Grandstream Networks GSM gateway User's Manual
Contents
1. 10 95 Humidit umidity non condensing Compliance FCC CE C Tick Februaray 2006 Page 9 42 MasterVolP VolP to GSM Gateway 3 1 3 Basic Operations 3 1 3 1 Getting Familiar with the Key Pad and the Voice Prompt VoIP Client ATA has a stored voice prompt menu for quick browsing and simple configuration To enter this voice prompt menu simple pick up the phone and press the button on the VoIP Client ATA or pick up the phone and dial The following table shows how to use the voice prompt menu to configure the device for required voice prompts Menu Voice Prompt User s Options Main Menu Enter a Menu Option Enter to next option and back to main menu or Dial 01 06 47 86 or 99 Menu option Static IP Mode or Dial 9 to toggle the selection Dynamic IP Mode If user selects Static IP Mode user will need to configure the all IP address information through menu 02 to 05 If user selects Dynamic IP Mode the device will retrieve all IP address information from DHCP server automatically when user reboots the device O30 a 05 S IP Address IP address The current WAN IP address is announced Enter 12 digit new IP address if in Static IP Mode Subnet IP address Same as Menu option 02 Gateway IP address Same as Menu option 02 DNS Server IP address Same as Menu option 02 TFTP Server IP address Same as Menu option 02 TFTP2. The Dual Cell to BRI Gateway product you have purchased is under warranty of 12 months from the date of purchase by the original purchaser In case of defects of materials or workmanship Eurotech Communication will replace it free of charge This warranty applies to hardware software but does not include SIM Cards This warranty will not be honoured if the device has been mishandled in any way We hope you enjoy our product and we will be happy to receive any comments you may have This will enable us to improve our products and the Technical Support that we give to every customer TABLE OF CONTENTS OWN 3 2 3 3 3 4 Getting Started ini A An 5 Check Your package tems iii 7 VOIP Cent ATA osado lod dd l tt dlrs 8 O 0S ce silencer atti anwiaiie de daca nase aot ahat gue eae eee aie Gee aide oan dalenee i antaeelosneesstaeteeetetoue se 8 Salil SA A A AN 8 Ove Hardware SOeCCIIC MIN e de 9 Oi gn ASIC Opera los lt it 10 3 1 3 1 Getting Familiar with the Key Pad and the Voice ProMpt ococcocccococccocococococononocononononononononononcnorancnnrancnnnns 10 Owe PIAGING Phone Calls ist niecli s 11 3 1441 Gallin phone Or extension number iia A beis 11 Salida Dec calco O ETE 11 ShA BINO Trans O OO A Par o O A 12 A Attended A a ao 12 A PP e O O 13 EN Fax OUDON PP PP O een eee 13 Ll ED LIGNE Patter inGle all 0 Mis costra 14 Comouranonm Gude meee rt ne ee o o to o o oa eve O 15 3 2 1 Configuring VOIP Client with a
3. VolPiViaster Version 4 x VoIP to GSM gateway Connecting Cellular Phones directly to Voice over IP worldwide networks User Manual MasterVolP VolP to GSM Gateway Usage Warnings 1 High voltage transients surges and other power irregularities can cause extensive damage It is the user s responsibility to provide a power protection system 2 It is the user s responsibility to install operate and maintain the system in accordance with all applicable codes regulations and safety measures Trademark and Patents All trademarks patents and copyrights apply General Manual Notes Without notice and without obligation the contents of this manual may be revised to incorporate changes and improvements Every effort has been made to ensure that the information in this manual is most complete and accurate while writing this time of publication Nevertheless Mega 2000 AS cannot be held responsible for errors or commissions Februaray 2006 Page 3 42 MasterVolP VolP to GSM Gateway Dear Customer We thank you for purchasing our VoIP Master VoIP to GSM Gateway The information in this manual does not constitute a warranty of performance although the information has been compiled and checked for accuracy by Eurotech Communication Ltd All our products are developed and produced by experienced engineers who aspire to achieve customer satisfaction utility value and reliability of products Warranty Policy
4. amp BRA CANNER File Edt Wew Favorites Took Help p A O gt Je ech ES Folders THE Address DAPR amp BRI amp WINNER Irmtalihied Softwere Corpora 2 JU 3 Floppy A E es Local Disk C E keJ My Dise D LA a ES _DISK IMAGE VoIP Master 2 E Took E Se Local Deck Es Install wolf Master MN Setup exe in the software disk Al InstallShield Software Corpora In Windows Explorer navigate to Icon Double click the Icon wait till the installation window will open VolP Master Installshield Wizard Welcome to the InstallShield Wizard lor VolP Master The InstallShield wizard wil install talk n talk on youl computer To continue click Next installato Februaray 2006 Page 35 42 MasterVolP VolP to GSM Gateway 4 Click Next The Setup Type window opens VolP Master InstallShield Wizard x Setup Type Select the sstuo Wos lo install Please select a setup type Y Complete sF Al program features wil be installed Requires the most disk space ls J pa PA A La 5 AS yy Select which progiam features you want nstaled Recommended fol advanced LESTE Installsiald 5 Select Complete and click Next The Begin Installation window opens E v2 5 1 InstallShield Wizard ES Ready to Install the Program The wead ts ready to begin nstalation Dick Install to begin the installation lf
5. DSL modem The HT 286 will attempt to establish a PPPoE session if any of the PPPoE fields is set In this mode the WAN side web access is disabled and TFTP upgrade for firmware is not feasible and HTTP upgrade is the only available solution If Static IP mode is selected then the IP address Subnet Mask Default Router IP address DNS Server 1 primary DNS Server 2 secondary fields will need to be configured These fields are reset to zero by default Time Zone This parameter controls how date time will be displayed according to the specified time zone Daylight Savings Time This parameter controls whether the displayed time will be daylight savings time or not If set to Yes then the displayed time will be 1 hour ahead of normal time In addition to the Basic Settings configuration page the end user also has access to the device Status page The following is a screen shot of the device Status page Februaray 2006 Page 16 42 MasterVolP VolP to GSM Gateway Februaray 2006 Here are the status details shown MAC Address WAN IP Address Product Model Software Version System Up Time Registered PPPoE Link Up NAT NAT Mapped IP NAT Mapped Port Total Inbound Calls Total Outbound Calls Total Missed Calls Total Call Time in minutes Total SIP Message Sent Total SIP Message Total RTP Packet Sent Total RTP Packet Received Total RTP Packet Loss Page 17 42 00 0B 52 01 56 4D 192 168 1 12
