay mode When an incoming SMS is Number received it is forwarded automatically to the SMS Forward GSM Number via the same GSM channel 34 GolIP User Manual This specifies the method used to transmit an incoming GSM caller ID to a SIP 2 CID Forward Mode party CID Forward Mode Disable kd Disable Use Remote Party ID Use CID as SIP Caller ID Disabled Incoming GSM caller ID is not transmitted to SIP Use Remote Party ID This enables the Remote Party ID field is sent as part of the INVITE message The incoming GSM caller ID is specified as part of this field Use CID as SIP Caller ID This causes the CID information in an INVITE CID Forward Mode Use CID as SIP Calli message is open Cl lr the incoming GSM Caller ID gt CID Prefix This parameter specifies a prefix to be inserted in front of the incoming GSM caller ID received The intension for this is to modify the caller ID so that it can be dialed back directly by the called party Please refer to Appendix E for more information 3 3 5 GSM Service Provider This section sets the mode of the GSM service provider selection The fac
26. The LAN port is set to DHCP mode as a factory default When you connect it to a network with a DHCP host it will obtain an IP address from the DHCP host automatically Via the GolP s GSM channel s there are two ways to find out the IP address that is assigned to this port i Dial the SIM number of anyone of the GSM channels available Once the call is answered dial Q1 to hear a voice prompt reporting the LAN port IP address ii Send the HHHINFOHHH SMS command to one of the GSM channels available The GolP will then return back the LAN port IP address Please refer to Appendix A Special SMS Commands for more information Once the LAN IP address is known you are now ready to access its built in http web server by typing its IP address in the address field of a web browser 2 Method 2 is to access the built in we server via the PC port As a factory default the PC port IP is preset to 192 168 8 1 Connect a computer to the LAN port of the device and configure its IP to 192 168 8 x x 2 to 254 Type the IP address 192 168 8 1 in the address field of a web browser M Once the IP address is entered the login window lt gt shown on the right pops up Enter the user name Hy and password There are three level of access via i three different user names 1 Administrative Level This offers a full access right to all parameters available in the built in The server 192 168 3 11 at Please Login req
27. X This syntax monitors the length of the number dialed when performing second dial operation Each X represents a single digit If a prefix is known X s can be replaced by the prefix Example Dial Plan 13XXXXXXXXX This rule monitors the number dialed with the starting prefix of 13 and a length of 11 digits Once this condition is met the number will be dialed out immediately The maximum length for the Dial Plan definition is 140 ASCII characters There is no limit on the number of rules defined Each rule must be ended with the character The rule matching starts from the beginning and stops once a match is found 4 Hunt Group Mode Hunt Group operation is discuss in details in Appendix D Please note that Hunt Group Disable Mode is a property of each GSM channel and is required to be set individually Host This enables the channel selected to be the Host of the Hunt Group operation All clients registers and update the host on their channel status The host then maintains a list of clients status and selects an idle channel to receive the next incoming GSM call via GSM call forwarding Client This enables the selected channel to be a client in Hunt Group mode The Host Address field specifies the IP address of the device with the Host channel A client registers and updates its channel status to the Host Sharing of GSM channels between two Hunt Groups is supported When all channels are in use in one group Ca
28. able GSM Call Forward List lt GSM Call Forward Always Enable amp Disable Busy Enable Disable No Answer Enable Disable No Service Enable Disable SMS Settings SMS Sender Enable Disable SMS Server IP Doo O SMS Server Port SMS Client ID O SMS ACK Enable Disable 32 GolIP User Manual Parameter Description Default Value SIM Card 1 SIM Card Number This specifies the phone number of the SIM Card that is inserted to the channel line selected This specifies the International Mobile Equipment Identity number The Factory device comes with a factory default value Once changed this value default cannot be restored Unlock PIN1 SIM Card unlock PIN code 1 Unlock PIN2 SIM Card unlock PIN code 2 Total Talk Time Limit This sets the limit for the total talk time available for the SIM card When m this limit is reached no more outgoing calls can be made The SIM card must be reset via the status page or via a special SMS command see xx Talk Time Limit m call This sets the limit for the maximum talk time per call When this limit is reached the call is dropped automatically Billing Increment s This is a call duration measurement unit expressed in seconds Depending on your service provider some services are measured and billed in sixty second increments one minute or the billing increment may be in durations of six or even ten seconds SMS Alert Number This specifies the
29. acket timestamp for dropped or out of sequence packet problems The data packets are sorted based on the packet timestamp 3 Adaptive The adaptive mode optimizes the size of the jitter buffer delay and depth in response to network conditions in addition to the sequential mode functions This specifies the fixed jitter delay for both Fixed and Sequential Jitter Buffer mode This specifies the minimum jitter delay for Adaptive Jitter Buffer mode This specifies the maximum jitter delay for Adaptive Jitter Buffer mode Similar to Signaling Qos this parameter enables the QoS property for audio packets Both IP ToS Type of Service and DiffServe Differentiated Service method are supported Similar to Signaling Encryption item 6 in this table this parameter enables the encryption for audio packets Encryption methods supported are RC4 and ET263 GolIP User Manual 17 Symmetric RTP Network environment in some enterprise may require Symmetric RTP Please check with your network administrator for further support 18 Media NAT Traversal This setting is not required if the target SIP server PBX supports NAT traversal However if your ISP blocks VoIP traffics you could try to use Relay Proxy setting Depending on how your ISP blocks VoIP traffics the Relay Server method may or may not work in your network environment Two NAT Traversal methods are supported 3 Stun Server An external Stun Server is required This allows the dev
