voptech GoIP User Manual
  1. Manuals
  2. Brands
  3. voptech Manuals
  4. Gateway
  5. GoIP
  6. User manual

voptech GoIP User Manual

Voip gsm gateways
Hide thumbs
Table of Contents

Advertisement

Quick Links

Download this manual
GoIP User Manual
VoIP GSM Gateways
Models:
GoIP
GoIP-4/4i
GoIP-8/8i, WoIP-8
GoIP-16
GoIP-32
Revision: 1.5
2016/6/24
1

Advertisement

Table of Contents
loading
Need help?

Need help?

Do you have a question about the GoIP and is the answer not in the manual?

Questions and answers

Summary of Contents for voptech GoIP

  • Page 1 GoIP User Manual VoIP GSM Gateways Models: GoIP GoIP-4/4i GoIP-8/8i, WoIP-8 GoIP-16 GoIP-32 Revision: 1.5 2016/6/24...

  • Page 2: Table Of Contents

    Content Summary................................4 General................................4 Special Notes..............................4 Introduction............................4 Protocols..............................5 Hardware Features..........................6 Software Features..........................6 Package Content.............................6 LED Indicators............................8 Installation..............................9 Configuration..............................12 HTTP WEB Server Login........................12 Status..............................13 5.2.1 Summary...............................13 5.2.2 General..............................15 5.2.3 GSM................................16 5.2.4 SIM Call Forward..........................17 Configuration............................19 5.3.1 Preference.............................19 5.3.3...
  • Page 3 5.4.11 Backup / Restore..........................61 5.4.12 Reset..............................62 5.4.13 Reboot..............................62 Appendix A. Special SMS Commands.......................63 Appendix B. SMS To VoIP..........................64 Appendix C. Custom Network Tones....................... 68 Appendix D. GSM Group Mode......................... 69 Appendix E. CID Call Forward.......................... 70 Appendix F. Volume Adjustment........................71...

  • Page 4: Summary

    GSM/WCDMA phones are getting more and more popular all over the world with lower and lower service charges, the emergence of GoIP and WoIP bridge the gap between the traditional telephone networks and VoIP networks as shown in the diagram below.

  • Page 5: Protocols

    The idea is to route all your incoming GSM calls to a GoIP / WoIP via call forward or simply insert your SIM card to a GoIP / WoIP. You can then setup the GoIP / WoIP to forward all incoming calls to another GSM number in the world via a VoIP service provider.

  • Page 6: Hardware Features

    Dynamic Host Configuration Protocol (DHCP)  Domain Name System (DNS)  User Account Authentication (via MD5)  Proprietary Relay Protocol (Avoiding VoIP Blockings)  3.3 Hardware Features ARM processor  DSP for voice signal processing  Two 10/100MB Ethernet ports (IEEE 802.3 standard) with status LEDs ...
  • Page 7 Item Appearance Description 1 x Main Unit GoIP (1-Channel) GoIP-4 (4-Channel) GoIP-8 (8-Channel) WoIP-8 (8-Channel) GoIP-16 (16-Channel) GoIP-32 (32-Channel) AC/DC Power Adapter:  GoIP1: 12V/500mA...

  • Page 8: Led Indicators

    1 x Ethernet CAT5 Cable (2M) 3.6 LED Indicators LED indicators (shown above for GoIP-8) are used to show the current status of the device. They are often used to determine if the device is working normally or not. Description...

  • Page 9: Installation

    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.
  • Page 10 For GoIP-16, the SIM card insertion orientation is shown in the figure below. The cut corner is pointing downward with the metal contacts facing the front of the GoIP. For GoIP-32, the SIM card insertion orientation is shown in the figure below.
  • Page 11 2. The RJ-45 port, labelled “LAN”, is intended for intranet or internet connection. Depending on your network environment, it can be connected various type of network equipment, such as network router, network switch / Hub, xDSL/Cable modem, etc. 3. The RJ-45 port, labelled “PC”, is intended for network sharing and it supports both bridge and router modes.

  • Page 12: Configuration

    5 Configuration The device can be configured via its built-in http web server or via an Auto Provision Server. Auto Provision Server is a free utility supporting both Window and Linux OS. This utility is developed for the sole purpose of automating the configuration of our products.

