HOUYUAN IP PBX-02\04\08-User Manual-V2.0-EN
Contents
1. he tee de dee nba de iens 20 Overview of the IP PBX 02N0408 00 0 0 21 2 0 Access to the IP PBX 02Y04408 aiana 21 2 1 HOW 16 U 21 Np pec 22 PU ls c e 22 23 RS232 Console Post or MINICOM 00 00 2 13 3 0 Web Operation or IPPBX DE OUS odi rendered ride i D Rd EI Er RE 24 Gal Elortie 4 a 1 1 1 12121 1 1 eet o an eren de EO E ae ee tel 24 3 2 Extensions eee ete ra eet e HER E ERE dba Ee Eod 26 3 3 PBX LII 28 3 311 Outgoing Calling Plule bito remet en aine rove karma QE E epo rues Riad 28 3 312 Incoming Calling Rule 23 24 iranata 1 IGNES sirtida 25 85 8150616 3314 26 VoiceMail CAI T 3 3915 3 916 MUSIC oni Hold re merece E 27 ce Wee ERIT m 282. Attach recordings This option defines whether or not voicemails arelCheck box Check to e mail sent to the Users e mail addresses as attachments Note You need to have an smtp server configured for this functionality Template for From Str Voicemail Emails ourcompan y null Subject New voicemail from VM CALLERID for VM MAILBOX Template Variables VM MAILBOX The recipient s extension VM CALLERID The caller id of the person Hello VM_NAME you t TAB received a message VM NAME Recipient s firstname andlasting VM DUR at lastname VM DATE from VM DUR The duration of the voicemail VM_CALLERID message This is message VM MSGNUM in your voicemail Inbox who left the message VM_MSGNUM The message number in your mailbox S VM DATE The date and time the message 3 415 Directory Setting Dialing the Directory Extension would present to the caller a directory of users listed in the system telephone directory from which they can search by First or Last Name To add or remove a user from the system telephone directory edit the In Directory field of the user Preferences for Dialing by Name Directory Directory setting PBX Management system Extensions Directory Settings the caller a directory of users listed in the 5 ove a user from the syste Dialing the Director Firs ephone directo edit the In Directo Directory
3. m IntervalA name for the time interval Str Name By day offChoice an available day of week for the time interval Mon Tue week Wed Thu Fri Sat Sun By Days of aChoice some available days of month for the time Dateof Month interval January Febr uary March April May J une july Aug ust Septemb er October n ovember De cember all Time Choice an available time slot for the time interval 00 00 24 0 0j 1 Time intervals using in incoming call 2 Time intervals application rule 00 00 24 00 mon sum 1 31 January February March April May June july August September O ctober november December all time intervals timeinterval date mon tue Monday to Tuesday of weekly 3 321 Conference rooms The conferencing function of Asterisk is similar to a Tele conference call where multiple callers can call in and participate in a two way conference like in a party room where everyone can talk and listen to one another or just to listen to a Tele presentation www houyuanhk com HOUYUAN PBX Management system New Conference Brid Extensi ge on 6300 Password Options Conference R Pin Code 000 P Admin PinCode 123 oom Options F Play hold music for first caller 7 J Enable caller menu Announce callers J Quiet Mode J 7 Waitfor marked user Update Conference Marked Admin user Extension 9989 Close conference when last marked user exits Extension E
4. Analog Hardware 2 1 Tone Region United States North America Reset all Previous Digital Trunks Information Advanced Settings Module Name wctdm24xxp Opermode Y FCC a law override fxs honor mode boostringer 7 fastringer 7 lowpower ring detect MWI mode Send CalleriD After m Region Select the tone region according to yourComboBox United country if it does not have your country s Status North name in the dropdown list please ask your America service operator which kind of tone region is used in your area Module Name The name of Module Textbox Wctdm24xxp Opermode Specifies On Hook Speed RingerComboBox USA Impedance Ringer Threshold current Limiting TIP RING voltage adjustment minimum Operational Look Current and so lon Please choose your country or your nearest neighboring country a law override Specifies the codec to be used for analogComboBox ulaw fxs honor mode This option allows the user to determine ifComboBox FXO modules they would like opermode characteristics applied to trunk FXO modules only or both trunk FXO and station FXS modules boostringer This option allows the user to defineComboBox nomal whether they require normal ringing voltage 40v or maximum ringing voltage 89v or analog phones attached to station FXS modoules www houyuanhk com HOUYUAN fastringer This option sometimes used in co
5. 3 918 Voice Menu Prompts iiss nia ne t d prec dp RE ER adn 30 Essa b E 30 3 320 Time Initervals ecrire eter Un send tte aE 32 3 321 Conference rooms esses nennen 33 Exc 35 3 4 System 51 E H 36 3 411 Configure HardWare ae 1 repairs nep eee eee Em br isa inan 36 3 412 Configure trunks usui oe aS EQ RM Ert SEN UR a RU dI 38 3 413 SMTP Setting 1 1 40 3 414 Voicemail 56111116 acs sendas i Rd adn e dendi 40 3 415 Directory Setting a d dies aae 03 3 416 Call Feature tenete nee eed nne bete rh rl eoe NAE EEE aeaii 44 ev cpm per C 45 3 418 IP table Firewall irte er ater ee LU eee 48 3 419 BACKUP isie roni eie e D e E nevi alles 48 3 5 DiagfioStlCS 33i cg pU ac o pande Hat ete ED correpta 48 3 51 Active Channels aia ists teo ee ee bee eth eae RS 48 SGAM ssi amma E Re nM M 49 3 61 CDR VIGWeTF iiec opere ener arcad de Aa a qe EO dH ed 49 3 62 VAX SUMING 50 Bo ees eae 11 11 DE DD enn nner rere rere eee rere 50 5f quac eem 51 3 65 Network SOUMING M 51 eA rli ep 52 3 57 Server MBSSS B 9 avisa En ish p eM E s ME Urt N
6. an IP network IAX Inter Asterisk Exchange Protocol is a communications protocol for setting up interactive user sessions IAX is similar to SIP RTP Real Time Transport Protocol RTP is used to encapsulate VoIP data packets inside UDP packets RTP provides end to end network transport functions suitable for applications transmitting real time data such as audio video or simulation data over multicast or unicast network services UDP User Datagram Protocol UDP is a communications protocol that offers a limited amount of service when messages are exchanged between computers in a network that uses the Internet Protocol IP TCP Transmission Control Protocol TCP is a set of rules protocol used along with the Internet Protocol IP to send data in the form of message units between computers over the Internet SMTP Simple Mail Transfer Protocol SMTP is the de facto standard for electronic mail transport across the Internet TOS Terms of service the ToS or TOS are rules by which one must agree to abide by in order to use a service Unless in violation of consumer protection laws such terms are usually legally binding DTMF Dual tone multi frequency DTMF signaling is used for telephone signaling over the line in the voice frequency band to the call switching center The version of DTMF used for telephone tone dialing is known by the trademarked term Touch Tone and is standardised by ITU T Recommendation Q 23 Other multi freque
