User manual
Contents
1. spiders tar spiderstar VoIP Interface Version 4 0 User manual spiderstar VoIP Interface Windows Internet Explorer E E lol xj go EI http 192 168 0 12 htmifindex2 php Bal 41 x Dis Datei Bearbeiten Ansicht Eavoriten Extras 2 Favoriten e spiderstar VoIP Interface p El 71 e y Seite y Sicherhet Extrasy Ei MSN 5910802 no assignment available Pstn Sip gateways Users Step 1 cc action participant duration s Call special application fax voicemail menu conference break voicemail forward Callrouting Status Logout MSN 5910804 1 conference 1 MSN 50402805 O 2009 Vanillatech GmbH Contents T Introduction EE 3 A EE 4 2 1 on an existing VMWare Server or Player 4 2 2 on an existing LINUX server 4 ECU AA A AAA 5 A vase ttereuivade ers ix beteredengerceavee 5 3 2 General SettihgS geesde geed SERA EEN Ee eh 5 SiO DIDS A wart hale ones wet Ante erie hE 6 KN ele 6 SiS USERS ve See dG nied lead deed deh ele ee Jat 8 Ke CU Te ele Le ME 10 RA Status Cia ege E gege EE REN 13 4 Special appltcattons eee eee eee eeeeeeeeeeeeeneeaeeneeaes 14 4 1 special applications ns cece cece eect ee a nan narrar a 14 4 2 voicemallmen ENNEN LA 4 3 A ett NENNEN EE SERA EEN EN LA 4 4 Conference POD NENNEN NENNEN NENNEN NENNEN ENEE ENNEN LA 4 5 database connection tocdr 14 4 6 branchi Off ICES cuina A NN NENNEN NENNEN LA 4 7 Time based FOULING tis cinse met NENNEN N2. Please consult your carrier what type of signaling to use If you also want to accept incoming calls please enable allow incoming You need to set NAT yes to connect to an ip gateway if you are behind a firewall and don t have a public ip address assigned to your server Also you will have to set your external ip address in the general settings section Some carriers also require a NAT fromdomain and a NAT fromuser setting page 6 Also some gateways have the possibility to redirect calls If you want to enable this feature check allow forwarding Be very careful with that since the pbx system will dial every number your recipient redirected his phone to E g if your recipient set his phone to a premium service your pbx will dial this number and you will have to pay for this Below the sip servers you will find a section called callrouting table Using this section you can define what service should be using dialing what number The area code defines the digits that should be dialed for the specified service E g if you define 0 as area code every number beginning with 0 is dialed with the specified gateway crop first digits means that the first n digits of your number are going to be cropped from the dialed number E g if you dial the number 0043 12345 on your phone and crop the first 2 digits the number 4312345 is going to be dialed If you define a number in add prefix a prefix is added before the dialed numbe
3. ENNEN KENNEN NNN eeh LA e Sending TAXES EE 14 page 2 1 Introduction With spiderstar VoIP Interface you can turn any server into a multi functional communications center in moments our software provides you with an immediately deployable VoIP server which is fully customizable via an easy to use web interface Available functions include fax mode voicemail and customizable voice menus Why use spiderstar VoIP Complete VoIP telephony system SIP Protocol with unlimited number of users 3 subscribers offered free of charge Possibility of connecting to internet telephone services Voice messages forwarded via email or made accessible through a voicemail system Fully integrated fax server fax email gateway for both incoming and outgoing faxes inclusive of T 38 support error free internet based faxing Menu driven operation via a web interface Multilingual German English Spanish and French languages available at present WLAN optimized make calls over currently laid wiring Connect your home office other location to your main office via almost any DSL connection using a VPN Simple video conferencing Hold telephone conferences in up to five conference rooms simultaneously Define preferred hold music with a personally chosen MP3 Easy creation of voice menus e g Choose 1 for general information 2 for Interface adaptable to every specific requirement page 3 2 Setup 2 1 on an