30. as shown below 38 GolIP User Manual Line Line SLine3 OLines4 Line5 OS Lines Line Lines Line 1 GSM Status LOGIN Line 165M Number 907 49138 SMS Content The procedures to send a SMS are a Select the Line GSM channel that you want to send a SMS The line status and the SIM GSM number are displayed b Enter the recipient s phone number GSM c Type the SMS message in the SMS Content box The maximum length of a message is 140 characters for 7 8 bit ASCII code and 70 characters for 16 bit Unicode d Click Send to send out the SMS SMS In Box Click SMS In Box to view the SMS messages received as shown below Select the desired line to view the latest 5 messages received for the corresponding GSM channel Line Line Line OLines4 OS LineS Linet Line OS Lines 06 01 16 55 13 Gs 8615817459136 but its taking time to save 06 01 16 54 43 s 48615817459136 sent message to you from my skype 06 01 16 54 33 s gt 48615817459136 see you on facebook 06 01 16 53 37 Gs 48615817459136 but i cannt logon facebook 06 01 16 53 17 beer Public furious over alleged rape of girls by official GSM Channel Shut Down This feature is implemented for two functions 1 Removing the power to a GSM channel before removing or inserting a SIM card This is a recommended procedure in order to prevent damages to the SIM card 2 Disabling a GSM channel temporary Click
31. d the call routing capabilities for both VoIP and GSM calls It is important to understand that GolP is designed to bridge calls between VoIP and GSM Therefore calls originated from a VoIP network is terminated to the GSM network via the GoIP and vice versa This means that a GoIP is handling both VoIP and GSM networks at the same time VoIP calls are mapped to the lines available according to the configuration mode The left hand column as shown below configures the parameters for each line which consists of both call routing and GSM SIM settings The VoIP line x is always mapped to the GSM channel x and this arrangement is referred as Line x in the Call Management page Call Management SIM Card Parameters GSM Number IMEI 353382282145791 F Unlock PIN2 fo Total Talk Time Limit m In rr Talk Time Limit mycal sd C Billing Increment s SMS AlertNumber TT SMS Alert Schedule Remain Talk Time m Hide My GSM Number Enable Disable GSM Call Forward List gt OMS Settings lt lt SMS Sender Enable Disable C SMS Server IP 192 168 2 1 a SMS Client ID SMS ACK Enable Disable 27 GolIP User Manual The parameters for each line are basically divided into 4 groups 1 Parameters for Call Out via GSM Call Management a Line Line Line Lined Lines Line6 Line Lines corresponding GSM channel When this call route is enabled the camourT via csm Enable Disable following para
32. ds 30 User Agent HYBERTONE Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 c Mode 3 SIP Message format The To field in the SIP INVITE message contains the phone numbers of both the called and calling parties These two numbers are concatenated by using the asterisk character with the number of the called party in the front The From field contains the SIP number of the line that is associated with the GSM channel Example SMS content 8675588228822 Sender s number 861380000000 SIP Server IP 192 168 2 1 SIP Number 20001 The INVITE message sent to the SIP Server is Sending Message to 192 168 2 1 5060 INVITE sip 8675588228822 8613800000000 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 237 5060 branch z9hG4bK363969813 From lt sip 20001 192 168 2 1 5060 gt user phone tag 65248630 To lt sip 8675588228822 8613902994477 192 168 2 1 gt Call ID 117025903 192 168 2 237 CSeq 2 INVITE Contact lt sip 20001 192 168 2 237 5060 gt Max Forwards 30 44 GolIP User Manual User Agent HYBERTONE Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 2 Relay Mode This mode supports SMS forwarding from GSM to SIP and from SIP to GSM a Received GSM SMS messages are forwarded to both SIP a
33. e ms f1 is the frequency of the first tone 300 to 3000Hz f2 is the frequency of the second tone 300 to 3000Hz f3 is the frequency of the third tone 300 to 3000Hz f4 is the frequency of the forth tone 300 to 3000Hz lI is the level for tone 1 range from 0 to 31 with 0 3dB 1dB for each increment 2 is the level for tone 2 range from 0 to 31 with 0 3dB 1dB for each increment 13 is the level for tone 3 range from 0 to 31 with 0 3dB 1dB for each increment l4 is the level for tone 4 range from 0 to 31 with 0 3dB 1dB for each increment Example 1 Dial tone definition 450Hz 20dB on continuously The dial tone script is 1 0 100 0 0 0 0 0 450 0 0 0 23 0 0 0 47 GolIP User Manual Appendix D GSM Group Mode The GSM Group mode is designed to simulate the function of one GSM number with multiple lines The idea is to form a GSM group with one number being the Server Only this GSM number is announced to the public Calls to this number are forwarded to other GSM numbers Clients in the group until all GSM channels are used up Effectively speaking if there are 40 GSM channels in a group a maximum of 40 concurrent calls can be achieved by just calling the GSM number of the Server channel The diagram below demonstrates this concept with only single channel GoIPs In fact GoIP with multiple channels can also be used Only one Server in a group and all the other channels must be set to Client individually
34. e GSM network This is done in order to avoid a long silent period before a ringback tone is returned from the GSM network SIP 183 The device sends back a SIP 183 Session In Progress message to the calling SIP device The calling SIP device then goes into early media mode to receive audio packets Since it may take a few to over 10 seconds for a ringback tone is returned from the GSM network the caller may hear a long silent period 3 No Answer Response This defines the response SIP message to be sent to the SIP Server Softswitch when a call dialed via the GSM channel cannot be connected successfully The device hangs up the call and then return the No Answer Response Depending on the SIP Server Softswitch settings the handling of the same No Answer Response may vary from server to server Please check out the server configuration against your application requirements in order to select the proper SIP message for No Answer Response Two SIP 486 choices are available 1 SIP 486 Busy Here 2 SIP 503 Service Unavailable This setting is intended to improve Answer Seizure Rate ASR if it is properly configured 4 CALL OUT PSTN Auth This setting defines how incoming VoIP calls are authenticated when the device is Mode configured for using SIP registration s This setting applies to Single Server Mode Config By Line and Config by Group modes This prevents unauthorized calls to be dialed out via the GSM Channel s The follo
35. eriod of time The channel is then enabled for outgoing calls Module G610 11 GolIP User Manual Service Provider CHINA MOBILE Number IMSI 460026009838591 ICCID 898600 Remain Time NO LIMIT Module Ver G610_V0C 58 0D_1T17 g Call Settings Click on this expands the window to show additional parameters i Call Waiting This shows the make of the GSM module used ii Call Forward This shows the name of the Carrier registered gt Always When it is On all incoming calls are forwarded unconditionally gt Busy When it is On incoming calls are forwarded when the channel is in use gt No Answer When it is On incoming calls are forwarded when they are not answered after x rings gt No Service When it is On incoming calls are forwarded when the channel fails to register to the carrier 3 3 Configuration Click Configuration on the left hand column to display the Configuration page and the following submenu 1 Preference 2 Network 3 VoIP 4 Call Management 5 GSM Service Provider 6 GSM Base Station Setting 12 GolIP User Manual 3 3 1 Preference Preference Langquage S English Network Tones China Time Zone GMT 8 DONS Enable Disable Time Server 192 168 5 1 DONS Address Auto provision Enable Disable DDNS Port 9800 ro la e CH CS cL cL wn OD Provision Server Doo Update Interval Provision Interval Aut
