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1. 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 CID number matches a the first action b is to removed b from the CID 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 sequence of digits and they are independent Example 1 CID Prefix 9 9 852 CID Number 9262124567 Actual CID Number forwarded 85226124567 This setting controls the device to accept select Enable or reject select Disable incoming calls via the selected GSM channel This parameter defines if an incoming call 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 E 164 number the VoIP network must be setup properly for diali
2. Line 3 Line 3 Output Gain 0 y Line 3 Input Gain 0 y Line 4 Line 4 Output Gain 0 y Line 4 Input Gain 0 y 61
3. GSM Carrier cH1 cH cH3 OcHa OCHS cH6 cH7 CH8 Service Provider When a GolP is installed at a location that is close to a country border it is possible that the default service provider is not selected based on the base station signal strength The GolP may then register to a GSM service provider that charges for expensive roaming fees In order to avoid this the Fixed mode should be selected in order to lock the channel to a preferred service provider When the Fixed mode is first selected you must press Save Changes to save the setting and then refresh the browser after a few minutes in order to view a list of GSM service providers Enter the provider code displayed in the Code entry and then press Save Changes Please view the screen captures below GSM Carrier OCHI CH OCH3 OcH4 OcH5 cHe cH CH8 Service Provider Fixed Code 45412 CMCC PEOPLES 45412 3 3 16 GSM Base Station This feature is currently in beta testing stage with certain customers 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 the GSM network Please contact us for help if you have a specific requirement on GSM base station settings GSM Base Station CH1 OcH OcH3 OcH4 cus cHe OcH7T CH8 Base Station Selection Available Cell IDs 45 GoIP User Manual Three modes for base station selection are available 1 2 Auto Thi
4. 43 Next the parameters defined below are channel dependent selected before making changes Parameter Description Default Value SIM Card 1 SMS Server gt SMS Server IP gt SMS Server Port gt SMS Client ID gt Password Validity Period Send SMSC Number SMS to Email gt Outgoing Email Server SMTP gt User Name gt Password gt Forwarding Email Address SMS Server is a Linux based utility which is used to manage GolPs registered to send and receive SMS messages In addition it also keeps an activity logs of each channel It is a free utility offered by the GolP s manufacturer and can be downloaded from the URL below http www hybertone com en download asp For more information please visit our website or contact support HyberTonetek com Once the SMS Server is installed and in operation The following parameters are require to enable the GolP to register to the SMS Server Please note that each channel must be programmed individually in order to register to the SMS Server This specifies the domain name or ip address of the SMS server This specifies the communication port that is used by the SMS server This must match the port value set in the SMS server This specifies the login ID for the channel selected SMS Client ID and password must first be created in the SMS Server This specifies the login password for the SMS Client ID This specifies the message expiration time
5. lt sip 13800000000 192 168 2 1 5060 gt tag 232569343 To lt sip 5000 0192 168 2 1 gt Call ID 18530689864 192 168 2 180 CSeq 2 INVITE Contact lt sIp 138000000000192 168 2 180 5060 gt Max Forwards 30 User Agent HBT Remote P arty 1D 13800000000 lt sip 13300000000 192 168 2 1 gt party calling screen no privacy 0ff Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type applications dp Content Length 226 60 GoIP 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 higher 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 Asa 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 fo y Line 1 Input Gain 2 v Line 2 Line 2 Output Gain 0 y Line 2 Input Gain 0 y
6. 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 sequence of digits and they are independent Example 1 Dial Plan 9 9 852 Number received 9262124567 Actual Number dialed 85226124567 Syntax XXX XXX 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 5 Hunt Group Mode Hunt Group operation is discussed in details in
7. MA This specifies the period for sending a re registration request ioe 3 3 4 Advanced VoIP The parameters in the Advance VoIP section are common for all configuration modes In general these parameters are preconfigured with factory defaults Users should only modify the parameters required 24 GoIP User Manual Advance SIP SIP Listening Port Mode Port SIP INVITE Response Call OUT PSTN Auth Mode Bulit in SIP Proxy 8 Enable Disable NAT Keep alive Enable Disable DIMF Signaling Signaling QoS Signaling Encryption None Signaling NAT Traversal None w No Answer Expiry 32 1905 180 NICT Expiry 2 1805 z ICT Expiry 5 3605 Retransmit T1 200 2000ms Retransmit T2 2000 8000ms 2000 GSM SIP Code Map gt gt The table below summaries all the parameters defined in this section Parameter Description Default Value Advanced VoIP 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 INVIITE Response One of the key function of the device is to allow call terminations from VoIP to GSM SIP 180 the
8. 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 Two formats are supported 1 HH MM When this is specified with a valid 24 hr time format 00 00 to 23 59 the goip is rebooted at this specified time Invalid time specified has no effect M This specifies the reboot duration in minutes The valid range for this is from O to X Changes saved are only effective after the device is rebooted 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 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 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 This specifies the IP address of the SIM Bank or the SIM Server This specifies the name to be appeared in the SIM Bank or the SIM Server This specifies the login password to the SIM Bank or the SIM Server Specify the network protocol UDP or TCP is used for Remote SIM communications Thi
9. Therefore 54 GoIP User Manual 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 c Mode 3 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 Forwards 30 User Agent HYBERTONE Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 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
10. lt limit gt lt limit gt is an integer Talk time allowed as per the period defined When the accumulative talk time since the period starts reaches the Talk Time Limit defined the SIM card is disabled for outgoing calls until the period expires If there is an overlap between two periods defined the first period takes priority The next period starts from the end of the first period If there is a gap between two periods the talk time for this gap is unlimited It is important to note that the Talk Time Limit is reset automatically to the one defined for the next period There is no need to reset a SIM via the webpage in this case Syntax lt period1 limit1 gt lt period2 limit2 gt lt Period gt is defined in 24 hr format with starting and ending time separated by a Example 1 00 00 24 00 This specifies a time period of one day starting from 00 00 till the end of the day Example 2 08 00 12 00 100 11 00 20 00 200 In this example the first period starts from 08 00 till 12 00 with a talk time limit of 100 minutes The second period starts from 12 00 till 20 00 with a talk time limit of 200 minutes From 20 00 till 08 00 the next day there is no talk time limit Note By default SIP Registration is disabled when a SIM card is disabled To change this default select the Enable option for SIP Registration when Talk Time Limit Expires in the SIM Page 8 Drop Call when the This parameter specifies if
11. Alert Trigger m 10 Hide My GSM Number Enable amp Disable Auto Config Other lines 39 GoIP User Manual The parameters for this section are described in details in the table below Parameter Description Default Value SIM Card 1 GPRS Registration This enables the GPRS registration mode for all GSM channels 2 SIP Registration when This parameter determines if SIP Registration is enabled or disabled when Disable Talk Time Limit Expires Talk Time Limit expires If it is enabled incoming calls are still enabled when Talk Time Limit expires The parameters below only apply to the channel selected 3 GSM Number This specifies the phone number of the SIM Card that is inserted to the channel selected This field MUST be specified when the Hunt Group mode is enabled Otherwise GSM Call Forward cannot be setup properly 4 IMEI 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 5 Unlock PIN1 SIM Card unlock PIN code 1 6 Unlock PIN2 SIM Card unlock PIN code 2 7 Talk Time Limit m There are two ways to define the Talk Time Limit for a SIM card 1 Total talk time without any time limit When the accumulative talk time since the last reset reaches this limit the SIM card is disabled The SIM card must be reset via the status page or via a special SMS command see Appendix A Syntax
12. 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 Please note that each GolP cannot have more than one Host channel assigned Sharing of GSM channels between two Hunt Groups is supported When all channels in one group are in use Call 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 To enable this feature the Backup Host Address must be set to the IP of the GolP which contains the Host channel of the other Hunt Group Hunt Group Mode Forward Mode Backup Host Address gt Forward Mode This parameter determines if the Host channel is going to be used for incoming call or not If the Forward Mode is set to Always the Host channel will not be used for answering any GoIP User Manual incoming calls If it is set to Busy then the Host channel will always to be used to answer an incoming call whenever it is available gt Backup Host Address Setting this parameter enables Hunt Group Sharing feature This parameter must set the IP of the GoIP which contains the Host channel o