  • Page 13: Status

    5.2.1 Summary The current VoIP and GSM statuses are listed in the Summary page as shown below (extracted from GoIP-4) . They are very essential to display the operation status of the device in order to determine if it is working...
  • Page 14 DIALING <phone number> – This occurs when the GoIP 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.

  • Page 15: General

    is the icon for resetting talking time remaining. When the accumulative talk time  reaches the Talk Time Limit, the remaining talk time reaches zero. is the icon for resetting the SMS count to zero.  is the icon for resetting the ACD and ASR to zero. ...

  • Page 16: Gsm

    Current Client IP – This shows the IP address of the PC for the current web access. Call Management section summarizes the both GoIP 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.

  • Page 17: Sim Call Forward

    The top table shows a number of GSM parameters which are useful to determine if the GSM channels in the gateway are working properly. Remote SIM - This tells if Remote SIM function is used or not. "DISABLE" means using the local SIM cards that are inserted to the device.
  • Page 18 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.

  • Page 19: Configuration

    5.3 Configuration Click “Configuration” on the left hand column to display the Configuration page. Under this section, there are 16 items in the submenu as shown below. 5.3.1 Preference...
  • Page 20 The preference page shown above consists of the following system level parameters and options as shown in the table below. Default Value Parameter Description (Preference Page) English 1. Language This sets the webpage and voice prompts language. Currently, only English and Simplified Chinese (Mandarin for voice prompt) are supported.
  • Page 21 Enabling this feature allows the SIM Cards to be installed in a SIM Bank rather than in the on-board SIM slots. GoIP can either register to a SIM Bank or a SIM Server. Please refer to the SIM Bank User Manual for more information.

  • Page 22: Network

    Please try the Short Command format if you have troubles to get the Remote SIM to work properly. 12. SMPP SMSC This parameter enables the GoIP to act as a SMPP SMSC (Short Message Service Center). The SMPP Client must use the ID, Password, and the Port number specified.
  • Page 23 environment to be connected. LAN Port There are 3 access methods available to configure the LAN port. 1. Static IP – This mode applies to both public and private IP network environment. In the LAN port configuration shown on the left, select “Static IP” and then fill in the parameters as provided by your network administrator.

  • Page 24: Basic Voip

    As more features are added, SIP and H.323 VoIP protocols are supported in two different firmware versions. Except GoIP-1, all other models are now shipped with the SIP protocol firmware as a factory default. H.323 protocol is required, the firmware of the device can be changed to the one that supports H.323 protocol.
  • Page 25 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.
  • Page 27 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 GoIP-1) Mode This mode is only applicable for multi-line models.
  • Page 28 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. 3. Config. By Group (for all models except GoIP-1) This mode is basically a combination of Single Server mode and Config. By Line mode.
  • Page 29 SIP server is required to support NAT. Both routers should also be setup to map the signaling port and media ports to the SIP Server and the GoIP properly. Otherwise, VoIP calls may fail to establish properly in this network environment.
  • Page 30 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). channel selection 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.

  • Page 31: Advanced Voip

    5.3.4 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. The table below summaries all the parameters defined in this section. Parameter Description Default Value...
  • Page 32 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. 3.
  • Page 33 Signaling NAT Traversal This setting is not required if the target SIP server / PBX supports NAT traversal. None However, if your ISP blocks VoIP traffics, you could try to use Relay Proxy setting. Depending on how your ISP blocks VoIP traffics, the Relay Server method may or may not work in your network environment.

  • Page 34: Media

    5.3.5 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.
  • Page 35 Parameters Description Default Value (Media) RTP Port Range This specifies the range of RTP port to be used for audio stream. 16384 - 32768 Packet length (ms) This specifies the length (in time) of each packet. However, the packet length is codec dependent as well.