6. GSM Gateway 3 4 5 Port and SIM Settings This chapter details port to SIM association as well as required and optional settings 3 4 5 1 Dial Settings for the GSM Port After defining the Comport press GSM Port in the left pane The Port Setting window is displayed AMIGO ON VOIP 2 3 1 Es Disconnect Refresh Exit Dial pause 10 50 0 041mg E sim 2 A Follow me Repeat access to VOIP OFF A Callback Debug Rx volume TA volume Write settings El Ya abrag tor Data e O Define dial settings in this window as follows 1 In the Dial pause box set the time interval whereupon a dialed number is dispatched after the designated delay time Each unit is 0 1 second For example if you want the number to be dispatched 3 seconds after you finish dialing enter 30 in this box 2 Upon completion of a call if you want to remain connected to the GSM Network set Repeat Access to VoIP to On 3 Set Receiving Rx and transmitting volumes in the Rx receiving and Transmitting Tx volume settings 4 Press Write Settings Februaray 2006 Page 39 42 MasterVolP VolP to GSM Gateway 3 4 5 2 SIM Settings After making the port settings press a SIM icon in the left pane The SIM Setting window opens AMIGO ON VOIP V 3 1 Refresh Exit Gateway Version SIM 1 Setting Gateway setting DE GSM Port Ja Skye A A A eS BB Follow me EA Callback E Debug Network PIN Code Write setings g
7. HT286 _ Program 1 0 6 3 Bootloader 10 10 HTML 10048 VOC 100 9 O day s 0 hour s 4 minute s Yes disabled detected NAT type is full cone 24 12 198 35 MasterVolP VolP to GSM Gateway MAC Address The device ID in HEX format This is very important ID for ISP troubleshooting WAN IP Address This field shows WAN port IP address Product Model This field contains the product model info Program This is the main software release This number is always used for firmware upgrade Bootloader This is normally not changed HTML This is the user interface normally not changed VOC This is the codec program normally not changed System Uptime i l This shows system up time since last reboot Registered i i veer This shows whether the unit is registered to service provider s server PPPoE Link Up l l l This shows whether the PPPoE is up if connected to DSL modem NAT This shows what kind NAT the VoIP Client ATA is connected to via its WAN port It is based on STUN protocol NAT Mapped IP l l l WAN side public IP if connected to LAN of a SOHO router NAT Mapped Port External port detected by STUN Software Version Statistical Status Self explainable Please refer to the page displayed Februaray 2006 Page 18 42 MasterVolP VolP to GSM Gateway 3 2 1 3 Advanced User Configuration To login to the Advanced User Configuration page follow the instruction in section 3 2 1 they will lead You to t
8. display or block Call waiting caller ID Hold Call Waiting Flash Call Transfer Call Forward in band and out of band DTMF Dial Plans etc e Supports fax pass through for PCMU and PCMA and T 38 FoIP Fax over IP e Supports Silence Suppression VAD Voice Activity Detection CNG Comfort Noise Generation Line Echo Cancellation G 168 and AGC Automatic Gain Control e Supports standard encryption and authentication DIGEST using MD5 and MD5 sess e Supports for Layer 2 802 1Q VLAN 802 1p and Layer 3 QoS ToS DiffServ MPLS e Supports automated NAT traversal without manual manipulation of firewall NAT e Supports device configuration via built in IVR Web browser or Central configuration files through TFTP or HTTP server e Supports firmware upgrade via TFTP or HTTP with encrypted configuration files e Supports PSTN pass through able to make and receive VoIP or PSTN calls using same connected analogue phone e Ultra compact wallet size and lightweight design great companion for travelers Februaray 2006 Page 8 42 MasterVolP VolP to GSM Gateway e Compact lightweight Universal Power adapter 3 1 2 Hardware Specification The following table describes the hardware specification of VoIP Client ATA LAN interface 1xRJ45 10Base T LED GREEN amp RED color Universal Input 100 240VAC Power Adaptor Output 5VDC 1200mA UL certified 65mm W 93mm D 27mm H Dimension Operating 32 1040F Temperature 0 400C
9. for direct IP to IP calling Februaray 2006 Page 24 42 MasterVolP VolP to GSM Gateway This value contains the dial plan prefix string typically an ASCII numeric string If it is not blank then this string will be used as a prefix to the target URI string in the To header field of an INVITE message No Key Entry Timeout Default is 4 seconds This parameter allows the user to configure the key to be used as the Send or Dial key Once set to Yes pressing this key will immediately trigger the sending of the dialed string collected so far In this case this key is essentially equivalent to the Re Dial key If set to No this key will then be included as part of the dial string to be sent out Dial Plan Prefix Use as Send Key Local SIP port This parameter defines the local SIP port the IP phone will listen and transmit on The default value is 5060 This parameter defines the local RTP RTCP port pair the IP phone will listen and transmit on It is the base RTP port for channel 0 When configured channel 0 will use this port value for RTP and the port_value 1 for its RTCP channel 1 will use port_value 2 for RTP and port_value 3 for its RTCP The default value is 5004 This parameter when set to Yes will force random generation of both the local SIP and RTP ports This is usually necessary when multiple IP phones are behind the same NAT The VoIP Client ATA sends
10. for uperade every i days default T days Proxy Require 192 163 1 20 SUBSCRIBE for E MATT No do not send SUBSCRIBE for Message Waiting Indication Yes send periodical SUBSCRIBE for Message Waiting Indication Ofhook Auto Dial User ITDiextension to dial automatically when ot hook Februaray 2006 Page 20 42 MasterVolP VolP to GSM Gateway Enable Call E Features No O ves af Yes Call Forwarding amp Call Wamting Disable are supported locally Disable Cail p Waiting Es No E Yes Send DTMF La inaudio viaRTP RFC2833 M via SIP INFO DIMPF Payload 101 Type Send Flash Event E No G Yes Flash will be sent as a DTMF event 1f set to Yes FXS Impedance current setting amp 800 Ohm North America Caller ID Scheme current seting i Belicore Onhook Foltage current setting is 36 o Polarity Reversal Ea No Yes reverse polarity upon call establishment and termination NTP Server time nist gow URI or IP addre 5 Send Anonymous S No L yes caller ID will be blocked if set to Yes Lock keypad update TA xo L yes configuration update via keypad is disabled if set to Yes Syslog Server 192 168 1 20 Syslog Level current seting 5 INFO Administrator password Only the administrator can configure the Advanced Admin Password Settings page Password field is purposely left blank for security reasons after Pressing update and save The maximum password