13. E 47 3 4 1 Oane UD A eer E N E IE ED EDER ERE UDE DES E DE SEE DE EPE EO FEE EL EDEL EL LERE 47 3 4 2 Change PASS WV OO sena 48 3 4 3 A An PP anise ATEA seu haensteukaawsteuhuantteutaas tera 48 3 4 4 ST IVI e O SE UERE SER BESS ERE NEUES ES EN BESS ERE SEVEL ESEN BESS ERE SEVEL EGER EU SSEDESE VEL ESENPESSEDESE VEL ES ER PESSEDE SEVEL E SERENE SS ERE SEVEL 49 3 4 5 AP EO on II 50 3 4 6 GSM Channel Control eraren EE ENEE EE NEEE EE NEEE NRE 50 3 4 7 Backup Restore 51 3 4 8 PP 51 3 4 9 a E e adenoubaqusaoedsadenoaaausaoedsadeboebaaustoutect 52 Appendix A Special SMS Commands aia 53 Appendix B A E E EAE EAA A 54 Appendix C Custom Network TONGS nara 58 Appendix D Appendix E Appendix F GoIP User Manual E sis ciaauaee si cccemseegecereesas RR REAR ERE 59 167 BX 82 OTA ardilla 60 Volume A a a 61 GoIP User Manual 1 General 1 1 Introduction GolP 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 are 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 tr
14. Server This is not the password Password to login to the built in webpage Please ask your Remote Server Administrator if it is not available 6 Network Tones 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 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 7 HTTP Port This sets the port that is used to access the built in web server The default port number is 80 The port range is from 1 to 65535 16 gt DDNS Address gt DDNS Port gt Update Interval 9 Auto Reboot gt Reboot Time 10 IVR 11 Remote SIM gt Server gt ID gt Password gt Net Protocol 12 SMPP SMSC 13 DTMF Tone Min Gap 200 400 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
15. 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 GoIP User Manual 5 G 729AB 8 39K with Silence Compression and Voice Activity Detection VAD 31 GoIP User Manual 3 3 7 Call OUT The Call Out page defines how each GSM channel handles calls when they are routed from VoIP This section MUST Be defined properly in order to enable each GSM channel to dial out calls based on your requirements In general you can achieve the followings Forward all incoming VolP calls to a fixed GSM or PSTN number Dial out all incoming VolP Calls based on the phone number received Use the Dial Plan to manipulate the phone number received and then dial out the modified number Use the Restricted mode to only dial out the phone numbers that match the rules defined in the dial plan 5 Use the Idle Interval to prevent calls from dialing out during this period This Idle Interval can be set to be active for any calls or for only answered calls 6 Only all authorized calls to be dialed out oe Suk ol The Auto Config Other Line button is provide to facilitate the programming of each channel After the parameters for CH1 are set clicking this button automatically duplicates the same settings to all other channels Please note that how a VolP call is rout
16. cause defined in GSM04 08 Annex H is returned from the network for call control This GSM cause must be returned back to the SIP server for call handling and control In order to meet various application requirements the GSM SIP Code Map allows the user to define the corresponding SIP message for each GSM cause listed below 27 3 3 5 GSM Reason Unassigned unallocated number No route to destination Channel unacceptable Operator determined barring Normal call clearing User busy No user responding User alerting no answer Call rejected Number changed Destination out of order Invalid number format incomplete number Facility rejected Normal unspecified No circuit channel available Network out of order Temporary failure Switching equipment congestion Access information discarded Requested circuittchannel not available Resources unavailable unspecified Others SIP Response Code D ej ej ej eaj IA IE a IE IE Mo e cp peo eA peo ca ER HE TEE pe pe oe GoIP User Manual 28 GoIP User Manual 3 3 6 Media This section allows the user to program various settings for media voice transmission and format Depending on your network environment and condition you may or may not need to change these settings Please see the parameter table below for more information Media Settings RTP Port Range 132768 PacketLength ms Jitter Buffer Fixed re Delay ms 60 Media QoS None TW Media E
17. 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 58 GoIP 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 PA SO os Server Mode GolP Client Mode GolP Client Mode GolP Client Mode GolP Incoming call forward to Client GolP 59 Appendix E CID Call Forward GoIP 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 P
18. mode and Config by Group mode since SIP registration is required If this parameter is blank and the Callee Number does not equal to the SIP Number defined for this line GoIP dials out the Caller Number according to the Dial Plan defined 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 al n 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 meaning b must be the same a or the beginning portion of a Example Callee Number 9262124567 Dial Plan 9 9 852 Actual Number dialed 85226124567 Syntax a b c d e b c specifies the range of a single digit Together with a they form a prefix for rule iw matching a can be a single or multiple digits d e are the actions to be t
19. select PPPOE year Mame E and then enter the information as provided by your ISP PC Port 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 i E PC Port Static IP device to create another network segment and it then functions as a router gateway for this new network F Address 132 168 8 1 segment Select Static IP for this new segment and Subnet Mask 255 255 255 0 then enter the PC port IP address and Subnet Mask DHCP Server Enable Disable accordingly Starting Address 192 168 8 100 Ending Address 192 166 686 120 It also has a built in DHCP server to assign IPs to the devices attached to this network segment Enable it Static DNS optional A and then enter the Starting Address 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 18 GoIP User Manual 2 Bridge Mode Select this mode if your network pr port Bridge mode topology requires the network devices attached to the PC port to be in the same network segment as the LAN static IP port Advanced Features 302 1 VLAN Enable Disable 1 VLAN This is a type QoS service and is intended to give VLAN ID Doo higher
20. the active call is dropped or not when the Total Total Talk Time Limit Talk Time Limit expires Expires 9 Talk Time Limit m This sets the limit for the maximum talk time per call When this limit is Call reached the call is dropped automatically 40 GoIP User Manual 10 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 11 SMS Alert Number This specifies the GSM number to receive a SMS Alert on the Total Talk Time If this parameter is blank no SMS Alert will be sent 12 SMS Alert ID This parameter is used to identify the channel sending the SMS Alert 13 SMS Alert Trigger 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 14 Hide My GSM Number This parameter determines if the caller party can receive the phone 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 3 3 12 SIM Forward This section allows the user to define the GSM Call Forward settings for each channel There are four Call Forward con
21. the local SIM cards that are inserted to the GolP SIM Y means the corresponding GSM module is able to access the designated SIM card properly GSM Y means the corresponding GSM module registers to the GSM network successfully RSSI Received Signal Strength Indicator Please see the description in Section 3 1 1 GPRS Login Y means access to a GPRS network This status is obtained from the command AT CREG GPRS Attach Y means GPRS Attach is successful and is ready for PDP This status is obtained from the command AT CGATT Carrier This shows the name of the current GSM carrier GSM BSC mode This shows the current setting for the GSM BSC mode which determines how the GolP selects a base station For more information please refers to the section 3 3 15 Cell ID This shows the Base Transceiver Station BTS ID LAC This shows the Location Area Code 13 GoIP User Manual The bottom table shows more detailed information on the onboard GSM modules and the SIM card inserted 3 2 4 Module The model number of the GSM module Firmware Ver The version number of the firmware installed in the module SIM Number The GSM number that is assigned to the SIM card User must enter this number manually IMEI International Mobile Station Equipment Identity IMSI International Mobile Subscriber Identity ICCID Integrated Circuit Card Identifier SIM Call Forward The table below lists the current call forward se
22. 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 HHHINFOHHH Sends an SMS response to the sender with the LAN port IP address HHHIinfoHHH 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 53 GoIP User Manual Appendix B SMS To VoIP The device receives SMS messages from both GSM and VolP networks and they are handled according to the modes defined below 1 Call Function 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 number 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 num
23. transmission priority to real time packets manos 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 PPTPYPN Enable Disable tunnel with the designated VPN Server The VPN PPTP Server Doo protocol supported is PPTP with no encryption or 40 bit PPTP username po encryption which is defined on the VPN server In a O general this option is used to avoid VoIP blockings Advarced o AAN Address es y Address 3 3 3 Basic VoIP The GolP can support both SIP and H 323 VoIP protocols For GolP 1 both protocol are embedded in one firmware User needs to select the VoIP protocol as shown below Endpoint Type SIP Phone Config Mode H 323 Phone SIP Phone As more features are added SIP and H 323 VoIP protocols are supported in two different firmware versions Except GolP 1 all other models are now shipped with the SIP protocol firmware as a factory 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 us or your supplier for the latest firmware upgrade links available 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 I