36. es the range of RTP port to be used for audio stream This specifies the length in time of each packet However the packet length is codec dependent as well The minimum packet length of a codec supersedes the valued specified here The table below summarizes the possible packet length for the codec 16384 32768 supported Codec Time Frame ms Time Packet G 711 a G 711 a law p law_ ois OOOO o 125 20 G 729 G 729A a 20 30 G 729AB G 723 1 30 60 A jitter buffer is designed to remove the effects of jitter from the decoded voice stream buffering each arriving packet for a short interval before playing it out This substitutes additional delay and packet loss discarded late packets for jitter If a jitter buffer is too small then an excessive number of packets may be discarded which can lead to call quality degradation If a jitter buffer is too large then the additional delay can lead to conversational difficulty A fixed jitter buffer maintains a constant size whereas an adaptive jitter buffer has the capability of adjusting its size dynamically in order to optimize the delay discard tradeoff Three modes of jitter buffer are supported 1 Fixed The fixed mode which is the default mode is a simple first in first out mode with a fixed jitter buffer delay 2 Sequential The sequential mode is also a fixed jitter buffer delay mode but in this mode the jitter buffer function looks at the p
37. first rule does not match it will continue to the second rule There is no matching number for the second rule It is then considered as a match and GolP dials the number 101 The maximum length for this parameter is 140 ASCII characters The number of rules can be adopted is limited by this length Each rule must end with the character When there is no match the incoming GSM call is handled as if the Forward Number is blank The Dial Plan specifies rules to process a number dialed after the call is answered This number is referred as a second dial number This enables a way to recognize second dial numbers that are in a known format and dial them out immediately Depending on the VoIP network connected a second number could be an extension or an E 164 number Syntax 1 a b c The portion on the left side a of specifies the prefix for matching The right portion b c is the action to be taken and they are optional When the beginning of the callee number matches a the first action b is to removed b from the Callee Number The second action c is to add c to the beginning of the number that is produced from the first action Please note that a b c could be a single digit or a Enabled 30 GolIP User Manual sequence of digits and they are independent Example 1 Dial Plan 9 9 852 Number received 9262124567 Actual Number dialed 85226124567 Syntax XXX XX
38. gher audio level An increase in the input gain means that the VoIP party hears a higher audio level Please note that changing these gain settings affects the DTMF tones in the corresponding path as well As a result DTMF tones for phone dialing may not be detected correctly Please change these settings with great care and make sure that DTMF detections are not affected Gain Settings Line 1 Line 1 Output Gain In sl Line 1 Input Gain 2 i Line 2 Line 2 Output Gain 0 ll Line 2 Input Gain 0 x Line 3 Line 3 Output Gain 0 sl Line 3 Input Gain 0 sl Line 4 Line 4 Output Gain 0 x Line 4 Input Gain U 50 GolIP User Manual Appendix G Device Characteristics REES mom aii w p o BLEER RELSE ER m ERE a GSM band Quad band 850M 900M 1800M 1900M Default ip eee in Power Max 5 W Max 12 W Max 20 W um POWER RUN LAN PC GSM one for each channel 0 10 Kg 0 45 Kg 1 2 Ke Without DC Adapter Operating 0 40 C a Operating 40 90 Non condensing Bed an EE C Tr 51
39. gistration which can be defined via teh following parameters This specifies the phone number for the SIP registration This specifies the authentication ID for the SIP registration This specifies the password for the SIP registration This specifies the period for sending a re registration request The section for Advanced Settings shown below is common for all Config Modes However not all parameters are applicable in all configuration modes In general factory defaults have been assigned to these parameters Users should only modify the parameters required SIP Listening Port Mode SIP INVITE Response No Answer Response Call OUT PSTN Auth Mode Bulit in SIP Proxy NAT Keep alive Regester Mode DTMF Signaling Signaling QoS Signaling Encryption Signaling NAT Traversal No Answer Expiry 32 130s NICT Expiry 2 180s ICT Expiry 5 360s Retransmit 11 200 2000ms Retransmit T2 2000 8000ms Advanced Signaling Settings Media Settings lt Random Le RTP Port Range s PacketLength ms SPa Te Jitter Buffer Fixed we Delay ms Media QoS Enable amp Disable Media Encryption Enable Disable El Symmetric RTP Mode 1 Media NAT Traversal None we Audio Codec Preference alaw ulaw None Haze Inband g729a g729ab g7231 Advanced Timing 180 200 2000 22 GolIP User Manual The table below summaries all the parameters defined in this section Parameter Descript
40. ice to obtain the public IP of the network used Relay Proxy This is a proprietary method developed by HYBERTONE Technology HYBERTONE s Relay Proxy server must be used A free copy of the Relay Proxy can be download from HYBERTONE s website www hybertone com Please contact techsupport hybertone com for further assistance if needed 19 Audio Codec Preference Six types of audio codec are supported and they are summarized in the table below Raw Data Ethernet 802 3 Bandwidth bps Data Bandwidth bps i ill 5 G 729AB 39K with Silence Compression and Voice Activity Detection VAD 6 G 723 1 5 3K 6 4K 26K 27K Note Time per packet 30ms is used for all bandwidth calculations For more calculations with other conditions please visit the VolP Bandwidth Calculator website http www bandcalc com Place a tick mark in the check box enable the corresponding codec The codecs are listed in a descending order of priority for codec selection This means that the top one in the table will have the highest priority to be selected when establishing a call To change the priority select the desired codec and the click on UP or DOWN button on the left Note The effective bandwidth for G 729AB is less since less data are transmitted when there is no voice activity 26 GolIP User Manual 3 3 4 Call Management The Call Management page defines all characteristics of each GSM channel an