11. length is 25 characters This field contains the URI string or the IP address and port if different from SIP Server 5060 of the SIP proxy server e g the following are some valid examples sip my voip provider com or sip my company sip server com or 192 168 1 200 5066 This field contains the URI string or the IP address and port if different from Outbound Proxy 5060 of the outbound proxy If there is no outbound proxy this field SHOULD be left blank If not blank all outgoing requests will be sent to this outbound proxy Februaray 2006 Page 21 42 MasterVolP VolP to GSM Gateway Februaray 2006 Page 22 42 MasterVolP VolP to GSM Gateway This field contains the user part of the SIP address for this phone e g if the SIP address is sip my_user_id my_provider com then the SIP User ID is my_user_id Please do NOT include the preceding sip scheme or the host portion of the SIP address in this field User account information provided by VoIP service provider ITSP usually has the digit form of a phone number or is actually a phone number Authenticate ID SIP service subscriber s ID used for authentication Can be identical to or different from SIP User ID Authenticate SIP service subscriber s account password for GXP 2000 to register to SIP Password servers of ITSP SIP service subscriber s name which will be used for Caller ID display G723 Rate This defines the encoding rate for G723 vocoder By d
12. telephone number number then this field will be set to Yes Otherwise set it to No If Yes a user phone parameter will be attached to the From header in SIP request SIP Registration This parameter controls whether the IP phone needs to send REGISTER messages to the proxy server The default setting is Yes Unregister On Default is No If set to Yes the SIP user s registration information will be Reboot cleared on reboot Registration This parameter allows the user to specify the time frequency in minutes the Expiration phone will refresh its registration with the specified registrar The default interval is 60 minutes or 1 hour The maximum interval is 65535 minutes about 45 days Early Dial This parameter controls whether the phone will attempt to send an early INVITE each time a key is pressed when a user is dialing a number If set to Yes an INVITE is sent using the dial numbers collected so far Otherwise no INVITE is sent until the Re Dial button is pressed or after about 5 seconds have elapsed if the user forgets to press the Re Dial button The Yes option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address responses Otherwise the call will most likely be rejected by the proxy with a 404 Not Found error Please note that this feature is NOT designed to work with and should NOT be enabled
13. try to recover the call The busy tone is just to indicate to the transferor that the transfer has failed e Busy tone keeps playing This means we have failed to receive the second NOTIFY from the transferee and decided to time out Note this does not indicate the transfer has been successful nor does it indicate the transfer has failed When transferee is a client that does not support the second NOTIFY such as our own earlier firmware this will be the case In bad network scenarios this could also happen although the transfer may have been completed successfully 3 1 4 4 Attended Transfer Assuming that call party A and B are in conversation A wants to Attend Transfer B to C 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 A then dial C s number then or wait for 4 seconds A and C now are in conversation 3 A can hang up Note When intended Transfer failed if A hangs up the HandTone 496 will ring user A again to remind A that B is still on the call by pressing FLASH or Hook again will restore the conversation between A and B Februaray 2006 Page 12 42 MasterVolP VolP to GSM Gateway 3 1 5 Call Features The Following table shows the call features of VoIP Client ATA Enable Call Waiting Per Call Unconditional Call Forward To use this feature dial 72 and get the dial tone Then dial the forw
14. you want lo review or change any ol yow installation seltinos cick Back Click Cancel lo ext the wizard 6 Click Install Wait till the VoIP Master Manager application will install itself 3 4 4 Define the Com port Connection After installing the manager application launch it and define the Comport to which the VoIP Master is connected 15 NOS 1 Launch the PRI Manager by pressing te on your computer desktop or by pressing Februaray 2006 Page 36 42 MasterVolP VolP to GSM Gateway Start gt Programs gt EuroTech Communications gt VolP Master Manager The VoIP Master Manager window opens LE AMIGO ON VOIP V2 3 1 Gateway Version Welcome to AMIGO ON VOIP V2 3 1 ae GSM Pot EA SIM 1 E SIM 2 Quick Help ga Folow me aa Calback Y Debug Connecting the Gateway Connect the power supply cable to the gateway 2 Correct the cable bo the com por Press the connect button on the S W and then choose the com port you wish bo use Press ok bo start connecting if connected ok green kght wil appear com that you ok to continue 3 Select the Com Port in the computer to which th The connection indicators in the lower right corner of the window blink green After installing the Manager and defining the port connection define port and SIM Settings as described in the following chapter Februaray 2006 Page 37 42 MasterVolP VolP to GSM Gateway Februaray 2006 Page 38 42 MasterVolP VolP to
15. F Finland Sweden e DTMF Denmark Onhook lic Select the onhook A ich to suit different area or PBX Polarity Reversal AS Polarity Reversal to adapt some call charge billing system Default is This parameter defines the URI or IP address of the NTP server which the IP NTP server phone will use to display the current date time If this parameter is set to Yes the From header in the outgoing INVITE Send Anonymous message will be set to anonymous essentially blocking the Caller ID from being displayed Lock keypad If this parameter is set to Yes the configuration update via keypad is update disabled eod Cerner The IP address or URL of the System log server This feature is especially useful yon for ITSP Internet Telephone Service Provider Caller ID Scheme Februaray 2006 Page 27 42 MasterVolP VolP to GSM Gateway Select the ATA to report the log level Default is NONE The level is one of Syslog Level DEBUG INFO WARNING or ERROR Syslog messages are sent based on the following events product model version on boot up INFO level NAT related info INFO level sent or received SIP message DEBUG level SIP message summary INFO level inbound and outbound calls INFO level registration status change INFO level negotiated codec INFO level Ethernet link up INFO level SLIC chip exception WARNING and ERROR levels memory exception ERROR level The Syslog uses USER facili