  • Page 36: Call Out

    Audio Codec Preference Six types of audio codec are supported and they are summarized in the table below. All codecs are enabled in the Codec Raw Data Ethernet 802.3 order Bandwidth (bps) Data Bandwidth preference (bps) shown below. 1. a-law ~ 85K a-law -law...
  • Page 37 The parameters available in this section are described in details in the table below. Parameter Description Default Value GSM Auto Redial This is a general parameter for all channels. It enables a call to be redialed automatically in Enabled x seconds after the last attempt fails. GSM Dialing Timeout (s) This is a general parameter for all channels.
  • Page 38 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. Dial Plan This parameter is used to define rules on how to handle a numbers received for dialing out via a GSM channel.

  • Page 39: Call Out Auth

    Example: Dial Plan: 8613[5-9]XXXXXXXX:-86+0| This rule performs the following actions: 1. Phone number must be 13-digit long. 2. Phone number must start with 86135, 86136, 86137, 86138, 86139. 3. The prefix 86 is removed. 4. A digit 0 is added. 5.

  • Page 40: Call In

    4. Forward incoming GSM calls (to the host channel) to other idle GSM channels. This enables a multi-channel GoIP and/or multiple GoIPs to simulate the “Calll Center” function. Therefore, only the phone number of the host channel is released for the public to call in. In fact, multiple lines are available to serve incoming calls.
  • Page 41 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 in the Remote Party. Use CID as SIP Caller ID –...
  • Page 42 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 GoIP which contains the Host channel of the other Hunt Group. Forward Mode ...

  • Page 43: Call In Auth

    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. Auto Incoming Call This option is used to block an incoming calls when the condition specified in the Trigger is met.

  • Page 44: Sim

    “Blacklist”. This setting specifies the Call Forward method when the GSM channel is configured as the server in HUNT Group mode. 1. Unconditional Call Forward – Incoming calls are always forwarded to an idle channel. If all Client channels are in use, the Host channel answers an incoming call. 2.
  • Page 45 6. Unlock PIN2 SIM Card unlock PIN code 2 7. Talk Time Limit (m) There are 3 ways to define the Talk Time Limit for a SIM card. 1. Total talk time allowed without any time limit When the accumulative talk time of a channel reaches this limit, the corresponding SIM card is disabled.

  • Page 46: Sim Forward

    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.

  • Page 47: Imei

    4. Call Forward Forward calls when the GSM channel cannot register to the carrier. Not Set Unreachable This specifies the phone number to receive forwarded calls under this  Forward Num condition. 5.3.12 IMEI This section allows IMEI modifications. Each GSM channel is programmed with an IMEI. It is a 15-digit number with the format described below.

  • Page 48: Sms

    The IMEI of each channel can be programmed manually. An exact 15-digit number must be entered; otherwise, the number entered is ignored and the old IMEI will still be used. The device calculates the check digit and then replace the last digit. However, the device does not check for the validity of the TAC. The IMEI Auto Change option enable the device to change only the “CCCCCC”...
  • Page 49 Parameter Description Default Value This parameter applies to ALL channels. SMS to VoIP Disabled This defines how the device handles received SMS messages. 1. 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.
  • Page 50 Once the SMS Server is installed and in operation. The following parameters are require to enable the GoIP to register to the SMS Server. Please note that each channel must be programmed individually in order to register to the SMS Server.

  • Page 51: Gsm Carrier

    GSM service provider based on the default preference set by the SIM card. When a GoIP 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.
  • Page 52 Three modes for base station selection are available: 1. Auto - This mode uses the default GSM base station selection mechanism. Use this default setting if you don’t have any preference. 2. Poll - This mode enables the device to switch to a different base station at the interval specified. Base Station Polling List is the list of base stations that can be used.

  • Page 53: Event Trigger

    4. 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. 3. Fixed - This mode locks the base station to the Cell ID specified. First Press to get the current list of base stations detected.
  • Page 54 Defined Actions: 1. Null – No action. 2. Disable VoIP connection – This causes the GoIP to deregister from the SIP Server for the corresponding line. 3. Disable Call Out via GSM – This prevents calls from dialing out from the corresponding GSM channel.

  • Page 55: Tools

    5.4 Tools Click “Tools” on the left hand menu to access the submenu as shown below. Please note the available options under the Tools menu. 5.4.1 Online Upgrade Click [Online Upgrade] to upgrade the device firmware. The current version is displayed as well as the last upgrade time.