7. the client server computing model TFTP Trivial File Transfer Protocol TFTP is a file transfer protocol with the functionality of a very basic form of File Transfer Protocol FTP TFTP could be implemented using a very small amount of memory It was therefore useful for booting computers such as routers which did not have any data storage devices It is still used to transfer small amounts of data between hosts on a network such as IP Phone firmware or operating system images when a remote X Window System terminal or any other thin client boots from a network host or server DNS Domain Name System The DNS is a distributed hierarchical naming system for computers services or any resource connected to the Internet or a private network It associates various information with domain names assigned to each of the participants Most importantly it translates domain names meaningful to humans into the numerical binary identifiers associated with networking equipment for the purpose of locating and addressing these devices worldwide MAC Media Access Control address The MAC is a unique identifier assigned to network adapters or network interface cards NICs usually by the manufacturer for identification If assigned by the manufacturer a MAC address usually encodes the manufacturer s registered identification number IPv4 Internet Protocol version 4 The IPv4 is the fourth revision in the development of the Internet Protocol IP and it is the firs
8. Network Setting PBX Management system Networking setting WAN Interface VLAN Interface for WAN DHCP no x VLAN Vian number 100 Vian IP address 2 Hostname 04 Domain switchfin org IP address 192 168 1 100 Subnet mask 255 255 255 0 Man Gateway 192 168 100 1 Gateway 192 168 1 1 DNS 192 168 1 1 NTP pool ntp org System TimeZone TimeZone America New_York www houyuanhk com HOUYUAN 3 66 Firmware PBX Management system Update Firmware 9 Web Update HTTP URL TFTP Server Select Fale Reset Configs 7 Keep Network Settings 3 67 Server Message Manage System Services List of Services asterisk Asterisk soft switch service asteriskwatch Asterisk Watchdog timer cron Time based job scheduler dahdi Digium Asterisk Hardware Device Interface dhcpd DHCP service for network Multicast DNS and DNS Service Discovery Network interface service Network Time Protocol Point to Point Protocol over Ethernet Virtual Local Area Network www houyuanhk com HOUYUAN 3 68 Server Notes PBX Management system New System Note Short Description news Created By Fdwin Note one by one Case of IP PBX 02 04 08 Figure Network Topology In the network topology above user 6020 6001 6002 6008 will be registered to IP PBX 02 04 08 After configuration it will realize the following function 1 Theinternal user 6002 and user 6001 can call each other directly 2 6001 6002
9. calling rules and can be ComboBo Null managed X CallerID The Caller ID CID string used when this user calls Textbox Null another internal user OutBound Caller ID that would be applied for out bound calls Textbox Null CallerlD from this user Note that your ability to manipulate your outbound Caller ID may be limited by your VoIP Enable Check this box if the user should have a voicemail Selected Not Voicemail for account selected VoiceMail Voicemail Password for this user Textbox Null Mailbox Voicemail Mailbox for this user Textbox Null Email Address The e mail address for this user Textbox Null SIP Check this option if the User or Phone is using SIP selected selected or ia CID IAX Check this option if the User or Phone is using IAX selected selected or is an IAX device www houyuanhk com HOUYUAN Analog If this user is attached to an analog port on the ComboBo Null Station system x please choose the port number here Codec Choose priority codec ComboBo u law GS NAT Try this setting when Asterisk is on a public IP selected selected communicating with devices hidden behind a NAT device broadband router If you have one way audio problems you usually have problems with your NAT configuration or your firewall s support of SIP RTP Can Reinvite By default Asterisk will route the media steams selected N
10. channels It is an embedded open source Linux system with built in SIP IAX2 proxy server and NAT functions It provides a solid uniform platform for Mobile and VoIP communications Targeting for SOHO user and SMB market with an easy to use graphical interface HOUYUAN IP PBX provides a cost saving solution on their telecommunication data needs With these devices company with branch offices in different countries can be easily combined together to work like a virtual single office through internet FXO FXS and PSTN network 1 2 Hardware CPU 400MHz Blackfin 532 Chip 2 x FXO FXS ports and four analog ports NAND flash 256 M SDRAM 64M 1 3 System Open Source uClinux 1 4 Features FXO FXS ISDN Support 0711 0729 codec Voicemail Voicemail groups 3 way Calling Conferencing Follow Me Call Feature In directory Call Waiting Call Queues Pickup Group Ring Group www houyuanhk com HOUYUAN Is Agent Music On Hold Voice Menus Voice menus Prompts Time intervals Backup Update 1 5 Applications SOHO SMB telephony system Hosted service IVR system 1 6 Interface 1 X RJ45 port 1 X Power port 1 X RS232 port 8 X FXO FXS channels www houyuanhk com HOUYUAN Overview of the IP PBX 08 VOIP Server 2 0 Access to the IP PBX 02 04 08 2 1 HOW to Login You need a PC to access to the IP PBX 02 04 08 there are four ways for you to access the IP PBX 02 04 08 1 Web page 2 SSH 3 Console
11. pplication Map Add an application for PBX Dial Options Dial Options t Option Allow the called party to transfer theCheck box Uncheck calling party by sending the DTMF sequence defined on the Feature Codes page T Option Allow the calling party to transfer theCheck box Uncheck called party by sending the DTMF sequence defined on the Feature Codes h Option Allow the called party to hang up byCheck box Uncheck Sorang the an tha Laeta Cadac dafina _ H H Option Allow the calling party to hang up by Check box Uncheck sending the k Option Allow the called party to enableCheck box Uncheck parking of the call by sending the DTMF sequence defined on the Feature Codes K Option Allow the calling party to enableCheck box Uncheck parking of the call by sending the DTMF sequence defined on the Feature Codes 3 417 Options This component is used for administrator to manage the system it includes the following modules General Preferences www houyuanhk com HOUYUAN PBX Management system General Preferences General Preferences Language Settings Change Password Reset Configuration Reboot Recording Settings DHCP Server Global OutBound CID 7 Operator Extension none Internal Ring Timeout 20 Outbound Ring Timeout 20 Extension preferences User Extensions 6000 Conference Extensions 6300 VoiceMenu Extensions 7000 Rin