41. iguration back to the factory default 40 J GolIP User Manual Reboot Click Reboot to restart the device Click OK in the pop up window shown below to confirm this action The reboot process will take couple of mins The page at www dbltek corm 50343 says Are you sure to reboot the device RT ne 41 GolIP User Manual Appendix A Special SMS Commands In order to manage the device special SMS commands can be sent to anyone of the GSM channel in order to read the LAN IP reset the device and reboot the device The table below summarizes the SMS command syntax lt and gt are not part of command text HHHIN FOHHH Sends an SMS response to the sender with the LAN port IP address HHHinfottitH RESET lt password gt Reset the device configuration back to the factory defaults and then reboot the device reset lt password gt lt password gt is the password for the administration level REBOOT lt password gt Reboot the device reboot lt password gt lt password gt is the password for the administration level 42 GolIP User Manual Appendix B SMS Modes The device receives SMS messages from both GSM and VoIP networks and they are handled according to the modes defined below 1 Dial Mode In this mode a received SMS is used to implement the Call Back function The concept is to establish a phone call between the called party and the calling party The phone n
42. ion Default Value Advanced Settings 1 SIP Listening Port Mode SIP Local port defines the network port number that the device listens for incoming SIP messages This port number is sent to the SIP Sever Proxy during SIP registration This setting defines if this port is pre assigned to a fixed number or a randomly generated port number 5060 to 6060 gt Port Number This specifies the port number when the SIP Local Port Mode is set to Fixed 2 SIP INVITE Response One of the key function of the device is to allow call terminations from VoIP to GSM In general a VoIP caller dials a PSTN or GSM number E 164 number and the SIP Server routes this call to the device by sending a SIP INVITE message This parameter specifies the response to the INVITE message The three possible responses are described in details below 1 SIP 200 OK This response inform the SIP server that the call is answered and the call duration timers starts immediately If billing applies the call is charged immediately even before the call is answered SIP 180 then 183 The device first sends back a SIP 180 Ringing to the calling SIP device to generate a local ring The caller hears a ringback tone immediately after the call is dialed When a ringback tone is received from the GSM network a SIP 183 Session In Progress message is sent to the calling party to start early media before the call is answered This allows to the caller to hear the ringback from th
43. ld shows the current DNS server for domain name interpretation It is possible that some domain names are blocked by local DNS servers Changing this to an overseas DNS server may solve the problem g VPN This field shows the VPN status If it is connected the local IP is also shown Not yet available The right hand column shows the parameters and settings for each GSM channel They are summarized as follows FH REMOTE SIM a Channel x This field shows the GSM module status of the channel x ON means that the power to the GSM module is shut down You can then process to check other channel information OFF means that the power to the GSM module is switched off You must first switch on the power to the GSM module via the web interface This setting is under the GSM Channel Shut Down which is located under the Tools menu b SIM This field shows INSERTED if the SIM card inserted in the SIM slot x is detected It displays NOT INSERTED if no SIM card is detected Please make sure that the SIM card is inserted properly as described in Section 2 It is also possible that SIM card is damaged or incompatible Please test the SIM card with a cell phone to confirm if the SIM card is working properly Please contact technical support for further assistance c IMEI This field shows the IMEI number in use d Signal This field shows the signal strength of Channel x Voice call quality with a signal strength above 15 is good and
44. lee Numbers starting with 9135 9136 9137 9138 9139 meet the prefix requirement The first action is to remove the first digit 9 from the number and then append 86 to the beginning of the number If the Callee Number is 913601234567 the actual number dialed is 8613601234567 4 Idle Interval In Between This setting defines an idle interval in between calls During this interval no outgoing calls Calls s are allowed to be made via the GSM channel 5 Call OUT Auth This parameter defines how incoming VoIP calls are authenticated before dialing them out via the GSM network Five options are available 1 None No authentication is required Calls are always dialed out via the GSM network Password The caller is prompted for entering the password before the call is dialed out Whitelist This is part of the Call Screen function Only the caller numbers listed on the Whitelist list are allowed to dial out via the device Password or Whitelist Either the password or the Whitelist authentication method will be used Biacklist The caller numbers listed on this list are rejected 6 Whitelist Blacklist Both Whitelist and Blacklist for call screening are supported Each list contains up to 15 Aarne Whitelist 1 Whitelist This list contains a list of caller numbers that are allowed to use the device to make outgoing GSM calls when Call OUT Authentication is set to Whitelist or Whitelist Password
45. live sends a NULL packet to the router regularly in order to keep the network ports used open This section consists of 5 basic timers in the SIP protocol Configure them carefully so that they are compatible with the SIP Server and your requirements This timer specifies the timeout for an unanswered call A SIP 408 Request Timeout command is sent to the SIP Server when this timer expires Note The default value is set the maximum value so that it will not interfere with the call unanswered timeout at SIP Proxy or PBX NICT Non Invite Client Transaction RFC 3261 Section 17 1 2 ICT Invite Client Transaction RFC 3261 Section 17 1 1 Round Trip Time RTT estimate RFC 3261 Section 17 1 1 This timer applies to the following timeout timer 1 INVITE request retransmission interval for UDP only 2 Non INVITE request retransmission interval UDP only 3 INVITE response retransmission interval The maximum retransmit interval for non INVITE requests and INVITE responses This setting specifies the DTMF dialing method 1 Inband DTMF tones are generated in the form of audio stream 2 Outband DTMF digits are sent in the form of digital commands RFC2833 SIP INFO DTMF tones are actually generated by the terminating party Outband This parameter is for outband DTMF dialing Select the proper format RFC2833 or SIP INFO as required by your SIP network This parameter specifies the payload type in RFC 2833 commands