  • Page 56: Send Ussd

    5.4.3 Send USSD Click [Send USSD] to access the webpage (as shown below) to send USSD commands. 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.

  • Page 57: Send Sms

    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. 5.4.4 Send SMS Click [Send SMS] to access the Send SMS webpage as shown below. The procedures to send a SMS are: Select the Line (GSM channel) that you want to send a SMS.

  • Page 58: Sms Outbox

    5.4.8 Ping Test GoIP can use the Ping to test the reachability of the domain name or the IP address specified. measures the round trip time of messages from transmission to reception and reports errors and packet...

  • Page 59: Dial Test

    Ping Test. The test results are then displayed in the window below. 5.4.9 Dial Test The Dial Test enables the GoIP to dial out a call manually via the line or channel selected. 1. Select the Line that you want to make an outgoing call.
  • Page 60 However, the CID of the SMS Response must be set so that the GoIP can check the SMS from this CID. Set the Text before the SIM phone number so that the GoIP can extract the phone number from the SMS.

  • Page 61: Backup / Restore

    4. Set SMS Service Number = the phone number of another GoIP channel. 5. Set SMS Content = Trigger String for an SMS Reply 6. Set CID of the SMS Response = the phone number of another GoIP Channel (same as the one set in SMS Service Number).

  • Page 62: Reset

    5.4.12 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. Click [OK] to reset the device configuration back to the factory default! 5.4.13 Reboot Click [Reboot] to restart the device.

  • Page 63: Appendix A. Special Sms Commands

    Appendix A. Special SMS Commands In order to manage the device, special SMS commands can be sent to anyone of the GSM channel in order to read the LAN IP, reset the device and reboot the device. The table below summarizes the SMS command syntax.

  • Page 64: Appendix B. Sms To Voip

    From: <sip:8613800000000@192.168.2.1:5060>;user=phone;tag=65248630 To: <sip:8675588228822@192.168.2.1> Call-ID: 117025903@192.168.2.237 CSeq: 2 INVITE Contact: <sip: 8613800000000@192.168.2.237:5060> Max-Forwards: 30 User-Agent: Voptech 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.
  • Page 65 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: <sip:20001@192.168.2.1:5060>;user=phone;tag=65248630 To: <sip:8675588228822*8613902994477@192.168.2.1> Call-ID: 117025903@192.168.2.237 CSeq: 2 INVITE Contact: <sip:20001@192.168.2.237:5060> Max-Forwards: 30 User-Agent:Voptech Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REFER, REGISTER, MESSAGE, INFO, SUBSCRIBE Content-Type: application/sdp...
  • Page 66 From: <sip:20001@192.168.2.1>;tag=667435795 To: <sip:3999@192.168.2.1> Call-ID: 2094144847@192.168.2.162 CSeq: 4 MESSAGE Contact: <sip:20001@192.168.2.162:5060> Max-Forwards: 30 User-Agent: Voptech Content-Type: text/plain Content-Length: 8613682626865 075583185700 Please note that the SIP Server side must be programmed to process this SIP MESSAGE according to the application needed.
  • Page 67 Example: A SIP SMS is sent from the SIP number 3999 to the SIP number 2001 (used by the device) and then the SMS is sent out to the phone number 1368266800 via the GSM channel associated with 2001. SIP SMS Sender = 3999 SIP SMS Recipient = 2001 SIP SMS Content = 13682626800 Hello world...

  • Page 68: Appendix C. Custom Network Tones

    Appendix C. Custom Network Tones This section describes how to define custom network tones. The “Custom” selection allows the following tones to be defined as shown on the right. 1. Dial Tone – When an incoming call is answered, this tone is generated to indicate to the caller to dial a number.

  • Page 69: Appendix D. Gsm Group Mode

    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.

  • Page 70: Appendix E. Cid Call Forward

    Appendix E. CID Call Forward 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.

  • Page 71: Appendix F. Volume Adjustment

    Appendix F. Volume Adjustment The volume adjustment of the device can be accessed via the URL below. http://<device address>/en_US/gaim.html The <device address> 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.

This manual is also suitable for:

Goip-4Goip-4iGoip-8Goip-8iWoip-8Goip-16 ... Show all

Table of Contents