12. 6008 can communicate with outside through IP PBX 02 04 08 by FXO FXS 3 User 6001 and 6030 can call each other through VoIP trunk although they are registered to different IP PBX 4 User 6020 and 6001 can call each other directly although they are not in the same network segment 5 Voicemail 6 IVR Conference Ring Groups Agents 0 Follow me 1 7 8 9 1 11 Call pickup www houyuanhk com HOUYUAN How to Make Internal Calls through IP PBX 02 04 08 Access to the Web Page of IP PBX 02 04 08 by Browser After connecting IP PBX 02 04 08 to LAN please open your browser of PC with OS and input the IP Address of IP PBX 02 04 08 the default IP address is 192 168 1 167 Please input the default Username admin Password admin in the presented screen above PBX Configuration Engine eame admin Password eeeee Add up Users from Web Page of IP PBX 02 04 08 First Add up a DialPlan Before users add up user they have to add up a DialPlan please click on Dial Plans New DialPlan the writer creates a DialPlan like the following PBX Management system Create New DialPlan DialPlan Name DialPlani Include Outgoing Calling Rules Include Local Context Rules You do not have any calling Rules defined 4 Default click here to manage calling rules 4 Fax 7 Parkedcalls 4 Spy 7 Conferences 4 Ringgroups 7 Voicemenus Queues 7 Voicemailgroups 7 Directory After configuring please c
13. Check box unCheck Wait for markedPrevent conference participants from hearingCheck box unCheck 1 Conferencing application MeetMe confno options pin Enters the user into a specified MeetMe conference ex MeetMe EXTEN MslqwxaA 1 disable you are currently the only person in this conference message for first member a set admin mode A set marked mode www houyuanhk com 34 HOUYUAN b run AGI script specified in B MEETME AGI BACKGROUND c announce user s count on joining a conference d dynamically add conference D dynamically add conference prompting for a PIN At the pin prompt if the user does NOT want a pin assigned to the conference they should hit the key e select an empty conference E select an empty pinless conference F Pass DTMF through the conference l announce user join leave with review l announce user join leave without review M enable music on hold when the conference has a single caller m set monitor only mode Listen only no talking p allow user to exit the conference by pressing P always prompt for the pin even if it is specified q quiet mode don t play enter leave sounds Record conference records as MEETME RECORDINGFILE using format MEETME RECORDINGFORMAT s Present menu user or admin when is received send to menu set t
14. DialPlan is a set of Calling Rules that can be assigned to one or more users Please select the Dial Plans option Click on New DialPlan button the following table displays the parameters of Dial Plans PBX Management system Create New DialPlan DialPlan Name DialPlani Include Outgoing Calling Rules Include Local Context Rules You do not have any calling Rules defined v Default click here to manage calling rules 7 Fax 7 Parkedcalls 4 Spy 7 Conferences 4 Ringgroups 7 Voicemenus 7 Queues 7 Voicemailgroups 7 Directory DialPlan Name he name of DialPlan which is a unique label os DialPlan1 to Include Outgoing Select outgoing call rule which you use selected Not seclect Calling Rules www houyuanhk com HOUYUAN Include Local Local context is used for general using check box Select all Contexts Rutes configuration 3 314 RingGroups Define Ring groups to dial more than one extension simultaneously or to ring more than one phone sequentially This feature may also be called Hunt groups Please select the Ring Groups option from the vertical menu on the left of the main page then they can get the following screen PBX Management system New RingGroup Extension for this ring group 6400 Ring Group Members Available Users 6000 SIP 6001 SIP 6000 IAX2 6001 IAX2 Ring Group Options Strategy Ring in Order Seconds to ring each member 39 If no
15. Extension Also read the extension number Use first name instead oflastname www houyuanhk com HOUYUAN Directory Extension to dial for accessing the Namelnt Extension Directory Also read theln addition to the name also read the extensionCheck box Uncheck extension number to the caller before presenting dialing number options Use first nameAllow the caller to enter the first name of a user inCheck box Uncheck instead of lastthe directory instead of using the last name name 1 Directory application Directory default default ef 3 416 Call Feature Feature Codes and Call parking preferences Features Codes PBX Management system Feature Codes amp Call Parking Preferences Feature Codes Call Parking Application Map Dial Options Feature Codes Blind Transfer default is 2 Disconnect defaultis Attended transfer Call Parking One Touch Recording Pickup Extension Cancel J Save Features Codes Blind Transfer default is Check H box amp amp lnt Disconnect default is Check box amp amp lnt Attended transfer Check box amp amp lnt Call Parking Packing a call Check box amp amp lnt www houyuanhk com HOUYUAN Call Parking Preferences Call Parking Extension to Dial to Park a call Int 700 Preferences What extensions to park calls on Int 701 720 Number of seconds a call can be parked for Time
16. HOUYUAN ip PBX 02 04 08 Product Guide Version 2 0 2012 ID TONT 2 HOUYUAN Contact HOUYUAN The Introduction of HOUYUAN HOUYUAN Technologies is a global leader providing next generation converged communication products and services to Small and Medium Sized Enterprises SMEs and service providers Our flagship IP PBX Series products seamlessly integrate voice data security IT applications and real time collaboration Our converged service platforms for enterprises create long term value for our customers by increasing revenue opportunities enhancing communication efficiency and reducing operational costs Contact Sales Address FL2 Block D 438 shajing east road baoandistrict Shenzhen China Tel t 86 755 668021 64 Fax 86 755 27286550 E mail houyuan houyuanhk com Contact Technical Support Tel 86 018261572711 E mail support houyuanhk com Website Address http Awww houyuanhk com Download Center http www houyuanhk com products voip html www houyuanhk com HOUYUAN Content 1 0 Introduction of IP PBX O2 O4 08 oo ee ee ccceeceeseeeceeceeeseeeeeeceecaeeaeeaeeceesaeeaeeeseeseesaeeaeeeteres 19 1 1 IPP M02 0406 e ree nee 19 1 2 Hardw re e ei Reade Rao Sia Ue Ruan Rei E dad Ree es 19 13 SY SIC IM e 19 14 FOAtUneS te M P 19 1 5 AppIICatioris io oes Bam rdiet Ba UU ee ee ee Ad 20 1 6 Interface ient tete ith