46. ll Forward will be set to the Host channel of the other group When a free channel is available again Call Forward is then set to the free channel instead Please note that the call forward mode of the Host is always set to Unconditional Call Forward to the selected Idle Client gt Backup Host Address This parameter specifies the device IP that contains the Server channel in the other GSM Group The Server channel then updates the other Server channel with the GSM number of an idle channel Therefore GSM call forwarding can then be set to this number when all channels are in use 5 Call IN Authentication This parameter defines how incoming GSM calls are authenticated before routing calls to the VoIP network connected Five options are available 1 None No authentication is required all incoming GSM calls are routed to VoIP 2 Password The caller is prompted for entering the password before the call is routed or a second dial tone is generated 3 Whitelist Only calls with caller IDs that are listed on the Whitelist are accepted by the GSM channel selected Calls with GSM numbers not on the list are not answered at all 4 Whitelist Password Both Whitelist and password are used to authenticate incoming GSM calls 5 Blacklist Calls with caller IDs that are listed on the Blacklist are not answered by the channel selected 6 Whitelist Blacklist Call screen list can be set to Whitelist or Blacklist A maximum of 15 e
47. meters are displayed for user inputs chm Dial Pian Loo ol idle Interval in poe Calls s Call OUT Auth No Auth This section defines how a VolP incoming call is routed to the Whitelst Blacklist gt gt 1 Call OUT via GSM This setting defines if the device is allowed to make outgoing calls via the on board GSM Enabled channel s The typical application is to terminate VoIP calls via the GSM network This setting 2 Forwarding to GSM If this parameter is specified GoIP dials this phone number via the corresponding GSM Number channel whenever an incoming VoIP call is received for this line This is a fixed forwarding method and has the highest priority This means that the Dial Plan setting does not apply in this case Please note that how this line is selected depends on the Config Mode and the Callee Number received The Callee Number is defined as the phone number specified in the To field of an INVITE message Please refer to Section 3 3 3 1 for more information Calling the SIP number directly routes the call to the corresponding line immediately If this parameter is blank and the Callee Number equals to the SIP Number defined for this line a second dial tone is generated to wait the caller to diala phone number Please note that this could only happen in the Single Server mode Config by Line mode and Config by Group mode since SIP registration is required If this parameter is blank and the Callee Number does
48. nd GSM depending on the settings of the SMS Forward Number and SMS Forward GSM Number SMS Mode Relay dm lpg Number SMS Forward GSM IT E Number When an incoming GSM SMS is received it can be forwarded automatically to another GSM number as specified by the SMS Forward GSM Number The received SMS can also be forwarded automatically to a a SIP number or extension as specified by the SMS Forward Number If this number is not set this feature is disabled Forwarding GSM SMS to SIP is achieved via the SIP MESSAGE command An example of a SIP MESSAGE is shown below Please note that the number of the GSM SMS Sender is added as part of the message the last two lines in the SIP MESSAGE command Example SMS SIP Recipient 3999 SIP Proxy 192 168 2 1 GSM SMS Sender 861361234567 GSM SMS Content 075583185700 SIP MESSAGE Sent to SIP Server MESSAGE sip 3999 192 168 2 1 SIP 2 0 Via SIP 2 0 UDP 192 168 2 162 5060 branch z9hG4bK 1967685528 From lt sip 20001 192 168 2 1 gt tag 667435795 To lt sip 3999 192 168 2 1 gt Call ID 2094144847 1 92 168 2 162 CSeq 4 MESSAGE Contact lt sip 20001 192 168 2 162 5060 gt Max Forwards 30 User Agent HYBERTONE Content Type text plain Content Length 28 8613682626865 075583185700 Please note that the SIP Server side must be programmed to process this SIP MESSAGE according to the application needed It can forward the message to the SIP number with the Caller ID as the GSM SMS Se
49. nd the default password is 1234 Note Administration Level allows changing the passwords for all 3 levels User Level Administration Level SMS Level Send USSD Click Send USSD to access the webpage as shown below to send USSD commands 37 GolIP User Manual Line 1 Line Line Lined OLineS Line Line Lines Line 1 GSM Status LOGIN Line 1 GSM Number OU 421 20 ussp comman C The procedures to send an USSD command are a Select the Line GSM channel that you want to send an USSD command to the service provider The line status and the SIM GSM number are displayed b Enter the USSD command c Click Send Example For the service provider PCCW in Hong Kong the USSD command to check balance is 122 Enter 122 and the click Start The following screen is then displayed LineLine Send Command 1 22 Sending Please Wait A few seconds later the service provider sends back a USSD message response as shown below LineLine1 Send Command 22 Return Thank you for using our service Your current balance is 32 76 valid until 16 07 2012 ERR ae eae aaa EPP PEP PEP EP EPP PNP Hack Click Back to return to the Send USSD command page For certain service requests user responses are required Just following USSD message and then send back a response via SEND USSD command 4 Send SMS Click Send SMS to access the Send SMS webpage
50. nder If the message content is a phone number for a called party it is then possible to implement the Call Back function by using the content of the SIP message If the SMS GSM Recipient is set the received GSM SMS is forwarded to this recipient via the same GSM channel which receives the SMS 45 b GolIP User Manual A SMS can be sent to the device via its SIP number The content of the SMS must be in the preset format The first line must contain a valid GSM number and then the text message begins at the second line and must meet the restrictions imposed by a normal GSM SMS A sample of a SIP MESSAGE sent from SIP device is shown below Example A SIP SMS is sent from the SIP number 3999 to the SIP number 2001 used by the device and then the SMS is sent out to the phone number 1368266800 via the GSM channel associated with 2001 SIP SMS Sender 3999 SIP SMS Recipient 2001 SIP SMS Content 13682626800 Hello world SIP MESSAGE Sent from the SIP Server MESSAGE sip 20001 192 168 2 162 5060 SIP 2 0 From lt sip 3999 1 92 168 2 89 gt tag 5031 To lt sip 20001 192 168 2 1 gt Call ID 808807EB A8B3 DD11 BBA6 005056C00008 192 168 2 89 CSeq 3 MESSAGE Contact lt sip 3999 192 168 2 89 gt max forwards 16 date Tue 18 Nov 2008 06 36 37 GMT user agent SIPPER for 3CX Phone p hint usrloc applied Content Type text plain Content Length 26 13682626800 Hello world 46 Appendix C Custom Network Tones