16. GSM Gateway Via TFTP Server This is the IP address of the configured tftp server If it is non zero or not blank the IP phone will attempt to retrieve new configuration file or new code image update from the specified tftp server at boot time It will make up to 3 attempts before timeout and then it will start the boot process using the existing code image in the Flash memory If a tftp server is configured and a new code image is retrieved the new downloaded image will be verified and then saved into the Flash memory Note DO NOT interrupt the TFTP upgrade process especially the power supply as this will damage the device Depending on the network environment this process can take up to 15 or 20 minutes Via HTTP Server The URL for the HTTP server used for firmware upgrade and configuration via HTTP For example http provisioning mycompany com 6688 Grandstream i 0 5 16 Here 6688 is the specific TCP port that the HTTP server is listening at it can be omitted if using default port 80 Note If Auto Upgrade is set to No VoIP Client ATA will only do HTTP download once at boot up Automatic HTTP Choose Yes to enable automatic HTTP upgrade and provisioning Upgrade In Check for new firmware every field Enter the number of days period VoIP Client ATA will check the HTTP server for firmware upgrade or configuration after the defined number of days When set to No VoIP Client ATA will only do HTTP up
17. O Near O Cost uu Ti Master Management Februaray 2006 Page 31 42 MasterVolP VolP to GSM Gateway 1 From your cellular phone you can dial to the VoIP device 2 The GSM module in the VoIP device provides you with a dial tone of Voice over Internet Protocol 3 You can now dial and make a telephone connection by way of the internet which has near 0 cost Main usage features e Up to 32 cellular phones can use the VoIP device for connection to the internet in parallel e A local desktop telephone can be connected to the VoIP device The desktop phone can send and receive calls via the internet as well as via the GSM network according to telephone prefixes e A follow me function can be activated to serve the desktop phone e If two systems install in remote offices a call from a mobile in one location let say N Y can call a remote cellular user let say in Japan in the cost of a local enterprise Cellphone cost only Februaray 2006 Page 32 42 MasterVolP VolP to GSM Gateway 3 4 2 Set up and Installation Insert SIM and connect the cables as described below 1 Insert SIM card into the VoIP Master as follows a Using the tip of the antenna or a similar object press on the small yellow button on the left side of the gateway as pictured below A SIM drawer pops out b Insert the SIM into the drawer as pictured below Ensure that the angled notch of the SIM is in the matching corner of the SIM drawer u
18. Web BrowSe l cccc ccc ecececee eee ec ec ee esac acess eseeeea ease ee eeeeeeeeaeaeaeaeeeeeeaeesenees 15 3 2 N21 ACCESS the WW Eb GCOnlguratlon Mens dt oil 15 32212 End User COMMGU Fall Missy o a es e tes pl a o a dn lin o o a 15 3 2 lio Advanced User COniGQuratiOn wecacsrsccececcateasscsamcseuaeatns edanesicsndeeuar pacas 19 3 2 144 Saving ine GonligurallOnvCManges ins is 29 3 2 1 5 Remotely rebooting VoIP Client ATA ooccoccccccncccnoconenononcncnoncnonononon cease ceeeeusesuseceeeseceseeeeeseseaeseseaesegeags 29 Restoring the Factory Default SQUINGS ja 30 VOM IAS O O 31 3 4 1 What is the VoIP Master and how it WOPKS cccccece cece cee ec eee e eens ee ee eats eens en eaeaeeeeeeeeaeaeaeeeeseeeaegeeeeneeeeseass 31 42 SETUP ANG Instala WO iveco cecilia 33 3 4 9 Installing the Manager APPIA lO Mica AN Ai 35 3 4 4 Detine the Com port CONNECTION ss alee Palace Phi kolieadaiaa ic EEN E 36 Oe OPO SIM SENGS sida A di meat le cea bd ete 39 3 4 0 1 Dial Settings for the GSM Polis iia 39 Oe IS NOS ea lea 40 3 A020 FONW NIG Seng S ns A a DOES 41 3420 4 AMB SNS eos 42 Februaray 2006 Page 4 42 MasterVolP VolP to GSM Gateway 1 Getting Started Eurotech Communication team is glad you have chosen to use the Eurotech s VoIP Master GSM to VoIP gateway for greatly saving your call costs We will do our best to make your installation efforts as well as day to day configuration and monitoring tasks be pleasant tasks as pos
19. a UDP package to the SIP server periodically in order to keep the port open on the router This parameter defines the interval time that HT286 send the UDP package The default setting is 20 second Use NAT IP NAT IP address used in SIP SDP message Default is blank Proxy Require SIP Extension to notify SIP server that the unit is behind the NAT Firewall Local RTP port Use Random Port keep alive interval This parameter defines whether the phone NAT traversal mechanism will be activated or not If activated by choosing Yes and a STUN server is also specified then the phone will behave according to the STUN client specification Under this mode the embedded STUN client inside the phone will attempt to detect if and what type of firewall NAT it is NAT Traversal behind through communication with the specified STUN server If the detected NAT is a Full Cone Restricted Cone or a Port Restricted Cone the phone will attempt to use its mapped public IP address and port in all the SIP and SDP messages it sends out If this field is set to Yes with no specified STUN server then the phone will periodically every 20 seconds by default send a blank UDP packet with no payload data to the SIP server to keep the hole on the NAT open Firmware Upgrade This radio button will enable VoIP Client ATA to download firmware or configuration file through either TFTP or HTTP Februaray 2006 Page 25 42 MasterVolP VolP to