24. 3 3 6 for more information on the Call Out Dial Plan 21 GoIP User Manual Config Mode 8 Line 1 J Line 2 Linea Line4 LA Line 5 LA Line 8 LA Line Y J Line 8 Phone Number OoOo O Display Name O Authentication ID O Password cc Routing Prefix OoOo O SIP Proxy o SIP Registrar lt 0c 0 O Re register Period s Outbound Proxy lt c 0 O Home Domain A Backup Server Enable Disable Auto Config Other lines 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 For more details please see Section 3 3 8 for Call In configuration Please note that the parameters defined in this mode are for each line Their definitions are the same as those defined in the parameters table for Single Server mode The Auto Config Other lines option is only available for Line 1 Clicking this button after Line 1 is configured will automatically configure other lines with the Line 1 settings with the changes displayed in the following message box Message from webpage b Auto config from line 1 P Phone Number increase step by 1 from line 1 Display Name increase step by 1 from line 1 Authentication ID increase step by 1 from line 1 Gateway Prefix increase step by 1 from line 1 Other configs the same as line 1 Config By Group for all models except GolP 1 This mode is basically a combination of Single Server mode and
25. 3 4 5 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 linel Line OLine3 OLine4 OLineS Linet Line Line 8 mer LO ment MEE 8615817459136 0 but its taking time to save 6 01 16 54 43 Ls 8615817459136 sent message to you from my skype 6 01 16 55 13 6 01 16 53 37 8615817459136 0 but cannt logon facebook 06 01 16 53 17 horsens Public furious over alleged rape of girls by official 6 01 16 54 32 Li 8615817459136 9 see you on facebook 3 4 6 GSM Channel Control This feature is implemented for two functions 1 Removing the power to a GSM channel before removing or inserting a SIM card This is the recommended procedure in order to prevent damages to the SIM card 2 Disabling a GSM channel temporary Click 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 The All Channels selection is a short cut to turn on or to shut down all channels 50 GoIP User Manual GSM Channel Control Shut Down Channel L Shut Down Channel L Shut Down Channel3 L Shut Down Channel4 Shut Down Channel5 _ Shut Down Channel6 L Shut Down Channel Shut Down C
26. Blacklist 34 GoIP User Manual 3 3 9 Call IN 1 This page defines how incoming calls to each GSM channel are handled GSM incoming calls can either be answered and prompted for second dial or forwarded to a phone extension or IVR in the VolP network connected The parameters in the Call In Page are divided into two sections The top section contact the parameters that are applicable to all channels and the bottom section are for each channel individually The CID Forward Mode is applicable intended to forward the GSM caller ID to VoIP and for all channels Please refer to Appendix E for more details 2 The Call IN via GSM parameter must be enabled in order to receive GSM incoming calls 3 The Hunt Group Mode is use to simulate Call Center operation where a single GSM number is used for all incoming calls To enable this function set the Hunt Group mode of one GSM channel to Host and all other channels to Client Please do not set more than one channel to Host as it may cause the Hunt Group mode not to function properly In addition the GSM Forwarding for all Client channels must be disabled 4 The Call IN Auth is used authenticate the incoming calls Calls can be accepted or rejected based on this setting and the corresponding White list or Black list 5 The Auto Incoming Call Block is used to block an incoming call when the same number is called consecutively for more than the limit defined Customer has requeste
27. CCW in Hong Kong the USSD command to check balance is 122 Enter 441224 and the click Start The following screen is then displayed LineLine Send Command 224 sending Please Wait A few seconds later the service provider sends back a USSD message response as shown below LineLine send Command 224 Thank vou for using our service Your current balance is 32 26 valid until 18 07 2012 dad ENA E E E A E a E E E O ENE O O E O E E O OO OO E 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 3 4 4 Send SMS Click Send SMS to access the Send SMS webpage as shown below Send SMS Line1 Oline OlLine3 OLine4 OlLine5 Lines Line Lines CAllLines Line 1 GSM Status LOGIN Line 1 GSM Number SMS Content 49 GoIP User Manual 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 The last option All Lines means that all channels are selected and the same SMS is sent via all channels provided that they are in the Login Status 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
28. Config By Line mode It allows lines to be split up into groups Each group only uses one SIP registration for all the lines assigned to the group Each line can be assigned to only one of the 4 predefined groups in the Grouping section In this mode the Routing Prefix is assigned to the Group rather than to the Channel Its function is the same as the Routing Prefix for Config by Line The syntax for the Routing Prefix is defined in the Parameter Table for Single Server Mode Please note that the parameters listed in this figure are the same as the parameters defined in the Config By Line mode except these are parameters for a Group rather than for a Line Backup Server and Grouping However these parameters are group properties rather than line properties 22 GoIP User Manual Config Mode 8 Grou p 1 O Group 2 O Group 3 O Group 4 Phone Number y Display Name Authentication ID A Routing Prefix A SIP Registrar Re register Period 5 0 0 O Grouping Line1 Oline2 OLine2 Line 4 blines Lies lLine CLine 2 In Group Group 1 we 4 Trunk Gateway Mode for all models except GolP 1 This mode offers a seamless interface with the SIP Trunk configuration in a SIP server In general SIP registration is 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 bo
29. FYBER TONG BEI GoIP User Manual VoIP GSM Gateways Models GoIP GolP 4 41 GoIP 8 8i GoIP 16 GoIP 32 2013 2 25 www hybertone com GoIP User Manual supportOhybertone com GoIP User Manual Content A ESTERE SEES TERE SEEST ERE SEES TERE SEE SEERE SEEST ERE SEE SEERE SEEST ERE SEE SEERE OAE 3 1 1 THOU CEI A o Pe E PU E O EEEE EEE 3 1 2 rn A 4 1 3 Hardware CAI OS ama cde tea ae a a ae a ae a a A aa a 4 1 4 Oa FCA ES E E E E EIE E A E E AOE OE OEA OE A EAA TOEA 4 1 5 Package Content ricota 5 1 6 LEDIN COTS n E T T E E EA 6 WANG OD dada 7 TIC O a T EEE E O E E A A OTOA 9 3 1 FE WW S ar BE 110 PP A E E E EE A E E A 9 3 2 SAU aia aiii ai aiii 10 3 2 1 A 10 RAL o AP E E E E E E E E E E E E E SE 11 3 2 3 E OPONE Un II UNA 13 3 2 4 SENE CAMELA oa 14 3 3 A A SE Eo II EU A 15 3 3 1 IL YO APPO nocaonke nocasnienoansnke nocassiaaasebenecasebe soca sabe necaoobesocesobences 15 3 3 2 o PP e e EP E o II E 18 3 3 3 Basto VOIP noria cies 19 3 3 4 PVM COI V OUP S NS ERE SENSE RENE ESSEN E REREESSERE E SENSE BENE VISER SE VERS BENE ESSEN EUR SEBEREES SERENE RSS EREE ES SENE EUR S ER 24 3 3 BTCC lo COP e seasons does E E E A 29 3 3 6 AAA e e O A 32 3 3 7 CAMI ron 34 3 3 8 AA UP PPP o A e II SES LE BESES FEE SESPRER 35 3 3 9 CINA PP e eee 38 3 3 10 y IPN UU AP A 39 3 3 11 SINE HOEY AECL lis 41 3 3 12 A A A 42 3 3 13 PPP A II E E E E 43 3 3 14 GOM CT A ono y AAE AAE AAEE 45 3 3 15 GSM Base Stade 45 3 4 ONG specs oes tere o RUE PO E
30. There are two methods to access the built in web server Method 1 is to access the built in web server via the LAN port When you connect it to a network with a DHCP host it will obtain an IP address from ip Once the IP address is entered the login window shown on the right pops up factory default The LAN port is set to DHCP mode as a 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 01 to hear a voice prompt reporting the LAN port IP address li Send the HHHINFOHHH SMS command to one of the GSM channels available Please refer to Appendix A Special SMS Commands for more return back the LAN port IP address information The GolP will then 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 Method 2 is to access the built in we server via the PC port Connect a computer to the LAN port of the device and configure its IP to Type the IP address 192 168 8 1 in the address field of a web browser preset to 192 168 8 1 192 168 8 x x 2 to 254 password There are three level of access via three different user names 1 Administrative Level This offers a full access right to all parameters available in the built
31. aditional 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 1144444 I VoIP A CBA Service GolP4 GolP8 Provider Xx Local Local GSM Network estad etwork World VolP N Telephone Phone N Network oN gt Q Telephone Cellphone Telephone You can now make a call from anywhere in the world via a VoIP network and then terminate the call via a GoIP 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 VolP Service provider is required for one side of the call termination For two fixed locations it is possible to setup GolPs at both ends for call terminations without subscribing to a VolP Service provider GolP can also be used to achieve GSM roaming via VolP The idea is to route all your incoming GSM calls to a GolP via call forward or simply insert your SIM card to a GolP You can then setup the GolP to forward all incoming calls to another GSM number in the world via a VolP service provider The charge per call from a VolP service provider is significantly lower than the roaming charge For office environment GolP offers a q