51. ntries is allowed Disabled 1 Whitelist This list contains a list of incoming GSM caller numbers that are accepted GolIP User Manual answering calls from these numbers only by the device when Call IN Authentication is set to Whitelist or Whitelist Password Blacklist This list contains a list of incoming GSM caller numbers that are rejected not answering calls from these numbers by the device when Call Authentication is set to Blacklist 7 GSM Call Forward This setting specifies the Call Forward method when the GSM channel is configured as the server in HUNT Group mode 1 Unconditional Call Forward Incoming calls are always forwarded to an idle channel This leaves the Host channel for outgoing calls If all Client channels are in use Call Forward mode is disabled and the Host channel will then answer incoming calls 2 Call Forward Busy This means that the Host channel always answers an incoming call unless it is busy 3 SIM Card Parameters The third set of parameters are related to the SIM Card property Each SIM card has its own set of parameters and they need to be programmed individually if required SIM Card Parameters lt GSM Number ee IMEI 353382282145791 Unlock PIN fo Unlock PIN2 fo Total Talk Time Limit m 10 Talk Time Limit mycal Billing Increment s SMS AlertNumber sd SMS Alert Schedule Remain Talk Time m Hide My GSM Number Enable Dis
52. o Reboot Enable Disable Remote Controle Reboot Time GI Remote Control VR Enable Disable Remote Server Remote SIM Enable Disable Remote Server Port Server Remote Server ID D Remote Server Key Password DTMF Detect Min P L Gap 200 400 The preference page shown above consists of the following system level parameters and options as shown in the table below Description Language This sets the webpage and voice prompts language Currently only English and Simplified Chinese Mandarin for voice prompt are supported This specifies the offset of the local time zone with respect to GMT The syntax should be GMT x where x is the offset This specifies IP address or the domain name of a network time server for computer clock synchronization The default is pool ntp org Auto provision The auto provision is optional When this option is enabled the device downloads its configuration from the Auto Provision Server at start up or at the time interval specified by the Provision Interval The configuration file name is lt Serial Number gt cfg which is just a text file not encrypted If encrypted format is required please contact technical support for further assistance Please note that Auto Provision Server is a free utility supporting both Linux and Window environment Please visit our website or contact technical support for more information gt Provision The specifies the Provision Sever addre
53. pecified by this parameter s Outbound Proxy The address of the Outbound Proxy used for VoIP communication is specified here Home Domain Home Domain is used in SIP identification It should be specified as required 10 Backup Server Backup Server improve service reliability and is used only when the primary server fails gt SIP Proxy This specifies the backup SIP Server address gt SIP Registrar Server This specifies the backup SIP Registrar Server address gt Home Domain This specifies the backup Home Domain address 19 GolIP User Manual 2 Config By Line for models GoIP 4 and GolP 8 This mode is only applicable for multi line models SIP Settings Each line associated with a corresponding GSM Config Mode Line1 Line2 Line5 Line Phone Number Config by Line e Line3 Line 4 Line Lines channel registers to a SIP server separately and The Gateway Prefix parameter see the table below is operates as an independent phone line adopted for each line Itis used to specify which line GSM channel to be used for making an outgoing call Pisplay Name When the SIP server routes a call to the GolP for Authentication ID making an outgoing call the Gateway Prefix must be Password specified at the beginning of the phone number to be dialed outgoing call fails Gateway Prefix Otherwise the attempt of making an SIP Proxy In addition it is necessary to SIP Registrar
54. r in order to install SIM cards First slide the metal clip to the direction as indicated on the top of the clip Insert a SIM card to each slot carefully and then place the metal clip back in place For the models with the SIM card slots located at the back just insert a SIM card to each slot as shown in the drawing on the right Please make sure that the orientation of the SIM Card is correct before inserting the card i AN For GolP 1 channel the SIM card insertion orientation is shown in the figure on the right The metal contacts must face down and the cut corner is inserted first For GolP 4 and GolP 8 the SIM card insertion orientation is HD Ze B we D em D e D e D e M we He shown in the figure on the right The metal contacts must ag Hu Se H H HBH Ed Y Y VY DU DU VW VY GU face up and the cut corner is inserted first 2 The LAN port is intended for intranet or internet connection Depending on your network environment it can be connected various type of network equipment such as network router network switch Hub xDSL Cable modem etc 3 The PC port is intended for network sharing and it supports both bridge and router modes In Bridge mode the PC port is connected to the same network segment as the LAN port In Router mode the PC port is set to a different network segment In this case please make sure that the PC network segment IP 192 168 x is different from the
55. re starting to disappear in developed countries and are not going to be built extensively in under developed countries GSM phones are getting more and more popular all over the world with lower and lower service charges the emergence of GolP bridges the gap between the traditional telephone networks and VoIP networks as shown in the diagram below As a result local and worldwide voice communications are more convenience lower cost and broader coverage Internet Intranet 41444444 li UI LULU GolP4 M GolP8 VolP Service Provider Local R E m Telephone GSM Network Network World VolP N Telephone Phone N Network oN Gi Q Telephone Cellphone Telephone You can now make a call from anywhere in the world via a VolP network and then terminate the call via a GolP to the local telephone network PSTN On the other hand you can also make a call from the local telephone network to a GolP the GSM phone number and then dial another number via a VolP network to anywhere in the world In these two cases a VoIP Service provider is required for one side of the call termination For two fixed locations it is possible to setup GoIPs at both ends for call terminations without subscribing to a VoIP Service provider GoIP can also be used to achieve GSM roaming via VoIP The idea is to route all your incoming GSM calls to a GolP via call forward or simply insert
56. s mode fixes the base station BTS to be used 3 4 Tools Click Tools on the left hand menu to access the submenu as shown below Online Upgrade Status Last Upgrade Time 2012 05 04 16 10 20 Configurations Current Version G5 4 01 38 Tools Upgrade Site Online Upgrade Change Password send USSD Send SMS ahiS Box Modules Control BackupRestore Configurations Reset Contig Feboot 36 i levels GolIP User Manual Online Upgrade Click Online Upgrade to upgrade the device firmware The current version is displayed as well as the last upgrade time Contact us or local agent supplier for the latest firmware version Enter the firmware link URL and then click Start to begin firmware upgrade Once the firmware upgrade is completed the device will reboots itself automatically Please wait patiently Note It is important not to disconnect the power during a firmware upgrade since the internal Flash may be corrupted If this happens pleases contact technical support for assistance c Please reboot the device if an upgrade attempt fails before performing another upgrade Change Password Click Change Password to change the password with respect to the login level There are three login Administrative Level Login ID is admin and the default password is admin 2 User Level Login ID is user and the default password is 1234 3 SMS Level Login ID is sms a