20. ard number and for a dial tone then hang up Cancel Unconditional Call Forward To cancel Unconditional Call Forward dial 73 and get the dial tone then hang up Busy Call Forward To use this feature dial 90 and get the dial tone Then dial the forward number and for a dial tone then hang up Cancel Busy Call Forward To cancel Busy Call Forward dial 91 and get the dial tone then hang up Delayed Call Forward To use this feature dial 92 and get the dial tone Then dial the forward number and for a dial tone then hang up Cancel Delayed Call Forward To cancel this Forward dial 93 and get the dial tone then hang up Flash Hook When in conversation this action will switch to the new incoming call if there is a call waiting indication When in conversation without an incoming call this action will switch to a new channel for a new call 3 1 6 Fax Support VoIP Client ATA supports FAX in two modes T 38 Fax over IP and fax pass through T 38 is the preferred method because it is more reliable and works well in most network conditions If the service provider supports T 38 please use this method by selecting Fax mode to be T 38 If the service provider does not support T 38 pass through mode may be used To send or receive faxes in fax pass through mode users will need to select all the Preferred Codecs to be PCMU PCMA Februaray 2006 Page 13 42 MasterVolP VolP t
21. available again At menu Cellular Gateways Please Give Us feedback to improve your BRI Gateway product Please let us know your feedback and enhancement ideas to improve the product to your best value Email Support mega2000 no 2 Check your package Items Please verify your package contains the following components some were ordered specific before installation e Main Hardware Device The VoIP Master Gateway e 110 220V Electric Power converter to 24V with cables supplied e VoIP master software Installation CD Installation kit for MS Windows Management Application this User Manual file and additional auxiliary utilities e GSM Antenna To be installed to the VoIP Master Gateway e RS 232 Serial PC COMport connection cable we ll be referred as Comport cable in this manual Februaray 2006 Page 7 42 MasterVolP VolP to GSM Gateway 3 VolP Client ATA Before using this device please perform the following actions 1 Connect the VolP Master Which include the VoIP Client ATA as a built in module to the IP network via the RJ 45 connection near the 2 LEDs and power supply side You must have an account with a VolP termination service provider or you should register an extension with a SIP Gateway Server Get all needed data from your provider such as user name Password server IP addresses ports etc 2 Connect a regular Analog telephone RJ 11 connection to the system and configure it first as a regular VolP client That
22. ceive a busy signal 2 If the telephone number of person B is listed in the Call Back settings of the VolPMaster manager when the phone call of person A is completed the VolPMaster gateway will call person B and provide a telephone line that was previously busy by person A To enable this feature perform the following 1 On the right side of the window set the box to ON Enter desired telephone numbers in the center of the window Next to each telephone number set the box to ON Press Write Settings NS 2 AMIGO ON VOIP 2 3 1 Sn Connect Refresh Exit Callback settine 2 Gateway setting 0 GSM Port FA SIM 1 E SIM 2 BB Follow me J Calback Debug olololalalalaisisa pc o o Y o Y ll o os IND 0I O amp moi i a E i 7 7 woe Write settings 06 G ojo 3 3 5 La lle A N i M m Ready After making these settings your VoIP gateway is ready for operation Februaray 2006 Page 42 42 MasterVolP VolP to GSM Gateway
23. configuration is done using a web interface You will find instructions on page 14 of this manual 3 Test that you can make and receive calls using your regular phone set 4 Run the GSM management software and configure it according to the manual and the interface menu 3 1 Product Overview The report will include various standards been used in each demo and any interoperability issue need to be considered regarding the need for certain standard support what section of the standard are mandatory and what standards implementation are recommended as an implementation reference 3 1 1 Key Features The document will be prepared as contribution of all partners where Albatronics will integrate the contributions Each partner will contribute information for its demo provided equipment regarding with standards support details e Supports SIP 2 0 RFC 3261 TCP UDP IP RTP RTCP HTTP ICMP ARP RARP DNS DHCP both client and server NTP PPPoE STUN TFTP etc e Powerful digital signal processing DSP to ensure superb audio quality advanced adaptive jitter control and packet loss concealment technology e Supports various codecs including G 711 PCM a law and u law G 723 1 5 3K 6 3K G 726 40K 32K 24K 16K as well as G 728 G 729 and iLBC e Supports Caller ID name display or block Call waiting caller ID Hold Call Waiting Flash Call Transfer Call Forward in band and out of band DTMF Dial Plans etc e Supports Caller ID name
24. efault 6 3kbps rate is chosen ILBC frame size This defines the size of the iLBC codec frame The default setting is 20ms ILBC payload type This defines the iLBC payload type The default setting is 97 Preferred Vocoder VoIP Client ATA supports up to 7 different vocoder types including G711 ulaw PCMU G711 alaw PCMA G723 G729A B G726 32 ADPCM G728 and iLBC Depending on the product model some of these vocoders may not be provided in a standard release A user can configure vocoders in a preference list that will be included with the same preference order in SDP message The first vocoder in this list can be entered by choosing the appropriate option in Choice 1 Similarly the last vocoder in this list can be entered by choosing the appropriate option in Choice 7 Silence Suppression This controls the silence suppression VAD feature of G723 and G729 If set to Yes when a silence is detected a small quantity of VAD packets instead of audio packets will be sent during the period of no talking If set to No this feature is disabled Layer 3 QoS This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or Diff Serv Default value is 48 Layer 2 QoS This setting includes two fields The 802 1Q VLAN Tag contains the value used for layer 2 VLAN tag Default setting is blank And 802 1p priority value contains the value of the priority value Use DNS SRV This parameter co