32. affics 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 1 Stun Server An external Stun Server is required This allows the device 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 downloaded from HYBERTONE s website www hybertone com Please contact support hybertone com for further assistance if needed 8 Audio Codec Preference Six types of audio codec are supported and they are summarized in the table below All codecs are enabled in the Raw Data Ethernet 802 3 order of Bandwidth bps Data Bandwidth preference bps shown below 1 a law 64K 85K a law 2 u law 64K 85K u law G 729 3 G 729 8K 39K G 729A G 729AB i en H 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
33. aken as described in the previous syntax Example 913 5 9 9 86 In this rule Callee 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 GolIP User Manual 33 GoIP User Manual 5 Restricted Dial Plan This option forces the GoIP to only dial out the phone numbers that match the rules defined in the dial plan 6 Sleep Interval in between This setting defines an idle interval in between calls During this interval no outgoing calls Calls are allowed to be made via the GSM channel 7 Sleep Interval Mode This parameter defines the condition of a call when the Sleep Interval is activated 1 Any Calls The Sleep interval is activated whenever an outgoing call is dialed regardless whether the call is completed successfully or not 2 Answered Calls The Sleep Interval is activated whenever an outgoing call is answered 3 3 8 Call OUT Auth The Call Out page defines how each GSM channel handles calls when they are routed from VoIP This section MUST Be defined properly in order to enable each GSM channel to dial out calls based on your requirements In general you can achieve the followings Call Out Auth cHi cH cH OcH4 OcHs cH O cH O cHe Call OUT Auth No Auth te Whitelist Blacklist
34. arty 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 29hG4bK1645487913 From lt sip 20001192 168 2 1 5060 gt user phone tag 406202416 To lt sip 5000 192 168 2 1 gt Call ID 8472302784192 168 2 180 CSeq 2 INVITE Contact lt sip 2000 192 168 2 180 5060 gt Max Forwards 30 User Agent HBT Remote Party ID 13800000000 lt sip 13800000000 0192 168 2 1 gt party calling screen no privacy 0ff Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application s dp Content Length 226 2 USD CID as SIP Caller number This parameter specifies the 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 z9hG4bkK 1450498491 From 13800000000
35. ber 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 86138000000000192 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 SIP 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
36. bled 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 192 168 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 Sender 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 A SMSS 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 t
37. d this feature to block test calls that are sent by the carrier 6 The Auto Config Other Line button is provide to facilitate the programming of each channel After the parameters for CH1 are set clicking this button automatically duplicates the CH1 settings to all other channels Call In CID Forward Mode Use Remote Party ID se Scam Oc Oc cH Ochs Oche Och Ocns Call IN via GSM Enable C Disable Forwarding to VolP Number fe Dial Plan fe Hunt Group Mode Disable we Auto Incoming Call Block O Enable Disable Auto Config Other lines Save Changes The parameters defined in this section are described in details in the Table below Default Value CID Forward Mode This specifies the method used to transmit an incoming GSM caller ID to VoIP Enabled CID Forward Mode Disable kA Disable Use Remote Party ID Use CID as SIP Caller ID 2 3 gt CID Prefix CALL IN via GSM Forwarding to VoIP Number GoIP User Manual 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 message is replaced by the incoming GSM Caller ID This parameter is intended to modify the incoming GSM number Syntax a b c The portion on the left side a of
38. ditions 1 Always Forward all incoming calls unconditionally 2 Busy Forward all incoming calls when busy 3 No Answer Forward the incoming calls when it is not answered 4 No Service Forward all incoming calls when the SIM cannot register to the network There are 3 choices for each Call Forward condition 1 Set This enable the Call Forward function This setting is sent to the network whenever the SIM is Starting a new registration to the network 2 Disable This disable the Call Forward function This setting is sent to the network whenever the SIM is Starting a new registration to the network 3 Not Set This leaves the current SIM Call Forward setting in the network unchanged This setting is NOT sent to the network at all SIM Forward c cH cHa O cH OCHS OcHe CH7 CHE GSM Call Forward Always Set Disable Not Set Forward Num OoOo Busy O Set Disable OC Not Set No Answer O set 8 Disable Not Set No Service O set Disable Not Set Parameter Description Default Value SIM Forward 4 Unconditional Call Forward gt Forward Num Call Forward Busy gt Forward Num Call Forward No Answer gt Forward Num Forward all incoming calls unconditionally to the number specified This specifies the phone number to receive forwarded calls under this condition Forward calls when the GSM Channel is in use This specifies the phone number to receive forwarded calls under t
39. e as shown below extracted from GolP 8 They are very essential to display the operation status of the GolP in order to determine if it is working properly or not Line M SIM GSM VOIP Status SMS RSSI Carrier Cell ID Idle Remain Reset 1 ly N N N IDLE N 99 55 0 2 Y N N N IDLE N 99 55 0 3 Y N N N DLE N 99 55 13 A YIN N N DLE n ss 55 NO LIMIT 5 Y N N N DLE N 99 55 NO LIMIT 6 Y N N N IDLE N 99 55 0 7 Y N N N IDLE N 99 55 NO LIMIT a Y N N N DLE N 99 55 NO LIMIT ALL Here are the list of GolP parameters shown in this page 1 2 CH GoIP channel reference M GSM module status for the corresponding CH Y means Enabled and N means Disabled If a GSM module is disabled all other parameters for this channel are not active Clicking Y shuts down the channel selected Clicking N turns on the channel selected SIM SIM card status Y means that the corresponding GSM module can access the designated SIM card successfully N means unable to access the designated SIM card Please check if the SIM card is inserted properly or the SIM card is damaged If Remote SIM function is used please check the SIM Bank and or SIM Server configuration The problem could also be caused by bad network condition or improper network configuration GSM GSM registration status Y means Registered and N means Not Registered VoIP VoIP registration status Y means R
40. e the problem g VPN Status This shows the current VPN connection status It only appears when VPN is enabled Call Management section summarizes the both GolP and GSM configurations and their corresponding status It is important to note that the VoIP lines and the GSM channels are not mapped to each other as a one to one relationship For outgoing calls from VoIP to GSM the GSM channel selection is based on the Routing Prefix VoIP a Line This is used as a reference in VoIP line configuration b Mode This shows the current VoIP Registration mode S means Single Server Mode L means Config By Line mode Gx means Config by Group mode where x is the group reference number T means Trunk Gateway mode c Login This shows the current VolP registration status Y means that the corresponding line registers to the server successfully N means the corresponding line fails to register to the server d Routing Prefix This shows the current setting for the Routing Prefix Please refer to Section 3 3 3 for more information GSM e CH This corresponds to the physical GSM channel number f Login This shows the current GSM Registration status for voice calls g Call In This shows the Call IN setting for the corresponding GSM channel Y means incoming calls are enabled N means incoming calls are disabled and the corresponding channel rejects all incoming calls by sending back the hangup ATH command to the GSM n
41. ecified Select Fixed to enable this mode and then press Save Changes To view the available Cell IDs please wait couple of minutes and then refresh the screen then select and enter the desired Cell ID as shown below Remember to press Save Changes again to save the Cell ID entered 46 GoIP User Manual c cH cH O cH Ochs OcHe O cH OCHE Base Station Selection Enter Cell ID Available Cell IDs 32707 3 4 Tools Click Tools on the left hand menu to access the submenu as shown below Please note the available options under the Tools menu Status Online Upgrade Configurations Last Upgrade Time Current Version GS 4 01 04 Tools Online Upgrade Change Password send USSD send SMS SMS In Box GSM Channel Control Backup Restore Reset Config Reboot 3 4 1 Online Upgrade Click Online Upgrade to upgrade the device firmware The current version is displayed as well as the last upgrade time Online Upgrade Last Upgrade Time 2014 02 11 17 09 28 Current Version 3 4 01 65 j1 Upgrade Site 192 168 2 1 update GS 4 01 65 1 pkg 47 GoIP User Manual Contact us or your 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 as this process may take a few minutes Note It is important
42. ed to a GSM channel depends on the Configuration mode selected Please refer to the Basic VoIP section for more information Call Out GSM Auto Redail Enable Disable c cH Oc cH OcHs cH O cH Cocue Call OUT via GSM Enable Disable Forwarding te GSM Number lt 0 Of Dial Plan DO Restricted Dial Plan J Enable Disable Sleep Interval in between Calls s Sleep Interval Mode O Any Calls amp Answered Calls Auto Config Other lines Save Changes The parameters available in this section are described in details in the table below 1 GSM Auto Redial This is a general parameter for all channels Enabled 2 Call OUT via GSM This setting defines if the device is allowed to make outgoing calls via the on board GSM channel s The typical application is to terminate VoIP calls via the GSM network This setting If this parameter is specified GolP dials this phone number via the corresponding GSM 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 32 3 4 Forwarding to GSM Number Dial Plan 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 informati