17. Parity None Stop bits 1 Flow control None LIII Ede www houyuanhk com 3 0 Web Operation of IPPBX 02 04 08 3 1 Home In the system status screen it displays the functions users configured such as trunks extensions conference and so on The following table is the options description of trunks OOo 07 Extensions FBX Features System Setup Diagnostics Admin System Home Uptime 20 21 34 up 1 21 load average 0 29 0 06 0 0 Trunks Extensions E Free Busy A UnAvailable 5 Ringing No Extension assigned Check Voicemails VoiceMailMain No Extension assigned Dial by Names Directory www houyuanhk com HOUYUAN Description Status The register status of trunks Trunk The name of trunks Type The type of trunks Username The username of SIP IAX trunk Port Hostname IP Address port 1 The register status of trunks include three kinds Unregistered Request Sent Registered 2 The type of trunks VoIP trunk including SIP and IAX Analog trunk Service Provider The parameter of extensions in the following table Name Description Extension The status of users Name label The name of users Status Display voice message Type SIP users IAX users Analog users 1 There are four kinds status of users when the light of Extension list displays gray means the user does not register that is Unavailable when the light of Extens
18. Prepend these digits before amp aling cancer Saws 4 Hook on the outgoing calling rules in dial plan in IP PBX 08 www houyuanhk com HOUYUAN Now users can call from 6001 to 6030 by dialing 96030 Voicemail Users can configure Voicemail in the option of Users for example 6005 which the writer has configured in 3 319 Please click on Users Edit on 6001 users can see the configuration in the following picture especially pay attention to the configuration in the red ellipse frame Then when users want to listen to a message they can dial 6750 or the Mailbox 6001 How to realize the IVR IVR is Interactive Voice Response Voice Menus allow for more efficient routing of calls from incoming callers Also known as IVR menus or Digital Receptionist Upload Voice Menu Prompts If users want to configure the IVR which they need they must upload their voice prompt Users can click on Voice Menu Prompts users can see the screen like this screenshots Record a new Voice Menu prompt Upload a Voice Menu prompt No custom Voice Menu prompts found You can record a new Voicellenu Prompt by clicking on the Record a new Voice Menu prompt or click on the Upload a Voice Menu prompt button to upload a custom voice menu Users can click the button of Record a new Voice Menu prompt to record a voice prompt or users can click the button of Upload a Voice Menu prompt to upload their voice prompt www h
19. R ERRARE m iud 52 3 68 Server NOI6 S 5 idt ene Rr He eda Erb e ote obe be Mae Ease eh 53 Case of IP PBX 02104V08 00 0 nte nennen nnn 53 www houyuanhk com HOUYUAN How to Make Internal Calls through IP PBX 02V04V08 sse 54 Access to the Web Page of IP PBX 02 04 08 by Browser eee 54 Add up Users from Web Page of IP PBX 02Y0408 esee 54 Register a SIP user 6000 in IP PHONE entente 55 How to Communicate with Outside essent entente 56 How to Call VolP Trunk usen en ism atti epa kuxr iade nba ar nh edad vanes 58 Mec durile T 59 59 How to realize the IVR 20000001 M 61 Ring CAINS dC sane 61 nul 62 hints 64 MAUI 1 1 1 E EE 67 www houyuanhk com HOUYUAN 1 0 Introduction of IP PBX 02 04 08 1 1 IPPBX 02 04 08 The IP PBX 02 04 08 is a complete Asterisk Appliance with combination of FXO FXS
20. Ring Timeout Thrunk FXO devices must have a timeout nt 8000 to determine if there was a hangup before the line was answered www houyuanhk com HOUYUAN answeronpolarit yswitch polarity reversal will mark when a outgoing If this option is enabled the reception of aBoolean Call is answered by the remote party no hanguponpolarit In some countries a polarity reversal isBoolean yswitch used ta cinala tha diceannact a nhana lina Use CallerlD Enabling this option enabled Callerld Boolean yes detection Caller ID Start This option allows one to define the start ofComboBox Ring a CallerID Signal CallerID This option allows the lines to report the select box As Received Caller ID string as received from the telco or as a fixed value by using the custom option Pulse Dial If this option is enabled pulse modeBoolean No dialing instead of DTMF wil be enable CID Signalling This option defines the type of caller ID ComboBox Bell USA signaling to use bell v23 v23 jp or dtmf Flash Timing Flash Time defines the time inTextbox millseconds that is generated for a flash operation 750 Receive Flash Flash Time defines the time inTextbox Timing milliseconds that is generated for a flash operation 1250 1 Trunk name unique label to help users identify the trunk when listed in outgoing calling rules and incoming calling rules A VoIP se
21. Seconds Say message f this option is enabled the Caller ID of the partyCheck box Check Caller ID that left the message will be played back before the voicemail message begins playing Say messagelf this option is set the duration of the message inCheck box unCheck duration mintues will be played back before the voicemail message begins playing Play envelope Turn on off playing introductions about each Check box unCheck message when accessing them from the voicemail Allow users to Checking this option allows the caller to reviewCheck box Check review their message before it is submitted as a new voicemail message 1 Voice mail application Voicemail ARG u 2 Automatically generated configuration file etc asterisk voicemail conf mailbox_number and to register a name the name email where an 3 IPPBX Max m gt password name email user in sip conf or iax conf password the pass used to register a user in sip conf or iax conf which to be associated with the mailbox otification for the voicemail will come essages data 150M a Email Settings for Voice mails mailbox number the number you use in extension conf for VoiceMail command Send messageslf this option is set then voicemails will not beCheck box unCheck by e mail only heckable using a Phone Messages will be ent via e mail only Note You need to have an smtp www houyuanhk com HOUYUAN