57. ss IP or Domain name Server This specifies the interval in performing an auto provisioning event gt Provision Remote Control gt Remote Server gt Remote Server Port gt Remote Server ID gt Remote Server Password Network Tones gt DDNS Address gt DDNS Port gt Update Interval Auto Reboot gt Reboot Time GolIP User Manual LI onewa o o O This is a unique feature that allows remote access to the device s built in Web server even when it is installed behind NAT To achieve this function a Remote Control Server is required to be installed This server is a free Linux based utility and is available for download via our website Please contact technical support for further assistance if required Once installed please make sure that the Remote Server Port and Password are set properly This specifies the IP address or the domain of the Remote Control Server Check with your Remote Server administrator for the communication port This specifies the name to be appeared in the Remote Control Server It is used as a reference for the device This specifies the login password to the Remote Control Server This is not the password to login to the built in webpage Please ask your Remote Server Administrator if it is not available Network tones are the tones associated with the traditional PSTN telephone network such as dial tone ring back tone busy tone call waiting tones etc These tones
58. t SMS Server IP This specifies the domain name or ip address of the SMS server gt SMS Server Port This specifies the communication port that is used by the SMS server This must match the value set in the SMS server gt SMS Client ID This specifies the login ID for the channel gt Password This specifies the login password for the SMS Client ID 16 SMS ACK SMS ACK is the acknowledgement response to a SMS received If this parameter is disabled the SMS ACK is not sent and the carrier continues to send the same SMS periodically until the message is terminated 4 Miscellaneous Parameters The parameters listed in this section are device property instead of channel property as compared to the last 3 groups Default Value 1 SMS Mode This defines how the device handles SMS messages Disabled SMS Mode Disable EA Disable Dial This mode is used to support Call Back function via incoming GSM messages Appendix B describes the three different modes of operations in order to meet the different requirements from various SIP servers gt SMS Dial Please note that the SIP server registered must be configured for this operation gt SMS Dial Prefix This parameter specify which SMS Dial mode is used Please Appendix B for more information on the modes available The parameter is applicable for SMS Dial mode It allows a prefix to be added to the phone number of the called party Relay This mode forwards incoming GSM
59. t to be in the same network segment as the LAN port Advanced Features 802 14 VLAN 1 VLAN This is a type QoS service and is intended to give VLAN ID higher transmission priority to real time packets i ML AN QoS However your router switch and ISP network need to support this feature as well 2 PPTP VPN This option allows the device to create a VPN PPTP YPN tunnel with the designated VPN Server The VPN PPIP Server protocol supported is PPTP with no encryption or 40 bit encryption which is defined on the VPN server In PPTP Username PPTP Password Ethernet Mac Address IP Broadcast Address Bridge mode Static IP Enable Disable Enable Disable DO 00 0 ol Advanced Doo ol 16 GolIP User Manual general this option is used to avoid VoIP blockings 17 GolIP User Manual 3 3 3 VoIP This section defines all VoIP related settings The device supports both SIP and H 323 protocols For GoIP 1 channel and earlier version of GoIP 4 8 both protocols are embedded in a single firmware version User must select the desired protocol via the parameter End Point Type as shown below in the VoIP Settings section Endpoint Type SIP Phone e Config Mode H 323 Phone SIF Phone As more features are added SIP and H 323 VoIP protocols are supported in two different firmware versions GoIP 4 and GolP 8 are now shipped with the SIP protocol firmware as a factor
60. the whole segment IP addresses 0 255 Calls originated from the IP segement are accepted Example SIP Trunk Gateway2 123 124 125 x This example shows that Calls originated from 123 124 125 0 to 123 124 125 255 are accepted Please note that selection of GSM channels for outgoing calls is done by the device automatically Please use Config By Line or Config By Group mode if you would like to route a call to a specific GSM channel For received GSM calls they will be routed to SIP Trunk Gateway1 provided that an unique IP is used SIP Registration is only supported for SIP Trunk Gateway1 Just fill in the SIP parameters list to enable this operation The parameters available in this mode are listed in the table below Parameter Description Default Value Trunk Gateway mode SIP Trunk Gateway1 This specifies the first SIP Trunk Gateway address Use X or x for the last part of the address to specify the entire segment 0 255 SIP Trunk Gateway2 This specifies the second SIP Trunk Gateway address Use X or x for the last part of the address to specify the entire segment 0 255 SIP Trunk Gateway3 This specifies the third SIP Trunk Gateway address Use X or x for the last part of the address to specify the entire segment 0 255 21 SIP Registration to Trunk Gateway 1 Phone Number Authentication ID Password Re register Period GolIP User Manual Some Trunk gateway connection requires a SIP re
61. tory default setting is Auto and this enables auto GSM Service Provider selection to the SIM card default GSM Service Provider Settings Line 1 Line Linea Lined Lines Line6 Line7 Lines Service Provider Aug Je save Changes When a GolP is installed in a border location where GSM roaming could occur the Fixed mode could be used to manually select the preferred GSM service provider This could avoid expensive roaming charges Select Fixed mode and then press Save Changes to save the setting You need to refresh the browser after a few minutes to view a list of GSM service providers as shown below Enter the provider code 35 GolIP User Manual displayed in the Code entry and then press Save Changes GSM Service Provider Settings S Une OLine2 OLine3 OLined Line5 Line OLine Linea Service Provider Fred Je Code 46000 CHINA MOBILE 46000 Save Changes 3 3 6 GSM Base Station Settings This feature is currently in beta testing and it is intended for advanced users only Don t attempt to change the default settings if you do not have a good understanding on GSM network Please contact us for help if you have a specific requirement on GSM base station settings Three modes for base station selection are available 1 Auto This mode uses the default GSM base selection mechanism A Poll This mode limits the number and the list of base stations BTS to be used 3 Fixed Thi