43. egistered and N means Not Registered If GSM Registration status is N VoIP registration is disabled and its status should be ignored Status VoIP line status If VoIP registration status is N the VoIP line status shown should be ignored Once VoIP registration status is Y the current VoIP line status is then shown in this field Here are a list of available statuses a IDLE The VoIP line is not engaged in any call activities b CONNECTED An active call between VoIP and GSM is in progress c ACTIVE A second dial tone is generated when a VoIP call is answered without making a GSM call or when a GSM call is answered without making a VoIP call The generation of a second dial tone prompts the caller to press a phone number The Status changes to ACTIVE since the start of the second dial tone till a phone number is received for dialing or the call is terminated 10 10 11 12 13 3 2 2 GoIP User Manual d DIALING lt phone number gt This occurs when the GolP is dialing out a phone number via the corresponding GSM channel or The DIALING status shows that a number is being dialed out via the corresponding GSM channel or a VoIP line The phone number dialed is also shown in the Status e ALERTING After a phone number is dialed the Status changes to ALERTING when a ringback signal is received from the network f INCOMING This occurs when a GSM incoming call is calling and the call is not answe
44. elay 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 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 downloaded from HYBERTONE s website www hybertone com Please contact support hybertone com for further assistance if needed 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 When making a GSM call a GSM specific
45. election algorithm is the same as the one described in the Single Server Mode The syntax for the Routing Prefix is defined in the Parameter Table for Single Server Mode 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 Fill in the SIP registration information as required The Re Register Period must be set to a non zero value If SIP Registration is not required the Re Register Period must be set to zero The parameters available in this mode are listed in the table below Parameter Description Default Value Trunk Gateway mode 1 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 2 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 3 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 SIP Registration to Trunk Gateway1 Some Trunk gateway connection requires a SIP registration which can be defined via the following parameters This specifies the phone number for the SIP registration ME This specifies the authentication ID for the SIP registration re This specifies the password for the SIP registration
46. es Dial Tone Ring Back Tone Busy Tone Indication Tone Customized w 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 11 12 13 14 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 tone ms f1 is the frequency of the first tone 300 to 3000Hz f2 is the frequency of the second tone 300 to 3000Hz 3 is the frequency of the third tone 300 to 3000Hz 4 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 3 is the level for tone 3 range from 0 to 31 with 0 3dB 1dB for each increment
47. etwork h Call Out This shows the Call Out setting for the corresponding GSM channel Y means outgoing calls are enabled and N means outgoing calls are disabled When the Remain Time for outgoing calls reaches zero the Call Out setting is set to LOCK automatically To unlock the channel click the corresponding Reset button in the Summary page i Remain Time This is the same as the Remain shown in the Summary page If the Talk Time Limit in the SIM Page is set this parameter shows the remaining time allowed for outgoing calls 12 3 2 3 GoIP User Manual GSM The GSM page shows the current GSM channels status and information on the GSM modules and the SIM cards inserted Remote SIM DISABLE E Gps ls ame a x i i i GSM Details IMEI i 610 G610_V0C 00 00_T14 35507 3036020376 ES G610 G610_V0C 00 00_T14 355073035021077 3 G810 G610_V0C 00 00_T14 355073035995840 NN la 6610 6610 VOC 00 0D T14 355073036005765 NN 5 G610 G610_V0C 00 00_T14 355073038003978 e G610 G610_V0C 00 00_T14 3550730260 19800 UN 7 G610 G610_V0C 00 0D_T14 355073036009098 UN a G610 G610_V0C 00 00_T14 355073036019840 UN The top table shows a number of GSM parameters which are useful to determine if the GSM channels in the gateway are working properly 1 Ss SP 10 Remote SIM This tells if Remote SIM function is used or not DISABLE means using
48. f the other Hunt Group Client This assigns the selected channel to be a client in Hunt Group mode Hunt Group Mode Client Host Address cc ol Must Fill GSM Number gt Host Address The Host Address field specifies the IP address of the device that contains the Host channel If a Client and its Host belong to the same device its device IP is entered in this parameter A client registers and updates its channel status to the Host 6 Auto Incoming Call This option is used to block an incoming calls when the condition specified in the Trigger is met Disable Block The block operation is released when an incoming call with a different Caller ID is received gt Trigger This sets the number of consecutive calls with the same Caller ID required to trigger the Incoming Call Block function Auto Incoming Call Block 8 Enable Disable Trigger Number of Consecutive Calls with same Caller mo Current Blocked Number 3 3 10 Call IN Auth This page defines how incoming calls to each GSM channel are handled GSM incoming calls can either be answered and prompted for second dial or forwarded to a phone extension or IVR in the VolP network connected The parameters in the Call In Page are divided into two sections The top section contact the parameters that are applicable to all channels and the bottom section are for each channel individually Call In Auth cH1 cH OcHa OcHa Ochs OcHe O cH Cocue Call IN Auth Ne Auth w W
49. fferent network segment In this case please make sure that the PC network segment IP 192 168 x is different from the one in the LAN port 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 5 The Reset button is recessed inside the GolP 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 3 Configuration The device can be configured via its built in http web server or via an Auto Provision Server Server is a free utility supporting both Window and Linux OS Technology for the sole purpose of automating the configuration of our products GoIP User Manual Auto Provision This utility is developed by HyberTone 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 and Chrome are supported Both Firefox and Mozilla may not work properly depending on the version and the operating system used 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
50. gistrar OoOo 2 Re register Period s Outbound Proxy DO Backup Server O Enable Disable Routing Prefix Line Line C Liez Lines Lines Olines OLine Lines Le 1 Routing Prefix 20 GoIP User Manual Single Server Mode MA A Ld AAA 5 SIP Proxy The domain name or IP of the SIP Proxy or Server is specified here If the SIP Proxy is using the standard 5060 signaling port then there is no need to specify the port number Otherwise the port number can be specified by adding and then the port number at the end of the SIP Proxy address A CS AA LA 7 Re register Period s Register to the SIP Server at an interval specified by this parameter 6 SIP Registrar The address of the SIP Registrar Server is specified here 8 Outbound Proxy The address of the Outbound Proxy used for VolP communication is specified here 9 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 Disabled gt SIP Proxy This specifies the backup SIP Server address gt SIP Registrar This specifies the backup SIP Registrar Server address gt Home Domain This specifies the backup Home Domain address 11 Routing Prefix This parameter is used for call routing When this is set the corresponding channel is only used to dial out a phone number with the matching Routing Prefix Syn
51. gt Save Changes Default Value 1 Call OUT Auth This parameter defines how incoming VoIP calls are authenticated before dialing them out None 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 2 Whitelist Blacklist Both Whitelist and Blacklist for call screening are supported Each list contains up to 15 entries 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 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 2 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
52. hannels _ All Channels Save 3 4 7 Backup Restore The device configuration can be backup or restore via this page Click Backup Resotre Confguration to access the page shown below Backup Configurations Password optional Restore Configurations Load from file Choose File No file chosen Passwordoptiona sd 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 3 4 8 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 wim dbltek comisla s says Are you sure to reset to factory default Click OK to reset the device configuration back to the factory default 51 GoIP User Manual 3 4 9 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 wianwdbltek comisls23 says Are you sure to reboot the device TET aner 52 GoIP 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
53. he text message begins at the 56 GoIP User Manual 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 ASIP 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 57 Appendix C Custom Network Tones This section describes how to define custom network tones 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 GoIP User Manual The Custom selection allows the following Network Ton