22. age time 5 seconds v Playback Options Say message Caller ID Say message duration Play envelope Allow users to review Extension fordefines the extension that Users call in order toNO 6750 checking access their voicemail accounts messages Direct Check this to enable direct voicemail dial ForCheck box unCheck Voicemail Dial instance if John s extension is 6001 you would be able to directly dial into John s voicemailbox by dialing 6001 to leave him a message 41 HOUYUAN Max greeting Set the maximum number of seconds for a User sNo 30 in seconds voicemail greeting Dial 0 for Enable Callers to exit the voicemail application andCheck box Check Operator connect to an operator extension The operator extension must be defined from the Options panel Maximum This select box sets the maximum number of 10 25 100 25 messages permessages that a user may have in any of their200 500 10 folder folders 00 Max message This select box sets the maximum duration of a 1 minute 2 minutes time voicemail message in seconds Message recording 2 minutes 5 Will not occur for times greater than this amount minutes 15 minutes 30 minutes um limited Min messageThis select box sets the minimum duration of afno 1 seconds time voicemail message in seconds Messages belowminimum 1 this threshold will be automatically deleted Seconds 2 seconds 3 seconds 4 seconds 5
23. alk only mode Talk only no listening T set talker detectio v video mode w wait until the marked user enters the conference plays music on hold until marked user enters if M is used All other connected users will hear MusicOnHold until the marked user enters X allow user to exit the conference by entering a valid single digit extension of the context specified in MEETME EXIT CONTEXT or the current context if that variable is not defined x close the conference when last marked user exits 3 322 Follow Me If A calls B B does not answer the call will be transferred to C who is set up in follow me PBX Management system Edit User 6000 Status Enable Disable Music On Hold Class default v DialPlan DialPlani Destinations e001 20 seconds Status Enable Disable FollowMe for this user Choice Disable Music On Hold Music On Hold class that the caller would hearChoice Default Class while tracking the user www houyuanhk com HOUYUAN DialPlan DialPlan that would be used for dialing the FollowMe numbers By default this would be the same dialplan as that of the user Choice Destinations List of extensions numbers that would be dialed to reach the user during FollowMe Destinations New FollowMe Number Add a new FollowMe number which could be a Local Extension or an Outside Number The selected dialpl
24. an should have permissions to dial any outside numbers defined Dial Local Extension Dial Outside Number Dial Order This is the order in which the FollowMe destinations are dialed to reach the user Ring X after Trying previous extension nu m ber Ring along with previous extension nu m ber Ring after Trying previous extension n u mbe Follow me Option Playback the unreachable status message if we ve run out of steps to reach the or the callee has elected not to be reachable Check box Uncheck Playback the unreachable status message if we ve run out of steps to reach the or the callee has elected not to be reachable Check box Uncheck Playback the unreachable status message if we ve run out of steps to reach the or the callee has elected not to be reachable Check box Uncheck 1 General config file etc asterisk followme conf 3 4 System Steup 3 411 Configure Hardware In the configure hardware page it includes the following components analog hardware tone region advanced settings Pay attention that some browsers do not display the configure it is unimportant Analog Hardware When users boot the IP PBX 08 which will detect the FXO and FXS modules automatically www houyuanhk com 36 HOUYUAN the analog hardware component displays the modules which are detected correctly PBX Management system Digital Card Configuration Wizard
25. e front of the dialing string before the call is placed via the trunk selected in Use Trunk For example want users to dial 9 before their long distance calls however one does not dial 9 before those callsre placed onto analog lines and the PSTN so one should strip 1 digit from the front before the call is placed 22 www houyuanhk com HOUYUAN The way of outgoing calling Every time you dial a number asterisk will do the following in strict order Examine the number you dialed Compare the number with the pattern that you have defined in your first outgoing rule and if matches it will initiate the call using that trunk If it does not match it will compare the number with the pattern that you have defined in the second outgoing rule and so on Passthe number to the appropriate trunk to make the call 3 312 Incoming Calling Rule This is where the behavior of incoming calls from all trunks is being handled When an incoming call from PSTN or VoIP trunk is received asterisk needs to know where to direct it It can be directed to a ring group an extension digital receptionist voice menu or queue For this purpose Incoming Calling Rules need to be set up PBX Management system New Incoming Rule Trunk FXO Trunki Time Interval None no TimeIntervals matched Pattem 5 Destination Operator Name Description Type default runk Choice the trunk for the incoming rule analog serv
26. enEmpty Strict Periodic Announcement if u know ext dia Frequency Sec 0 JoinEmpty No 1 Enable Exit to Hold TimeOut Agents 4 6000 6001 MaxLen 0 P Report Hold Time 71 T Members 7 0 7 1Ax2 6000 SIP 6001 711AX2 6001 DAHDU2 DENIS Description Type default Extension Extension for call queue may be dialed to reach Int 6500 the call queue Name Name for call queue Str Strategy Strategy this option sets the ringing strategy for ringall Ro ring all this queue the options are undrobin 1 Ring all ring all available agents simultaneously until one answers astrecent 2 RoundRbin Take turns ringing each available F ewest agent calls Rand 3 LeastRecent Ring the agent which was least om Rrmo recently called m ery 4 FewestCalls Ring the agent with the fewest completed calls 5 Random Ring a Random agent 6 RRmemory RoundRobin with Memoryn Music On Select the Music on Hold Class for this Queue Choice default Hold Music on Hold classes can be managed from the the Music On Hold panel on the left www houyuanhk com HOUYUAN LeaveWhen Empty This option controls whether callers already on hold are forced out of a queue that has no agents There are three options Yes Callers are forced out of a queue when no agents are logged in No Callers will remain in a queue with no agents Strict Callers are forced out of a queue with n