25. grade once at boot up SUBSCRIBE for Default is No When set to Yes a SUBSCRIBE for Message Waiting MWI Indication will be sent periodically Offhook This parameter allows the user to configure a User ID or extension number to Auto Dial be automatically dialed upon offhook Please note that only the user part of a SIP address needs to be entered here The phone will automatically append the 0 and the host portion of the corresponding SIP address Enable Call Feature Default is No If set to Yes Call Forwarding amp Do Not Disturb are supported locally Disable Call Default is No Waiting Send DTMF This parameter controls the way DTMF events are transmitted There are 3 ways in audio which means DTMF is combined with the audio signal not very reliable with low bit rate codec via RTP RFC2833 or via SIP INFO DTMF Payload This parameter sets the payload type for DTMF using RFC2833 Type Februaray 2006 Page 26 42 MasterVolP VolP to GSM Gateway This parameter allows the user to control whether to send an SIP NOTIFY Send Flash Event message indicating the Flash event or just to switch to the voice channel when the user presses the Flash key FXS Impedance Selects the impedance of the analog telephone connected to the Phone port Select the Caller ID Scheme to suit the standard of different area e Bellcore North America e ETSI FSK France Germany Norway Taiwan UK CCA e ETSI DTM
26. he following page The password is case sensitive with a maximum length of 25 characters and the factory default password for Advanced User is admin Password Advanced User configuration page includes not only the end user configuration but also some advanced settings such as SIP configuration Codec selection NAT Traversal Setting and other miscellaneous settings Following is the screen shot of the Advanced configuration page Maurice ontiqiiration VeEVICe LODMMUrATION Admin Password purposely not displayed for security protection SIP Server sini e g sip mycompany com or IP address Outbound Proxy e g proxy myprovider com or IP address if any SIP User ID 212220 the user part of an SIP address Authenticate ID 21252 can be identical to or different from SIP User ID Authenticate es purposely not displayed for security protection Password Name optional e g John Doe Advanced Options Preferred Vocoder ia current setting is PEMU in listed order current setting is POMA choice 2 i ies 3 current setting is G723 e current setting is G729 choice 4 i current setting is 6728 32 T choice choice 6 Surentsettngis 6728 choice 7 G723 rate EX 6 3kbps encoding rate 5 3kbps encoding rate iLBC frame size de 20ms E 30ms iLBC payload type LN between 96 and 127 default is 97 Silence Suppression Ea xo Ei yes current sett
27. ing is iLBC Voice Frames per a up to 1020 32 64 for G711 G726 G723 other codecs respectively Fax Mode ES 38 Auto Detect L Pass Through Layer 3 Qos Diff Serv or Precedence value Februaray 2006 Page 19 42 MasterVolP VolP to GSM Gateway The following window if for advanced configuration regarding IP SIP QoS NAT IP Telephony modes setting Layer 2 QoS 802 1Q VLAN Tag 802 1p priority a 0 7 Use DNS SRF E xo Li yes User ID is phone r number Es No M Yes SIP Registration E Yes L No Unregister On p Reboot L Yes L No Register Expiration aa in minutes default 1 hour max 45 days Early Dial Ei xy EX yes use Yes only if proxy supports 484 response Dial Plan Prefix this prefix string is added to each dialed number No Key Entry i Aa l ins i in seconds default is 4 seconds Use as Dial Key G No i yes if set to Yes will function as the Re Dial key local SIP port 50 aca 5060 local RTP port Ea 024 65535 default 5004 Use random port E No G Ves NAT Traversal TG No ia STUN Ty Company om my o Yes STUN server 1s URI or IP port Keep alive interval in seconds default 20 seconds Use NAT IP if specified this IP address is used in SIP SDP message if specified the content will appear in Proxy Require header Firmware Upgrade Ey TFTP Server 192 168 EN KZ via HTTP Server Automatic HTTP Upgrade E No C Yes check
28. mber call 1 Dial the extension number directly and wait for 4 seconds Default No Key Entry Timeout Or 2 Dial the number directly and press assuming that Use as dial key is selected in the web configuration Other functions available during the call are call waiting flash call transfer and call forwarding supplementary call services 3 1 4 2 Direct IP calls Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy VoIP calls can be made between two phones if e Both VOIP Client ATA and the other VoIP device i e another VOIP Client ATA or other SIP products have public IP addresses or e Both VOIP Client ATA and the other VoIP device i e another VOIP Client ATA or other SIP produces are on the same LAN using private or public IP addresses or e Both VOIP Client ATA and the other VoIP device i e another VOIP Client ATA or other SIP products can be connected through a router using public or private IP addresses To make a direct IP call first pick up the analog phone or turn on the speakerphone on the analog phone then access the voice menu prompt by dial or press the button on the HT286 and dial 47 to access the direct IP call menu User will hear a voice prompt Direct IP Calling and a dial tone Enter a 12 digit target IP address to make a call The follow is a table of the encoding scheme for the most commonly used characters Exa
29. mples If the target IP address is 192 168 0 160 the dialing convention is Voice Prompt with option 47 then 192168000160 followed by pressing the key if it is configured as a send key or wait 4 seconds In this case the default destination port 5060 is used if no port is specified If the target IP address port is 192 168 1 20 5062 then the dialing convention would be Voice Prompt with option 47 then 192168001020 45062 followed by pressing the key if it is configured as a send key or wait for 4 seconds Februaray 2006 Page 11 42 MasterVolP VolP to GSM Gateway 3 1 4 3 Blind Transfer Assuming that call party A and B are in conversation A wants to Blind Transfer B to C 1 A presses FLASH on the analog phone or Hook Flash for old model phones to get a dial tone 2 Then A dials 87 then dials C s number and then or waits for 4 seconds 3 A can hang up Note Call Feature has to be set to YES A can hold on to the phone and wait for one of the three following behaviors e A quick confirmation tone temporarily using the call waiting indication tone followed by a dial tone This indicates the transfer is successful transferee has received a 200 OK from transfer target At this point A can either hang up or make another call e A quick busy tone followed by a restored call on supported platforms only This means the transferee has received a 4xx response for the INVITE and we will