54. his condition Forward the call when an incoming call is not answered This specifies the phone number to receive forwarded calls under this condition GoIP User Manual Not Set Not Set Not Set Call Forward Unreachable gt Forward Num Forward calls when the GSM channel cannot register to the carrier Not Set This specifies the phone number to receive forwarded calls under this condition At the bottom of the IMEI list user can also enable an option to change the IMEI of each channel automatically at an interval defined by the Change 3 3 13 IMEI This section allows IMEI modifications for all Channel in a single page Period The minimum Change Period allowed is 10 minutes which is currently set as the default value Changing IMEI only occurs when the channel is idle When it occurs the GSM channel drops its current registration and then use the new IMEI to register to the network IMEI Settings Line1 IMEI 869269013962353 Line IMEI 869269013253118 Line3 IMEI 869269019435750 Line4 IMEI 869269019622308 Line5 IMEI 869269011789220 Line IMEI 869269014161005 Line IMEI 869269011240422 Lines IMEI 869269017556813 IMEI Auto Change minute 42 GoIP User Manual 3 3 14 SMS SMS to VoIP Disable cH1 cH Oc a cH chs Ochs Och Ochs SMS Server enable Disable Send SMSC Number enable Disable 5M8 Forward to Eai OEnable Disable Auto Config Other lines This page covers parameter
55. hitelist Blacklist gt gt Save Changes Default Value 1 Call In Auth This parameter defines how incoming GSM calls are authenticated before routing calls to the None VoIP network connected Five options are available None No authentication is required all incoming GSM calls are routed to VoIP Password The caller is prompted for entering the password before the call is routed or a second dial tone is generated 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 Whitelist Password Both Whitelist and password are used to authenticate incoming 38 GoIP User Manual GSM calls 5 Blacklist Calls with caller IDs that are listed on the Blacklist are not answered by the channel selected 2 Whitelist Blacklist Call screen list can be set to Whitelist or Blacklist A maximum of 15 entries is allowed 1 Whitelist This list contains a list of incoming GSM caller numbers that are accepted 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 This setting specifies the Call Forward method when the GSM chan
56. in webpage The user name and password for the administrative level are admin and admin respectively User Level This level restricts user from accessing the Call Setting page User will not be able to change any VolP related settings The user name and password for the user level are user and 1234 respectively SMS Level This level only allows user to access the Send SMS and SMS Box functions under the Tool menu The user name and password for the SMS level is sms and 1234 As a factory default the PC port IP is Enter the user name and y The server 192 168 3 11 at Please Login requires a username and password Warning This server is requesting that your username and password be sent in an insecure manner basic authentication without a secure connection User name admin bi Password EEEN Remember my password Cancel GoIP User Manual 3 2 Status There are four pages under the Status category and they are Summary General GSM and SIM Call Forward Each page is refreshed every 5 seconds however this feature is not supported when the Firefox browser is used Sample pages shown in this section are captured from a GolP 8 Please refer to the actual web pages of the models of interest It is important to understand the information shown in these pages in order to debug or report problems encountered 3 2 1 Summary The current VoIP and GSM statuses are listed in the Summary pag
57. in the Message Center When this period expires the undelivered message is discarded The Validity Period VP is an integer from O to 255 and VP Value Actual Time VP 1 x 5 minutes i e 5 minutes intervals up to 12 hours When this option is enabled the SMSC Number stored on the SIM card is read and sent to the Message Center when sending a message This enables the forwarding of SMS received via the channel selected to a designated email address This specifies the SMTP Server for sending emails This specifies the User Name for SMTP Server authentication Leave this blank if authentication is not required This specifies the Password for SMTP Server authentication Leave this blank if authentication is not required This specifies the email recipient of the forwarded SMS GoIP User Manual Please make sure that the desired channle is 44 GoIP User Manual The Auto Config Other Line button is provided to facilitate the programming of each channel After the parameters for the CH1 SIM are set clicking this button automatically duplicates the same settings to all other channels with the exception that the SMS Client ID is incremented by 1 with respect to the ID of the previous channel 3 3 15 GSM Carrier This section sets the mode of the GSM service provider selection The factory default setting is Auto for automatic selection of GSM service provider based on the default preference set by the SIM card
58. 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 Server The specifies the Provision Sever address IP or Domain name gt Provision Interval This specifies the interval in performing an auto provisioning event 5 Remote Control 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 gt Remote Server This specifies the IP address or the domain of the Remote Control Server gt Remote Server Check with your Remote Server administrator for the communication port Port gt Remote Server ID This specifies the name to be appeared in the Remote Control Server It is used as a reference for the device gt Remote Server This specifies the login password to the Remote Control
59. irst out mode with a fixed jitter buffer delay Sequential The sequential mode is also a fixed jitter buffer delay mode but in this mode the jitter buffer function looks at the packet timestamp for dropped or out of sequence packet problems The data packets are sorted based on the packet timestamp 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 gt Delay This specifies the fixed jitter delay for both Fixed and Sequential Jitter Buffer mode 60 gt Min Delay ms This specifies the minimum jitter delay for Adaptive Jitter Buffer mode gt Max Delay ms This specifies the maximum jitter delay for Adaptive Jitter Buffer mode 4 Media QoS Similar to Signaling QoS this parameter enables the QoS property for audio packets None Both IP ToS Type of Service and DiffServe Differentiated Service method are supported Media Encryption Similar to Signaling Encryption item 6 in this table this parameter enables the None encryption for audio packets Encryption methods supported are RC4 and ET263 Symmetric RTP Network environment in some enterprise may require Symmetric RTP Please check Disabled not with your network administrator for further support selected Media NAT Traversal This setting is not required if the target SIP server PBX supports NAT traversal However if your ISP blocks VoIP tr
60. is LED is red and illuminates when the LAN 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 VolP is ready for making calls Slow Blink Be This LED is red and illuminates when the PC port is connected and blinks when data transmission occurs GoIP User Manual Each GSM channel has its own status LED and its color is green 1 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 2 Installation The same installation procedure applies for all models with the differences in the number of channels ports available and the SIM card insertion t is important to note that the power to the SIM slot MUST BE disconnected removed before removing or inserting a SIM Card The power is removed by either disconnecting the power to the GolP or shutting each GSM module individually via its built in web interface 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 b
61. larly in order to Enabled keep the network ports used open This setting specifies the DTMF dialing method Outband 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 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 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 OO et TE a ee ES 26 9 Signaling NAT Traversal 10 Advanced Timings No Answer Expiry 32 180s NICT Expiry 2 180s ICT Expiry 5 360s Retransmit T1 200 2000ms Retransmit T2 20000 8000ms 11 GSM SIP Code Map GoIP User Manual 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 R
62. ll Routing Prefixes are set Try to match the number received for making an outgoing call against the Routing Prefix of each channel If only one match is found the corresponding channel is used to make the outgoing call If more than one matches are found the best available channel among the matched channels is selected If no match is found the call is rejected by sending back a SIP 503 message b Only some Routing Prefixes are set Try to match the number received for making an outgoing call against those Routing Prefixes that are set If only one match is found the corresponding channel is used to make the outgoing call If more than one matches are found the best available channel among the matched channels is selected If no match is found an idle channel without a Routing Prefix is used to make the call Otherwise the call is rejected by sending back a SIP 503 message c None of the Routing Prefixes are set An idle channel is selected to dial out the call If no idle channel is available the call is rejected by sending back a SIP 503 message It is important to note that the Routing Prefix P must be removed via the dial plan before the number is dialed out The dial plan syntax to remove the Routing Prefix is P P Please refer to section 3 3 6 for more information on the Call Out Dial Plan SIP Config Mode Single Server Mode v Phone Number DO O Display Mame Sd Authentication ID OoOo 2 Password SIP Re
63. n In general a VoIP caller dials a PSTN or GSM number E 164 number and the SIP Server 183 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 the 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 25 Call OUT Auth Mode Built in SIP Proxy gt Password NAT Keep Alive DTMF Signaling gt Outband DTMF Type gt RTP Payload Type Signaling QoS Signal Encryption 55 GO IP User Manual User Manual receive audio packets Since it may take a few t