27. ent at a time the Queue Will complete as many calls simultaneously to the checkbox Auto pause Enabling this option pauses an agent if they fail to answer a call This means that the agent is still logged into the queue but they will not receive calls from the queue Once paused an agent can unpause by logging into the queue using the regular agent checkbox Report Hold Time Enabling this option causes Asterisk to report to the Agent the hold time of the caller before the caller is connected to the Agent checkbox KeyPress Events If a caller presses a key while waiting in the queue this setting selects which voice menu should process the key press choice Agent This selection shows all Users defined as Agents in their User conf Checking a User here makes them a member of the current Queue checkbox www houyuanhk com HOUYUAN 1 Call queue application Queue EXTEN 2 Change agents status Login Login out agents in System Info 3 Hear the music if all agents are busy until non conversation busy 3 318 Voice Menu prompts This component is used for recording custom voice menu PBX Management system Record a new Voice Menu prompt File Name RMUSICK GSM Dial this User Extension to record a new voice prompt 6000 Cancel Recora menulFile Name Str RMUSIC dial this User Extension to record a ne
28. er provider Voip Time Interval Choice the time interval for the incoming rule Choice Non timeinterv al Pattern Pattern of the incoming rule Dialplan S matched Destination Incoming to destination users voicelVR mail ring group www houyuanhk com HOUYUAN 1 A trunk support a number of this time intervals to support a number of Destination 2 Pattern All patterns are prefixed by the character In patterns some characters have special meanings X Any Digit from 0 9 Z Any Digit from 1 9 N Any Digit from 2 9 12345 9 Any Digit in the brackets in this example 1 2 3 4 5 6 7 8 9 Wildcard matches anything remaining i e 9011 Matches anything starting with 9011 excluding 9011 itself Wildcard causes the matching process to complete as soon as it can unambiguously determine that no other matches are possible For example the extension NXXXXXX would match normal 7 digit dialings while AINXXNXXXXX would represent a three digit area code plus phone number proceeded by a one 3 Note users will most likely need to add a rule with the pattern s without the quotation marks for each trunk This signifies catch all meaning all calls with a DID not matching any other rules will match this If users have multiple SIP trunks from the same provider they will want to set this pattern to whatever you specified as Contact Extension 1 313Dial Plan A
29. gGroup Extensions 6400 Queue Extensions 6500 VoiceMail Group Extensions 6600 Fax2email Extensions 6701 DENIS Description Type default Global OutBound This is default global CallerlD that is used for all outgoing Int CID calls when no other CallerlD is defined that has a higher priority gt When making outgoing calls the following rules are used to determine which CallerlD will be used if they exist The first CallerlD used is a CallerlD set for the user making the call defined in the Users tab The second CallerlD is the one that is set in the VoIP Trunks configuration if applicable The last CallerlD used for outgoing calls is the Global CID defined in the Options tab Operator The Operator Extension is the extension which will Chioce Extension be dialed when a caller presses 0 to exit Voicemail It is also available as a Voice Menu option Ring Timeout Number of seconds to ring a device before sending Time 20 to the user s Voicemail Box Call Record Dir Call Record Dir Str tmp Call Record Call Record Format Choice FXO FXS Format www houyuanhk com _ dissi o HOUYUAN Extension User Extensions Int 6001 629 preferences 9 Conference Extensions Int 6300 639 9 VoiceMenu Extensions Int 7001 710 0 RingGroup Extensions Int 6400 649 9 Queue Extensions Int 6500 659 9 VoiceMail Group Extensions Int 6600 669 9 Resert to default p sult Languages he Language setting all
30. hannel Transfer ransfer channel 3 6 Admin 3 61 CDR Viewer PBX Management system CDR Files Management List of CDR Files 1 www houyuanhk com HOUYUAN 3 62 IAX Setting PBX Management system IAX Inter Asterisk Exchange Protocol Configuration General Preferences Jitter Buffer Registration Codecs BindPort 4569 Bind Address IAX1 Compatibility 7 No Checksums F Delay Rejed ADSI 2 Music On Hold Interpret default Music On Hold Suggest Language en Bandwidth low lena 9 Call Token Optional 0 0 0 0 0 0 0 0 3 63 SIP Setting PBX Management system SIP Session Intitation Protocol Configuration General Preferences TOS Debug Notify NAT Misc Jitter Buffer Codecs Always auth reject default Realm for digest authentication asterisk UDP Porto bindto 5060 IP address to bind to 0 0 0 0 Domain Allow guest calls Overlap dialing support Allow Transfers Enable DNS SRV lookups on outbound calls Pedantic F Always auth reject v SIP Domain Support From Domain Auto Domain Allow External Domains www houyuanhk com HOUYUAN 3 64 File Edit PBX Management system File Editor users conf users conf fullname userbase 6000 hasvoicemail yes vmsecret 1234 hassip yes hasiax yes hasmanager no callwaiting yes threewaycalling yes callwaitingcallerid yes transfer yes canpark yes cancallforward yes callreturn yes callgroup 1 pickupgroup 1 3 65
31. ion list displays green means the user is Free when the light of Extension list displays orange means the user is Ringing when the light of Extension list displays red means the user is Busy 2 Status This parameter displays if other users leave messages Messages 0 0 the figure front of displays the new messages amount the figure behind of displays the old messages amount www houyuanhk com HOUYUAN 3 2 Extensions PBX Management system Create New User General Extension 0 Name DialPlan Internal CalleriD 0 External CallerlD 7 Enable Voicemail for this User Access PIN code Mailbox 6000 Email Address Technology sip V tax V P Call Token Required E Analog Station None flash 750 ndiash 1250 Codecs First a law Second u law Third GSM Fourth None Fifth None VolP Settings Qualify NAT V Can Reinvite E DTMF Mode RFC2833 e insecure no e SIPAAX Password Other Options 3 Way Calling Elin Directory Ei Cali waiting Ecen Eis Agent Pickup Group 1 e cance update Users component is used to add or remove Analog SIP IAX extension Click on Create New User button in the web of IP PBX 02 04 08 users can create SIP IAX User and Analog EUIS Description Type Default Extension The numbered extension Textbox 6001 Name A character based name for this user Textbox Null DialPlan DialPlans are sets of