62. uires a Username and password Warning This server is requesting that your username and webpage The user name and password for password be sent in an insecure manner basic authentication the administrative level are admin and without a secure connection admin respectively 2 User Level This level restricts user from accessing the Call Setting page User will not Password rr be able to change any VoIP related settings The user name and password for the user level are user and 1234 respectively 3 SMS Level This level only allows user to access Ss Cancel the Send SMS and SMS Box functions under the Tool menu The user name and password for User name admin ig Remember my password GolIP User Manual the SMS level is sms and 1234 3 2 Status This is the default page when you first login to the built in HTTP Web server It consists of three columns as shown in the diagram below All three models use the same status page design It is important to understand the information shown in this page in order to debug or report problems encountered VolP Hardware GSM Line 1 SIN GOIP2E9T111052134 Remote SIM DISABLE Channel 1 Login OK Firmware 55 4 01 44 1 SIM INSERTED Status IDLE Model GolPxs G610 IMEI 353382282141295 a BEEN signal 25 Line 2 Local Time 2072 12 03 13 49 32 Login OK Login FAIL Detall gt gt er Channel 2 status IDLE Network SIM NOT INSERTED Line 3 LAN Port 192 16
63. umber of the called party is specified in the GSM SMS message received The phone number of the calling party is the SMS sender s number The device then sends a SIP INVITE message containing these phone numbers to the SIP Server registered Three different SIP INVITE message formats are supported and are described below a Mode 1 SIP Message format The To field in the SIP INVITE message contains the phone number of the called party The From field contains the phone number of the calling party Once the SIP server receives these two numbers via a SIP INVITE message it then terminates the SIP call SIP INVITE and then call both parties via its own phone network The device may or may not take part in the actual call conversation Example SMS content 8675588228822 Sender s number 861380000000 SIP Server IP 192 168 2 1 SIP Number 20001 The INVITE message sent to the SIP Server is INVITE sip 8675588228822 192 168 2 1 5060 transport udp SIP 2 0 Via SIP 2 0 UDP 192 168 2 237 5060 branch z9hG4bK363969813 From lt sip 8613800000000 192 168 2 1 5060 gt user phone tag 65248630 To lt sip 8675588228822 192 168 2 1 gt Call ID 117025903 192 168 2 237 CSeq 2 INVITE Contact lt sip 8613800000000 192 168 2 237 5060 gt Max Forwards 30 User Agent HYBERTONE Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 b Mode 2 S
64. wing authentication methods are available None No authentication is used for calls received This could be a simple arrangement if calls are routed from a SIP Server in the same local network IP only calls received from the registered SIP Server s are accepted Password A SIP 401 message is sent to the SIP server for password authentication of the corresponding SIP account when a call is received IP and Password Both authentication methods are used 5 Bulit in SIP Proxy Note This setting has no effect when the device is set to Trunk Gateway mode A simple SIP Proxy is embedded in the device Choose Enable to activate this SIP 23 gt Password NAT Keep Alive Advanced Timings No Answer Expiry 32 180s NICT Expiry 2 180s ICT Expiry 5 360s Retransmit T1 200 2000ms Retransmit T2 20000 8000ms r DTMF Signaling gt Outband DTMF Type gt RTP Payload Type Signaling QoS 10 Signaling Encryption 11 Signaling NAT Traversal GolIP User Manual proxy to accept any SIP registrations with the correct password which is specified in the parameter Password There is no need to create a SIP account in this server Users will have to manage the SIP numbers used on their own This facilitates the setup of a simple SIP network for customers who do not have their own SIP servers This sets the password for SIP registration to the built in SIP server When enabled NAT Keep A
65. y are often used LED indicators shown above for GoIP 8 are used to show the current status of the device to determine if the GolP is working normally or not This LED is red and illuminates when power is connected LAN This LED is red and illuminates when the LAN port is connected and blinks when data transmission occurs Be This LED is red and illuminates when the PC port is connected and blinks when data transmission occurs This LED is green and blinks at a rate of every 100ms when VolP is not ready for making calls Fast Blink It blinks at a rate of every second when VoIP is ready for making calls Slow Blink Each GSM channel has its own status LED and its color is green It blinks at a rate of every 100ms Fast blink when the corresponding GSM channel is not yet registered to a GSM network Channel x It blinks at a rate of every second Slow blink when the corresponding GSM channel is ready for making or receiving calls registered to a GSM network It illuminates when GSM call activities occurs in use ringing GolIP User Manual 2 Installation The installation procedures for GoIP GoIP 4 and GoIP 8 are the same The only different is on the number SIM cards to be installed 1 SIM card slots are located either at the bottom for old hardware or at the back for new hardware of the main unit For the models with the SIM card slots located at the bottom you need to open the bottom SIM cove
66. y default If H 323 protocol is required the firmware of the device can be changed to the one that supports H 323 protocol Please visit our website for the latest firmware versions or contact your supplier for more information To find out the protocol installed in the device please check the firmware version as described below 1 SIP firmware version is in the format GS 4 01 xx 2 H 323 firmware version is in the format GH 4 01 xx Please contact your vendor or visit our website for the upgrade link to the latest version The upgrade method is discussed in Section x x You can always switch the firmware from SIP to H 323 or vice versa In general it is important to understand your VoIP application with the device before proceeding to device configuration If the device is going to work with a IP PBX please make sure that you know how to configure your IP PBX It is very important that you send us your application requirements in full details when seeking for technical support in configuring the device 3 3 3 1 SIP SIP settings are categorized into three groups of parameters in order to simplify the configuration Click on VoIP selection on the left column menu to access the SIP Settings page Depending on the models a total of four configuration modes are supported Contig Mode Trunk Gateway Mode e Single Server Mode Contig by Line Contig by Group Trunk Gateway Mode 18 GolIP User Manual 1 Single Server

GoIP User Manual - VoIP

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