64. ncryption None vw L symmetric RTP Media NAT Traversal None pa Audio Codec Preference Parameters Description Default Value Media 1 RTP Port Range This specifies the range of RTP port to be used for audio stream 16384 32768 G 729 G 729A 10 20 30 G 729AB 10 6 723 1 30 60 2 Packet length ms This specifies the length in time of each packet However the packet length is codec 20 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 supported ms 6 711 a law praw 0125 29 GoIP User Manual 3 Jitter Buffer A jitter buffer is designed to remove the effects of jitter from the decoded voice stream Fixed 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 f
65. nel is configured as the server in HUNT Group mode 1 Unconditional Call Forward Incoming calls are always forwarded to an idle channel If all Client channels are in use the Host channel answers an incoming call 2 Call Forward Busy The Host channel only forward in incoming call when the line is busy 3 3 11 SIM This section contains a set of parameters are related to the SIM Card of the channel selected The Total Talk Time Limit and the Talk Time Limit Call can used to limit the phone usage of the SIM selected It is useful for customers who are using a sim card that has a higher calling rates when a certain limit is exceeded or at different time periods Please program this section as required The Auto Config Other Line button is provide to facilitate the programming of each channel After the parameters for the CH1 SIM are set clicking this button automatically duplicates the same settings to all other channels with the exception that the SMS Alert ID is incremented by 1 with respect to the ID of the previous channel SIM GPRS Registration Enable Disable SIP Registration when Talk Time Limit expires Enable Disable cm OcH2 O cH3 OcHa OcH5 OcH6 OCH O che GSM Number Required by the Hunt Group Mode IMEI 355073036020376 Unlock PIN Ooo E Unlock PIN2 Talk Time Limit m Drop Call when Talk Time Limit expires Enable Disable Talk Time Limit m Call ee Billing Increment 5 60 SMS
66. ng 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 acomplete 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 GoIP dial the number 108 Ifthe first rule does not match it will continue to the second rule There is no matching number for the second rule Itis 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 Enable GoIP User Manual When there is no match the incoming GSM call is handled as if the Forward Number is blank 4 Dial Plan 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
67. 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 3 4 2 Change Password Click Change Password to change the password with respect to the login level There are three login levels 1 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 and the default password is 1234 Note Administration Level allows changing the passwords for all 3 levels User Level Administration Level SMS Level 3 4 3 Send USSD Click Send USSD to access the webpage as shown below to send USSD commands Send USSD Line1 Line Oline3 Olline4 OLine5 OlLine6 OLine7 Lines A Lines Line 1 GSM Status LOGIN Line 165M Number 48 GoIP User Manual 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 The last option All Lines means that all channels are selected and the same USSD command is sent via all channels provided that they are in the Login Status b Enter the USSD command c Click Send Example For the service provider P
68. 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 0192 168 2 237 5060 gt Max Forwards 30 User Agent HYBERTONE 55 GoIP User Manual Allow INVITE ACK BYE CANCEL OPTIONS NOTIFY REFER REGISTER MESSAGE INFO SUBSCRIBE Content Type application sdp Content Length 226 2 Forward Function This mode supports SMS forwarding from GSM to SIP and from SIP to GSM a b Received GSM SMS messages are forwarded to both SIP and GSM depending on the settings of the SMS Forward Number and SMS Forward GSM Number SMS plode Relay mdr len ggdg Number a EEEEEEEEE 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 disa
69. o over 10 seconds for a ringback tone is returned from the GSM network the caller may hear a long silent period This setting defines how incoming VoIP calls are authenticated when the device is 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 following 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 2 IP only calls received from the registered SIP Server s are accepted 3 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 A simple SIP Proxy is embedded in the device Choose Enable to activate this SIP Disabled 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 Alive sends a NULL packet to the router regu
70. on 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 dial a 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 not equal to the SIP Number defined for this line GolP dials out the Caller Number according to the Dial Plan defined If this parameter is specified GolP dials this phone number via the corresponding GSM channel whenever an incoming VolP 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 dial a phone number Please note that this could only happen in the Single Server mode Config by Line
71. on DHCP client amp Server QoS VLAN VPN PPTP Online firmware upgrade Remote Control Mechanism for remote technical support Proprietary Auto Provisioning Mechanism Remote SIM function Short Messages SMS support standalone and server based Call Management and Routing GoIP User Manual GoIP 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 Goz SLIT GolP 4 4 Channel a GolP 8 8 Channel print GoIP User Manual AC DC Power Adapter GolP1 12V 500mA GolP4 12V 2A GolP8 12V 3A GolP16 12V 4A GolP32 12V 4 5A 1 x Ethernet CAT5 Cable 2M 1 6 LED Indicators POWER RUN LAN Channel 4 Channel 2 Channel 4 M oo o 0 mm E E iD qe lt oc O O They are often used 0 Ta to T D oO E c E a co G a O O O LED indicators shown above for GolP 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 Th
72. ottom you need to open the bottom SIM cover 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 For GoIP 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 shown in the figure on the right The metal contacts must face up and the cut corner is inserted first E ae YUU MS me EL El EL El ft G4 E El O y WY OP O DW WF WF ee ee ee ee eee eee eee GSM8 GSM 1 GoIP User Manual 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 EY en Y To Intranet Internet Power 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 di
73. red yet SMS SMS Server registration status Y means Registered and N means Not Registered RSSI This indicates the Received Signal Strength Indicator of the current cell It ranges from 0 to 31 which represents a signal level ranging from 113 dBm to 51 dBm each increment in rssi values means 2 dBm increment 99 means that the signal level is unknown or undetected Carrier This shows the name of the current GSM carrier Cell ID This shows the Base Transceiver station BTS ID Idle This shows the time elapsed since the last call Remain This shows the time remaining if the Total Talk Time Limit m is set Once the Remain time reaches zero the corresponding channel is locked and its VoIP registration is also suspended default setting However there is an option in Section 3 3 10 to enable SIP registration even when the Talk Time Limit expires Remain 0 Reset Click this button to reset the Remain Timer to the Total Talk Time Clicking on the Reset button located at the bottom the row that is labeled All resets all Remain Timers of all channels General The General page covers basic information on the hardware network and call status and setting These information are useful for debugging the device operation and status 1 Hardware SIN test2 Firmware 65 4 01 81 3 Model GolPx8 5010 Local Time 2013 12 23 11 28 06 Network LAN Port 192 1688 2 250 LAN MAC PC Port 192 185 58 1 PPPoE DISABLED Gatewa
74. rver pool ntp org C Enable E Disable Auto prowision Remote Control lt lt Remote Control Remote Server Remote Server Port 1820 Remote Server ID Remote Server Key 118 142451 162 Network Tones HTTP Port DDNS DONS Address DDNS Port Update Interval Auto Reboot Reboot Time IVR Remote SIM Server ID Password Net protocol SMPP SMSC ID Password Port DTMF Detect Min Gap 200 400 GoIP User Manual 8 Enable Disable volpddns net 339800 Enable Disable 4 00 e Enable O Disable Enable Disable 192 168 2 1 O UDP TCP Enable Disable il 15 GolIP User Manual The preference page shown above consists of the following system level parameters and options as shown in the table below Parameter Description Preference 1 Language This sets the webpage and voice prompts language Currently only English and Simplified Chinese Mandarin for voice prompt are supported 2 Time Zone This specifies the offset of the local time zone with respect to GMT The syntax m should be GMT x where x is the offset 3 Time Server This specifies IP address or the domain name of a network time server for computer clock synchronization The default is pool ntp org 4 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
75. s mode uses the default GSM base selection mechanism Poll This mode limits the number and the list of base stations BTS that could be selected Either Whitelist or Blacklist method can be used for BTS selection GSM Base Station CH1 OcH2 OcH3 OcH4 OCHS OCH6 OCH7T OCHS Base Station Selection Maximum Polling ES Channels Channel Switching Interval m White Black List Auth White Black Lists Available Cell IDs Save Changes Parameter Description Default Value GSM Base Station 1 2 3 4 Maximum Polling This limits the maximum number of channels for polling Channels Channel Switching This defines the duration in minutes when the next base station switching Interval m should occur The base station switching only occurs when the corresponding channel becomes idle Whitelist Blackllist Whitelist defines the base stations that are going to be used Blacklist defines the base stations that are NOT going to be used White Black List Auth There are three ways to define how to switch to a different base station If it is disabled the base station selection cycles through the neighbor list If the Whitelist authentication mode is selected then the channel selection cycles through the base stations defined in the Whitelist If the Blacklist authentication mode is selected then the channels listed in the Blacklist will not be selected Fixed This mode locks the base station to the Cell ID sp