4. data records f Settings z ID date caller destination last action duration A Pstn RI 2 2006 12 29 20 59 04 01 25 1125 VoiceMailMain 3 H Sip gateways 1 2006 12 29 20 58 31 01 25 1125 VoiceMailMain 1 A 5 Users a 5 Callrouting E Status 9 Sip users i H Cdr Li A f Syslogs 8 5 H Logout Fertig TF Tf TP Oen Fa 100 7 page 13 4 Special applications 4 1 special applications Some callrouting require spezial data to be entered in the callrouting table For more info contact the support 4 2 voicemailmenu To contact the voicemailmenu to record your unavailable message you have to set an internal number for the user Administrator This is the internal base number for the voicemail menu Default 11 The voicemail menu of a given user can be reached then by dialing Administrator number user So e g if Administrator has number 11 and user has number 25 his voicemailmenu can be reached dialing 1125 4 3 transfer To transfer a user please press after that the number the call should be transferred to 4 4 conference rooms To connect to a conference room dial and after that the room number 1 5 If you want to invite users to a conference dial the user and after that tranfer him or her to ther conference room pressing and after that the room number 4 5 database connection to cdr The cdr database can be accessed using ODBC for postgresql Therefor connect to the th
5. e server and enter user postgres password postgres and database cdrdb The postgresql db listens on port 5433 4 6 branch offices To interconnect 2 pbxes do the following add a new ip gateway and enable incoming calls on both machines with same username password Enter one callrouting table on both machines Add one user sip on both machines who has the same username as the user defined in the ip gateway section Also ensure that the passwords are identical 4 7 Time based routing To define a timebased routing please use the Special application function in callrouting As Command please enter Gotolftime and as argument use Time duo user 1 e g 9 00 17 00 duo0 23 1 In this case a call would be directed to user 23 if it is e g 10 00 am I fit is later e g 06 00 pm the call would be directed to the next step in callrouting 4 8 Sending faxes The server supports faxes with T 38 protocol This protocol has to be supported by your carrier If not the usual g711 codec will be used and transmission failures will be likely On your spiderstar VoIP Interface there is a fully integrated smtp server installed This server is responsable for sending and receiving faxes To send a fax please create a new Email and address it to fax ip address of your server To get this working please configure your default SMTP Server to rely mails to this server or use the integrated smtp server as default mailserver for you
6. existing VMWare Server or Player To install the spiderstar VoIP Interface on a VMWare Server you need to download the recent VMWare Image at http download spiderstar com vmware After unpacking the image it can be started automatically with a VMWare player or it can easily be imported in an existing VMWare server After running the image it will automatically download the recent version of spiderstar VoIP Interface on port 80 and save it to the VMWare image Because of this please be aware that the VMware image already has internet access on first boot When the download has finished you will be able to do settings directly on the vmware console e g changing the ip address 2 2 on an existing Linux server For setting up spiderstar VoIP Interface on an existing Linux server please download the recent version of the software with the following command wget http download spiderstar com linux spiderstar latest tar gz and store it in the root directory of your server After unpacking the archive tar xfz spiderstar latest tar gz please copy the init script that you can obtain at http download spiderstar com linux startup script spiderstar in the following directory etc init d If you want to start the voip server automatically on system boot please set the appropriate links e g ln s etc init d spiderstar etc rc3 d S30spiderstar ln s etc init d spiderstar etc rc0 d Kl0spiderstar page 4 3 Features 3 1 Log
7. fax2emails should be routed to internal voicemailbox on absence The voicemailbox will answer on internal calls if the sip phone isn t logged on internal voicemailbox no response The voicemailbox will pickup if the user doesn t answer after n seconds voicemailbox internal on busy The internal voicemailbox will pickup if the user is busy voicembailbox internal always The internal voicemailbox will pickup always no internal calls will be accepted callgroup An incoming call on a phone that belongs to a callgroup can be picked up by another user how belongs to the same callgroup The user can do this by dialing 8 on his phone Send voicemails Defines whether the system should send voicemails to the user s email address in case someone left a voicemail message If not checked the user will only be able to get his voicemails using the voicemail menu Say callerid announce time Defines whether user hears the callerid respectively the calling time when checking his voicemailbox Store no voicemails If set there will be no voicemails stored on the server and user will not be able to retrieve his voicemails with the voicemail menu Billing account Account for that user that will be stored in the db for every call Important for later billing page 8 CLIP Callerid presentation The callerid that will be displayed on the called phone when making phone calls Direct connection If checked the server tr