32. l which trunk and what dial pattern the call used are configured in outgoing calling rules Please select the Outgoing Calling Rules option then Click on New Calling Rule button the parameters of the Outgoing Calling Rules are in the following table PBX Management system New CallingRule Calling Rule Name OuttoPSTN Pattern senato Local Destination Destination Send this call through trunk Use Trunk FXO Trunk v Record Calls Strip 0 digits from front and Prepend these digits before dialing Use FailOver Trunk fail over Trunk iy Stip digits from front and Prepend these digits before dialing www houyuanhk com HOUYUAN EUIS Description Type Default Calling Rule Name The name of the Calling rule Textbox Null Pattern The dialing rule Textbox Null Send to Local If this option is checked and Destination selected no select Destination is defined calls matching the specified pattern Destination Choose the Local ComboBox Null Destionation User VoiceMenu Hungup Use trunk Defines the Trunk that calls matching the ComboBox Null specified pattern will be placed through Strip Allows the user to specify the number of Textbox Null digits that will be stripped from the front of the dialing string before the call is placed via the Prepend Allows the user to specify digits that are Textbox Null these digits prepended before the call is placed via the tru
33. les users can configure it like this www houyuanhk com 60 HOUYUAN PBK Management system New Incoming Rule Trunk FXO Trunki Le Time interval None no Tuneintervals matched w Patiern Destinaton yoicemenu ivr Then when others call you through the analog1 they can here the IVR and do the operation which they need Conference In order to realize the conference option the users which will attend to the conference must have registered Here the writer uses 6001 6002 6008 Now please click Conferencing New conference Bridge users can see the screen like the following screenshots PBX Management system New Conference Bridge Extension 6300 2 Marked Admin user Extension 9989 Password Options Pin Code 000 P Admin PinCode 123 Conference Room Options J Play hold musicforfirstcaller 7 Close conference when last marked user exits J Enable caller menu Announce callers J gt Quiet Mode J Wait for marked user Update Then please click on Update button and click on Apply Changes button in up right corner of the main page Here the writer configures it like the screenshots above Then 6001 dial 6300 and input Pin Code Users can hear a voice promt and wait others then you can hear the music 6002 does the same operation 6008 dial 9989 and input Admin PinCode Now all the users are in the conference Ring Groups Define Ring groups to dial more than one extension simultaneously or to ri
34. lick on Save button and click on Apply Changes button in up right corner of the main page www houyuanhk com HOUYUAN Next Add up SIP user 6000 After logging into the web page of IP PBX 02 04 08 please click on Users Create New User the writer configure user 6000 like the following PBX Management system Create New User General Extension 0 Name DialPlan Internal CalleriD 0 External CalleriD Enable Voicemail for this User Access PIN code Mailbox 6000 Email Address Technology sip V tax Call Token Required Analog Station None flash 750 ndlash 1250 Codecs First a law Second u law Third GSM Fourth None Fifth None e VoIP Settings Qualify NAT V Can Reinite C DTMF Mode RrC2833 insecure no SIPAAX Password Other Options 3 Way Calling Elin Directory El Call Waiting Ecen Elis Agent Pickup Group 1 e Update At last please click on Update button and click on Apply Changes button in up right corner of the main page Register a SIP user 6000 in IP PHONE After logging into the web page of IP Phone IP PHONE please select VOIP option Enable this SIP account User Details Display Name Fdwin User name 6000 Password III Authorization user name 6000 Domain 192 168 1 167 After configuring please click on the APPLY button Users can see the Register status is Registered if user do not register successf
35. lling channel 3 Allow keypress events Must be voice menus have application Background file e xBackground a music when keypress events 4 Advance edit Change dialplan for voice menus e x include z default exten s 1 NoOp Incoming DID exten s 2 Answer exten s 3 Background record GreetingNew exten s 4 Background record Make YourSelection exten s 5 Background fpm sunshine exten s 8 Voicemail 6002 u exten 1 1 Goto voicemenu custom 2 s 1 exten 2 1 Voicemail 6002 u exten 5 1 Goto voicemenu custom 3 s 1 Want to control music on hold play time include default exten s 1 NoOp Incoming DID exten s 2 Answer exten s 3 Background record GreetingNew exten s 4 Background record MakeYourSelection exten s 5 Set TIMEOUT absolute 8 exten s 6 Background fpm sunshine exten s 7 Set TIMEOUT absolute 60 exten s 8 Voicemail 6002 u exten 1 1 Goto voicemenu custom 2 s 1 exten 2 1 Voicemail 6002 u exten 5 1 Goto voicemenu custom 3 s 1 3 320 Time Intervals Time Intervals defines ranges of working time that will be used by call routing features Please select the Time Intervals option from the vertical menu on the left of the main page www houyuanhk com HOUYUAN PBX Management system New Time Interval Time Interval Name TimelInterval 9 By day of week Tue to Sun By Days of a Month Date Month Time Entire Day Start Time End Time
36. lplan After logging into the web page of IP PBX 02 04 08 please click on Dial Plans Edit DialPlan1 Next Create a User After logging into the web page of IP PBX 02 04 08 please click on Users Create New User the writer configure user 6000 like the following T www houyuanhk com HOUYUAN PBX Management system Create New User General Extension 6000 Name DialPlan Internal CalleriD 0 External CalleriD Enable Voicemail for this User Access PIN code Mailbox 6000 Email Address Technology SIP V tax Call Token Required Analog Station None e fash 750 ndash 1250 Codecs First a law Second u law Third GSM Fourth None Fifth None VoIP Settings Qualify 2 NATE Reinite 2 DTMF Mode RFC2833 v insecure no SIPMAX Password Other Options 3 Way Calling In Directory Call Waiting 2 cn Is Agent Pickup Group 1 z Cancet Lpaate At last please click on Update button and click on Apply Changes button in up right corner of the main page Next Create an Outgoing Calling Rule PBX Management system New Incoming Rule Trunk FXO Trunki Time Interval None no TimeIntervals matched Pattern Destination Operator Update At last please click on Update button and click on Apply Changes button in up right corner of the main page Here the users use the first channel Then when the