30. ntrols whether the IP phone supports the DNS SRV route function Februaray 2006 Page 23 42 MasterVolP VolP to GSM Gateway Voice Frames per This field contains the number of voice frames to be transmitted in a single TX packet When setting this value the user should be aware of the requested packet time used in SDP message as a result of configuring this parameter This parameter is associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time e g if the first vocoder is configured as G723 and the Voice Frames per TX is set to be 2 then the ptime value in the SDP message of an INVITE request will be 60ms because each G723 voice frame contains 30ms of audio Similarly if this field is set to be 2 and if the first vocoder chosen is G729 or G711 or G726 then the ptime value in the SDP message of an INVITE request will be 20ms If the configured voice frames per TX exceeds the maximum allowed value the phone will use and save the maximum allowed value for the corresponding first vocoder choice The maximum value for PCM is 10 x10ms frames for G726 it is 20 x10ms frames for G723 it is 32 x30ms frames for G729 G728 64 x10ms and 64 x2 5ms frames respectively Fax Mode T 38 Auto Detect FoIP by default or Pass Through must use codec PCMU PCMA User ID is phone If the VoIP Client ATA has an assigned PSTN
31. o GSM Gateway 3 1 7 LED Light Pattern Indication Following are the LED light pattern indications RED LED indicates abnormal status DHCP Failed or WAN No Cable flash every 2 seconds if DHCP is configured flash 2 ds if SIP i figured VOIP Client 486 fails to register ash every 2 seconds if SIP Is configured Message Waiting Indication Februaray 2006 Page 14 42 MasterVolP VolP to GSM Gateway 3 2 Configuration Guide 3 2 1 Configuring VOIP Client with a Web Browser VoIP Client ATA has an embedded Web server that will respond to HTTP GET POST requests VoIP Client ATA is enabled with embedded HTML pages which allow a user to configure the IP phone through a Web browser such as Microsoft s IE and AOL s Netscape 3 2 1 1 Access the Web Configuration Menu First get the IP address of the VOIP Client through section 2 1 with menu option 02 Then access the VOIP Clients Web Configuration Menu using the following URI http Phone IP Address where the Phone IP Address is the IP address of the phone 3 2 1 2 End User Configuration Once this request is entered and sent from a Web browser the IP phone will respond with the following login screen Password The password is case sensitive with a maximum length of 25 characters The factory default password for End User is admin After the correct password is entered in the login screen the embedded Web server inside the IP phone will respond with the following Basic Set
32. o reset the unit to the factory defaults The steps are as follows e Step 1 Find the MAC Address of the device The MAC address of the device is located at the bottom of the device It is a 12 digits hex number e Step 2 Encode the MAC address to decimal digits Please use the following mapping 0 9 0 9 A 22 B 222 C 2222 D 33 E 333 Fi 3939 For example for the MAC address 00 0b 82 00 e3 95 the User encoding should be 00 0222 82 00 333 3 95 e Step 3 Access the voice menu by pressing or the LED button then dial 99 and get the voice prompt RESET e Step 4 Key in the encoded MAC address decimal digits after hearing the IVR prompt Once the correct encoded MAC address is entered the device will reboot automatically and restore the factory default settings Februaray 2006 Page 30 42 MasterVolP VolP to GSM Gateway 3 4 VolP Master 3 4 1 What is the VoIP Master and how it works This device connects GSM cellular telephones to the internet by way of VoIP Voice over Internet Protocol A GSM module including a SIM card is installed inside the VoIP device A SIM card is a smart card that is received with a subscription to a cellular telephone network This following is the communication solution architecture enabled by the VoIPMaster One Location in the Another Location in the GSM GSM Cellphones i Cellphones Cellphones j Base Stations QS T Y T MENO VoIP VoIP
33. pper left corner Ensure that the SIM is flat in the drawer c Return the SIM drawer to the SIM slot on the left side of a Free Gate 2 Connect cables as follows a Insert the antenna to a connector on the right side of the VoIP Master b Insert the communication cable from the PC COMport to the serial COM port socket on the left side of the VoIPMaster c Insert the network cable into a socket on the right side of the gateway and connect it to the computer with the internet connection ort a Thr a 7 EA AAA nagement PC To Internet Landline Phone d Insert the telephone jack from your land line telephone into a socket on the left side of the gateway e Plug the transformer into a wall socket and insert the power cable into its socket on the left side of the gateway After connecting the cables install the Manager and configure the settings for the VoIP Master gateway as described in the following chapters Februaray 2006 Page 33 42 MasterVolP VolP to GSM Gateway Februaray 2006 Page 34 42 MasterVolP VolP to GSM Gateway 3 4 3 Installing the Manager Application Before operation configuration settings must be made in the VoIP Master gateway Configuration is done by a manger application in the computer Install the manager application on the software cd then define the comport connection as described in this chapter 1 Insert the VoIP Master CD into the computer drive PRI
34. server is used to update the firmware of the device 01 02 03 04 05 Direct IP Calling No Voice Messages or If there are voice messages user can dial 9 and dial pre Voice Messages Pending configured phone number to retrieve voice message When entered user will be prompted by dial tone dial the 12 digit IP address to make a direct IP call For details see 4 2 2 Make a Direct IP Call Dial 9 to confirm the RESET or Enter MAC address to restore factory RESET default setting For detail see section 8 EA Invalid Entry Automatically return to Main Menu Notes e Once the LED button is pressed it enters the voice prompt main menu If the button is pressed e again while it is already in the voice prompt menu state it will jump to the Direct IP Calling option dial tone plays in this state x shifts down to the next menu option returns to the main menu 9 functions as the ENTER key in many cases to confirm an option All entered digit sequences have known lengths 2 digits for menu option and 12 digits for IP address Once all digits are accumulated it automatically processes them e Key entry cannot be deleted but the phone may prompt error once it is detected Februaray 2006 Page 10 42 MasterVolP VolP to GSM Gateway 3 1 4 Placing Phone Calls 3 1 4 1 Calling phone or extension numbers There are currently two methods to make an extension nu