8. fter s 8 on busy always callgroup voicemail settings extended send voicemails iv say callerid y announce time IT store no voicemails SIP settings extended billing account 12 direct connection H NAT Mode E Port 5060 E qualify Codecs ulaw alaw g729 DTMF Mode Digital RFC2833 y 2003 2009 spidersta more settings x es ns RR 3 6 Callrouting Using the callcouting section you can define what should happen when a call is received Every did and every ip gateway where accept incoming is flagged will be listet in that section Below every entry you have the possibility to define new assignments Every assignment is one step in the callrouting table And in every step you are enabled to do different actions like call a participant or let answer the voicemailbox page 10 spiderstar YoIP Interface Windows Internet Explorer H El x Ge y http 1192 168 0 12 htmifindex2 php az Datei Bearbeiten Ansicht Favoriten Extras yy Favoriten SS spiderstar VoIP Interface E 7 Ar Seter Sicherheit Extras Si spiderstar Callrouting Settings Psin Select MSN an Si ateways da ic MSN 1206995 Users asert new ster 1 call e pe seconds change Callrouting ser ee 2 Status MSN 14 Logout insert new step JE 1915190108 191910D10d5 1015180108 Insert step bb Action participant duration s OK CANCEL new ass
9. ies to make direct connections between users and speech is not routed through the server allow reinvite DTMF Mode If set to digital DTMF tones will be transmitted using the RFC2833 standard Otherwise they will be transmitted via audio NAT Mode If set the user will be regarded as behind a firewall This is only useful if your voip server is located outside the firewall Port Port for sip users Default is 5060 Qualify Sometimes it is necessary that the status online offline of a user is checked periodically Codecs Defines the possible codecs and speech compression for the given user Alaw and ulaw g711 define uncompressed speech processing while g729 uses compression Please note that g729 requires the purchase of an appropriate license page 9 gt spiderstar YoIP Interface Windows Internet Explorer E Ce ER http 192 168 0 12 htmisindex2 php Datei Bearbeiten Ansicht Favoriten Extras 2 3 Favoriten S spiderstar VoIP Interface tm v Seite Sicherheit v spiderstar internal participants Settings participant internal number email Pstn administrator 11 SE christian 13 christian clas spiderstar de adm Sip gateways RE administer participant Callrouting username internal number Status numeric password Logout S email address christian clas spiderstar de internal voicemailbox de 1015190108 1915100105 301518010 on absence no response a
10. ignment Clicking on the button new assignment respectively insert new step a box appears where you are able to define what actions should be performed in the specific step Insert step Action participant duration s OK CANCEL change pressing the change button allows you to change a specific action delete Pressing delete removes a specific step or action page 11 You can choose among the following actions Call The action call dials a specific internal pariticipant who was entered in the users section This can be an ip address the internal number of a participant or the participants name If you want to enter an ip address you will have to enter the ip address as the name of the user in the users section If you specify a duration only the participant is only called the time you entered in seconds After that time the next step will be performed If no other step is available the caller will hear a busy tone Voicemail If you select voicemail the caller will be connected to the voicemailbox of the specified internal user forward Using the action forward you can forward a user to a given numer voicemail menu To listen to your voicemails from remote you can call your voicemail menu fax Having set the fax action incoming faxes will be redirected to the users email address So internal number is the internal number
11. o be played one after the other page 5 E spiderstar YoIP Interface Windows Internet Explorer 2 al oO x G htp 192 168 0 12 htmifindex2 php Soogle Dir Datei Bearbeiten Ansicht Favoriten Extras 2 q Favoriten Za Y mp Seite y Sicherheit Extras e spiderstar VoIP Interface System settings Settings at 09 04 Pstn Sip gateways St t Users System reload reboot halt Callrouting Status Softwareupdate Logout System language language english Systembackup Music on hold gt 11 Fehler auf der Seite III TI TI TS internet fa Rio 7 3 3 DIDs Menu point PSTN allows the administration of external DID numbers These will be saved in a list To use those DIDs your VoIP gateway has to support signaling of those numbers If your voip gateway doesn t support signaling of DIDs incoming calls will be routed to the base number of your provider that is automatically inserted in Call1routing when enabling the option allow incoming in the settings of a VoIP gateway 3 4 Services If you want to connect to an ip gateway for calling participants in the whole world you will have to enter these gateways in the sip gateways section below the Services menu point For adding a new server please press add new server After that enter the name of the gateway the servers ip address or dns name username and password for connecting to the server Signaling is either SIP or IAX2