76. s parameter enables the support of SMPP protocol Please note that GolP is acting as a SMSC Short Message Service Center Fill in the SMPP ID Password and port number for SMPP data communications This parameter specifies the maximum dropout time for a DTMF tone When making a call from SIP to GSM or Drapout from GSM to SIP by using the second ee dial method the device needs to detect the dialing digits from the DTMF tones DTMF tone for one digit 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 GoIP User Manual voipddns net 39800 120 mins 17 GoIP User Manual 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 en
77. s that are related to SMS The top parameter SMS to VoIP is a general setting for all channels This defines how received SMS messages are used for Call Back or Forward functions which are described in details in the table below Default Value 1 SMS to VoIP This defines how the device handles received SMS messages Disabled 5M5 to VolP Disable Call Function Forward Function Call Function 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 Please note that the SIP server registered must be configured for this operation SMS to VoIP Call Mode Dialing Prefix gt Call Mode This parameter specify which SMS Dial mode is used Please refer to Appendix B for more information on the modes available gt Dialing Prefix The parameter is applicable for SMS Dial mode It allows a prefix to be added to the phone number of the called party Forward Function This mode forwards incoming GSM SMS to a SIP terminal and a GSM number 5M5 to VolP To VolP Number To GSM Number gt VoIP Number Incoming GSM messages SMS received are forwarded to the SIP Phone Number specified in this parameter gt GSM Number Incoming GSM messages SMS received are forwarded to the GSM Phone Number specified in this parameter The channel received the message is used to forward the message
78. tax lt Prefix1 gt lt Prefix2 gt lt Prefix3 gt where Prefix is a text string which consists of digits alphabets and special characters The maximum length of the Routing Prefix is 120 characters 2 Config By Line for all models except GolP 1 Mode This mode is only applicable for multi line models Each line associated with a corresponding GSM channel registers to a SIP server separately and operates as an independent phone line A Routing Prefix for each channel must be assigned in order to enable the channel to allow making outgoing calls This allows to the SIP server to assign which channel to dial out the call for termination If a channel does not have its Routing Prefix set this channel will not be used to dial out any calls In this mode the prefix of the phone number to be dialed out must match one of the Routing Prefixes assigned The channel with the matching Routing Prefix will be used to dial out the call If no match is found the call will not be dialed out and a SIP 404 message is returned to the SIP Server If a match is found but no channel is available to dial out the call a SIP 503 message is returned to the SIP Server The syntax for the Routing Prefix is defined in the Parameter Table for Single Server Mode It is important to note that the Routing Prefix P must be removed via the dial plan before the number is dialed out The dial plan syntax to remove the Routing Prefix is P P Please refer to section
79. th SIP server and the device are using public IPs for reliable operation Installing either one or both SIP server and the device behind NAT may or may not work properly depending on 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 Otherwise VoIP calls may fail to establish properly in this network environment SIP Config Mode SIP Trunk Gateway DO SIP Trunk Gata gt Phone Number Re register Period s a Authentication 1D DO Routing Prefix Line Lines os Ps FNs Pa 8 Linel Line Fh oa ph oa aie om ia lol Line5 L_ LineS line 4 Lined Line 1 Routing 7 Prefix Auto Config Other lines The device accepts calls from up to 3 IP addresses SIP Trunk Gateway1 SIP Trunk Gateway2 SIP Trunk 23 GoIP User Manual 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 the whole segment IP addresses 0 255 Calls originated from the IP segment 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 Call routing to a GSM channel is now based on the Routing Prefix of each GSM channel Line x The channel s
80. tis very important that you send us your application requirements in full details when seeking for technical support in configuring the device In order to simplify SIP configuration SIP settings are categorized as Basic VoIP Advanced VoIP and Media In general Basic VoIP defines how the GolP handle SIP calls and four SIP modes are supported It is important to understand the differences between each mode in order to select a mode that is the most suitable for your application Depending on the SIP environment and network conditions you may or may not need to change the default settings in the Advanced VoIP and Media pages Once SIP settings are completed it is important to configure the device for making outgoing calls and receiving incoming calls Please see section 3 3 6 and 3 3 7 for more information on Call OUT and Call IN settings 19 GoIP User Manual The four modes of SIP operations are described below Config Mode Trunk Gateway Mode Single server Mode Contig by Line Config by Group Trunk Gateway Mode 1 Single Server Mode 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 in the SIP server to match the number of lines available in the device Call routing to a GSM channel is now based on the Routing Prefix of each GSM channel Here is the channel selection algorithm a A
81. ttings of the SIM card assigned to the corresponding channel There are 3 possible status A ON This means that the corresponding Call Forward mode is enabled and this setting is sent to the GSM network when a new GSM registration takes place OFF This means that the corresponding Call Forward mode is disabled and this setting is sent to the GSM network when a new GSM registration takes place Not Set This means that there is no change to the current Call Forwarding mode and nothing is sent to the GSM network when a new GSM registration takes place This is useful by leaving the current Call Forward mode unchanged CH Always Busy No nme ai i No Service 1 Not Set Mot Set Not Set Mot Set 2 Not Set Not Set Not Set Not Set 3 Not Set Mot Set Hot Set Not Set a Not Set Not Set Mot Set Mot Set 5 Not Set Not Set Not Set Mot Set 6 Mot Set Mot Set Not Set Not Set T Not Set Not Set Not Set Mot Set a Mot Set Mot Set Mot Set Mot Set 14 3 3 Configuration Click Configuration on the left hand column to display the Configuration page and the following submenu 1 Preference 2 Network 3 Basic VoIP 4 Advance VoIP 5 Media 6 Call Out 7 Call Out Auth 8 Call In 9 Call In Auth 10 SIM 11 SIM Forward 12 IMEI 13 SMS 14 GSM Carrier 15 GSM Base Station 3 3 1 Preference Preferences Language HE Engl ish Y Time Zone Time Se
82. uick 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 NM ON Re ROR ON RGR ANS NR Ne NR EK 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 VolP Blockings 1 3 Hardware Features SS Re RS AR SN ARM processor DSP for voice signal processing Two 10 100MB Ethernet ports IEEE 802 3 standard with status LEDs Quadband GSM module 850M 900M 1800M and 1900M External Antenna Internal Antenna option for selected models 1 4 Software Features SL A Re AK A AR A R A SK LINUX OS Built in Web Server for device configuration Built in SIP Proxy Simplified PPPoE Dial Up Router functi
83. vironment 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 upstream and downstream 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 There are 3 access methods available to configure the LAN port LAN Port Static IP E IP Address Po 1 Static IP This mode applies to both Subnet 27 public and private IP network Mask optional environment In the LAN port Default Route 0 configuration shown on the left select Primary DNS a Static IP and then fill in the parameters Secondary OY as provided by your network DNS optional administrator 2 DHCP default setting When the device is installed behind LAN Port DHCP we NAT and a DHCP host is available select DHCP to enable the device to obtain LAN IP address and other network information automatically 3 PPPoE ADSL and Cable modems very often use PPPoE dial LAN Port PPPOE sd we up to obtain network IPs If this is the case
84. y 192 168 2 253 DHS Server 2 8 8 8 Call Management VolP ah Line Mode Login Routing Prefix EN Login Call In Call Out Remain Time ER On fo ron or oro NO LIMIT Hardware mM oF Gt oo Z Z Z Z Z Z lt lt lt lt lt g on z Z on z 11 GoIP User Manual 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 d Local Time This shows the current system time Itis a good indication for normal network access provided that the network server address and time zone are set properly Please make sure that these information are provided when reporting a problem or requesting for technical support Network a LAN Port This field shows the IP address assigned to the LAN Port b LAN MAC This field shows the physical hardware address MAC assigned to the LAN port c PC Port This field shows the IP address assigned to the PC Port d PPPoE This field shows the PPPoE dial up status Itis only meaningful when PPPoE is enabled e Gateway This field shows the default gateway IP assigned for data traffic routing f DNS Server This field shows the current DNS server assigned 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 solv

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