35. sible We wish you a smooth operation while greatly saving your office mobile phone calls This chapter is your Map for installation configuration and monitoring tasks and includes a short explanation on each stage as well as references for more elaborated explanations drawings and examples in other chapters The following is a list of tasks you shall perform where you shall go over it sequentially or skip tasks that are optional and not required for your current needs It is advised that you will use the following tasks as your Do To List As a start Check your package Items at Chapter 2 Check your package Items Later proceed with the Gradstream HanyTone 286 VoIP Client ATA Analog to Telephone Adaptor which resides in the VoIPMaster gateway and enables a Web based configuration interface The VoIP Client ATA provides VoIP call origination and termination with PSTN network with some add on supplementary services which are reviewed at Chapter 3 The VoIPMaster gateway adds new capabilities of GSM to VoIP calls origination and termination to the client ATA The following topics of the VoIP Client ATA are reviewed in Chapter 3 Client ATA Product Overview Key Features Hardware Specification and Basic Operations as follows e Getting Familiar with the Key Pad and the Voice Prompt Placing Phone Calls Calling phone or extension numbers Direct IP calls Blind Transfer Attended Transfer Call Features Fax Support LED Light Pa
36. t Recening failed 068 1 In the PIN Code box enter the PIN number of the SIM 2 In the Network box enter the GSM network number of the SIM 3 In boxes 1 through 8 set enter telephone number prefixes to which this SIM can dial 4 Press Write Settings Februaray 2006 Page 40 42 MasterVolP VolP to GSM Gateway 3 4 5 3 Follow Me Settings If there is no answer on the local phone connected to your VoIP gateway you can use a Follow me feature The Follow me feature connects the incoming call to your cellular phone To activate after making SIM settings press Follow me in the left pane The Follow Me Setting window opens Cormect Refresh Exit Follow me setting z E Gateway setting 2 GSM Port SIM 2 s akd Called number Po BB Callback Debug Rings number Write settings Receiving failed 60 1 Set the Mode box to ON 2 In the Rings Number box enter the number of times the local phone will ring before being diverted to the Follow me function 3 In the Called Number box enter the telephone number that you want dialed when the follow me function is activated 4 Press Write Settings Februaray 2006 Page 41 42 MasterVolP VolP to GSM Gateway 3 4 5 4 Call Back Settings A call back feature is available with the VoIP master gateway If person A is having a conversion via the VoIP master gateway and person B attempts to make a phone call via the same VoIP 1 Person B will re
37. tings configuration page which is explained in details below e a joa 7 She a m tT d a Ww 7 i Bal i L _ YA i LE LS CEL J End User PO i purposely not displayed for security protection Password IP Address e i assigned via DHCP default or PPPoE will attempt PPPoE if DHCP fails and following 15 non blank PPPoE account ID Prefered DNS server eie po po S statically configured as IP Address 102 168 El ja Subnet Mask 255 255 255 o Default Router 182 168 1 foi DNS Server 2 24 127 18 Februaray 2006 Page 15 42 MasterVolP VolP to GSM Gateway current setting is GMT 5 00 US Eastern Time New York Davlight Savings E No E Yes if set to Yes display time will be 1 hour ahead of normal Time time edate The following table describes the various configurations to be performed End User This contains the password to access the Web Configuration Menu This field is case sensitive Password with max 25 characters IP Address There are 2 modes under which the IP phone can operate If DHCP mode is enabled then all the field values for the Static IP mode are not used even though they are still saved in the Flash memory and the IP phone will acquire its IP address from the first DHCP server it discovers on the LAN it attaches to To use PPPoE feature please set the PPPoE account settings if the HT 286 is connected directly to a
38. ttern Indication Now you shall start configure the VoIP Client ATA following the Configuration Guide at chapter 3 2 The following configuration actions shall be performed with several guidelines for optional actions Configuring VOIP Client with a Web Browser Access the Web Configuration Menu End User Configuration Advanced User Configuration Saving the Configuration Changes Remotely rebooting VoIP Client ATA Restoring the Factory Default Settings After you have completed the VoIP Client ATA you can start configuring the VoIP Master starting with learning the VoIPMaster concept rule in the network and the way it works at What is the VoIP Master and how it works chapter Februaray 2006 Page 5 42 MasterVolP VolP to GSM Gateway MasterVolP VolP to GSM Gateway Completing this you shall start the installation procedure of the VoIPMaster following the Set up and Installation as follows e Installing the Manager Application e Define the Com port Connection to enable a PC to VoIP master proper connection e Port and SIM Settings to associate and set SIM and Ports accordingly e Dial Settings for the GSM Port to define policies and profile of behaviour when dialling e SIM Settings regarding with usage limits and other optional modes e Call follow me settings to let the system call you while you are away from office as if you where in office e Call Back Settings to let waiting lines make the call when line is
39. ty In addition to standard Syslog payload it contains the following components GS_LOG device MAC address error code error message Here is an example May 19 02 40 38 192 168 1 14 GS_LOG 00 0b 82 00 a1 be 000 Ethernet link is up Februaray 2006 Page 28 42 MasterVolP VolP to GSM Gateway 3 2 1 4 Saving the Configuration Changes Once a change is made the user should press the Update button in the Configuration Menu The IP phone will then display the following screen to confirm that the changes have been saved Your configuration changes have been saved Users are recommended to power cycle the VOIP Client 488 after seeing the above message 3 2 1 5 Remotely rebooting VoIP Client ATA The administrator of the phone can remotely reboot the phone by pressing the Reboot button at the Configurations menu button Once done the following screen will be displayed to indicate that rebooting is underway The device 1s rebooting now You may relogin by clicking on the link below in 30 seconds Click to relogin At this point the user can relogin to the phone after waiting for about 30 seconds 3 3 Restoring the Factory Default Settings Warning Restoring the Factory Default Settings will DELETE all configuration information of the device Please backup or print out all the settings before attempting the following steps Please disconnect the network cable and power cycle the unit before trying t
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