12. of the user the fax should be sent to by email pdf attachment conference 5 conference rooms enable participants to meet each other virtually break break causes the pbx system to wait n seconds before doing the next step special If you have spezial requirements the support may help you application realizing these requirements using special applications ivr Using an ivr interactive voice response menu a user specific announcement can be uploaded as wav file With the dimension it is possible to define how many options the caller are given A dimension of 0 means that there is only the uploaded announcement played back to the user and after that the caller is redirected to the next step If the user is enabled to select an option by pressing a number on his phone dimension gt 0 there will be new numbers available in the callrouting section automatically named IVRx n whereas x ist he number of the ivr menu automatically assigned and n ist he possible option for the user Below these newly generated numbers new steps can be assigned page 12 3 7 Status gt cdr In the cdr section every call is going to be logged spiderstar YoIP Interface Windows Internet Explorer Go Y http 1192 168 0 12 htmlfindex2 php Datei Bearbeiten Ansicht Favoriten Extras 2 E 4 XK Google tas d jy Favoriten spiderstar VoIP Interface fy gt Bl ev Seite Sicherheit y Extras cali
13. on The system can be accessed using the web interface Therefore enter the system s ip address in the web browser Having entered the username and password in the logon screen press ok to enter the configuration menu If you logon the first time the username will be Administrator and the password is spiderstar VoIP Interface Sa password s a K a y s kal 7 gt EI 3 2 General settings A new nameserver can be set in the section general settings as well as a new smarthost A nameserver is needed e g for sending voicemails to the specified email address and for registering with ip gateways The FQDN is the dns name or the external ip address that points to your spiderstar net pbx system For example a FQDN extern ip address is needed by some ip gateways to allow registering behind a nat firewall You will have to try out wheather your provider will need this to be set or not To reload settings that you made e g in the section msn you will have to use the reload settings button in general settings Using the button software update you are able to update your pbx system with service packs you received from the support To upload your individual music on hold you are able to upload an mp3 file up to 10 MB using the function add mp3 Music on hold is played automatically while transferring a caller or placing someone on hold You can upload multiple files that are going t
14. r e g if you dial the number 0043 12345 on your phone and add the prefix 01080 the number 01080004312345 is going to be dialed provider is the ip gateway you had entered in the sip gateways section before Le spiderstar oIP Interface Windows Internet Explorer oO xj go y htp 192 168 0 12 htmlfindex2 php Dr Datei Bearbeiten Ansicht Favoriten Extras 2 7 Favoriten spiderstar VoIP Interface ad y spiderstar A Vol ntertace Settings callrouting table Pstn e area code crop first digits add prefix provider 0 0 stars17 hangs remove sip gateways 5 0 stars17 callrouting table Users Add assignment Callrouting area code crop first digits add prefix provider Status Logout OK CANCEL SUICIDIOS 1015100108 10158100108 101510010 a TTT Tei Fa Rios 7 page 7 3 5 Users Using the users section you can define internal participants that are able to connect to the pbx using sip procotol username login name for the web configuration panel internal number The number where the user can be reached internally This number is used in the callrouting section The internal number of the user Administrator is the number for the voicemail menu numeric password Password for the user that is used to authentify at the voicemailmenu and to identify the users sip phone email address email address of the pariticipant where voicemails and
15. r email client Enter the complete fax number in the subject line of your Email and attach a pdf document to it page 14 After sending this email the server will start connecting to the remote fax machine immediately page 15
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