37. n password in order to continue dialplan execution Background Play an audio file while waiting for digits of an extension to go to d Busy Tone Indicate the Busy condition Congestion Indicate the congestion condition to the calling channel f Digit Timeout set digit timeout DISA Password Allow someone from outside the telephone switch PBX to obtain an internal system dialtone and to place calls from it as if they were placing a call from within the switch Response Timeout set response timeout Macro macronamelarg1 arg2 Executes macro lt macroname gt Play Sound Plays back given file k Ringing Indicate ringing tone Set MusicOhHold Class select a music on hold SayAlpha Say each character in the string including letters numbers and other characters one by one a macro using the context 3l HOUYUAN k SayDigits Say the digits one by one l SayNumber Say a number e g six thousand five hundred and seventy two m Wait Pause dialplan execution for a specified number of seconds n WaitExten Wait for the user to enter a new extension for a specified number of seconds r To Destination go to destination 0 Set Language set language English Spanish French p To Directory go to directory q Dial an external Number Place a call outside the pbx using the selected trunk v AGI Executes an AGI compliant application r User Event Send an arbitrary event to the manager interface x Hangup Hang up the ca
38. n you can see the following screenshots Click the button of Login so that all the Agents have logined Then refresh the web users can see the page that all the agents have logined like the following screenshots Create a Call Queue Please click Call Queues Create New Queue then users can configure the options like this screenshots www houyuanhk com HOUYUAN PBX Management system New Queue Extension 6500 Name Qcalls Strategy ringall Music On Hold default LeaveWhenEmpty strict JoinEmpty No Hold TimeOut Agents 4 6000 V 6001 Queue Options TimeOut 15 Wrapup Time 15 MaxLen 0 Auto Fill El Auto Pause 1 Report Hold Time 1 KeyPress Events None 7 Enable initial Anouncement if U Know ext dia Wait Before 2 WaitAfter Periodic Announcement if u know ext dia Frequency Sec 30 Enable Exit to Members SIP 6000 V AX2 6000 J SIP 6001 LJ 1AX2 6001 DAHDI 2 Then 6000 have registered can call 6500 then 6001 6000 are all ringing together Acronyms www houyuanhk com HOUYUAN VoIP Voice over Internet Protocol FXO Foreign eXchange Office interface is the port that receives the analog line FXS Foreign eXchange Subscriber interface is the port that actually delivers the analog line to the subscriber SIP Session Initiation Protocol SIP is a signalling protocol used for establishing sessions in
39. ncy systems are used for signaling internal to the telephone network DHCP Dynamic Host Configuration Protocol DHCP is an auto configuration protocol used on IP networks DHCP allows a computer to be configured automatically eliminating the need for intervention by a network administrator It also provides a central database for keeping track of computers that have been connected to the network This prevents two computers from accidentally being configured with the same IP address NTP Network Time Procotol NTP is a protocol for synchronizing the clocks of computer systems over packet switched variable latency data networks It is designed particularly to resist the effects of variable latency by using a jitter buffer Vlan Virtual Local Area Network is a group of hosts with a common set of requirements that communicate as if they were attached to the same broadcast domain regardless of their physical location A VLAN has the same attributes as a physical LAN but it allows for end stations to be grouped together even if they are not located on the same network switch Network reconfiguration can be done through software instead of physically relocating devices www houyuanhk com HOUYUAN HTTP Hypertext Transfer Protocol The HTTP is a networking protocol for distributed collaborative hypermedia information systems HTTP is the foundation of data communication for the World Wide Web HTTP functions as a request response protocol in
40. ng more than one phone sequentially This feature may also be called Hunt groups Users can click Ring Groups New Ring Group then users can configure it like the following screenshots Of course 6001 61 www houyuanhk com HOUYUAN 6000 have registered Then 6000 dial 6400 you can hear 6001 6000 are ringing simultaneously If users want the phones are ringing sequentially they can configure the strategy as Ring in Order PBX Management system New RingGroup RingGroup Name Extension for this ring group 6400 Ring Group Members Available Users 6000 SIP 6001 SIP 6000 IAX2 6001 IAX2 Ring Group Options Strategy Ring in Order Seconds to ring each member 39 If not answered Goto Hangup Agents You need complete the following two steps when you need the function of Agents www houyuanhk com HOUYUAN Create Users as Agents Create New User General Extension 6000 Name DialPtan intemal Caller 6000 External CalleriD 1asd for Inss User Access PIN code Mailbox 000 Email Address Call Token Required Analog Stadon None Te flash 750 nfiash Second u taw m Third GSM iw Fourth None e Fifth None Qualify 7 NAT 7 Can Rewnite OTMF Mode RFC2833 insecure no w SIPAAX Password ler Opeons 3 Way Calling in Directory waang icn gt is Agent Pickup Group 1 e Update Like this have also created 6002 6008 1 you must click System Status the
41. njunctionComboBox homal with the Low Power Option allows the user to increase the ringing speed to 25HZ lowpower This option generally used in conjunctionComboBox nomal with the Fast Ringer Option allows the user to set the peak voltage during Fast Ringer Operation to 50V ring detect This option allows the user to choose from ComboBox standard normal ring detection or a full wave tian MWI mode This option allows the user to specify theComboBox none type of Message Waiting indicatori detection to be done on trunk FXO interfaces 3 412 Configure trunks To receive calls from PSTN and make calls to the outside world users have to use trunks Please select the Trunks option from the vertical menu on the left of the main page PBX Management system Manage Analog trunks New Analog Trunk Analog Trunks VOIP Trunks T1 E1 BRI Trunks Analog trunk is associated with FXO port and it will call outside by PSTN line Click on New Analog Trunk then users can see the parameters which are in the following table in the web Channels Display the FXO or FXO FXS modules __ selected no select Trunk Name The name you want to set for the trunk Textbox null Busy Detection Busy detection is used to detect far endBoolean hang up or for detecting busy signal Yes busycount If Busy Detection is enabled it is alsolnt 3 possible to specify how many busy tones to wait for before hanging up
42. nk If a user s trunk required 10 digit dialing but users were more comfortable performing 7 digit dialing this field could be used to prepend a 3 digit area code to all 7 digit strings before they are placed to the trunk User may also prepend a w character for analog Use Failover Failover trunks can be used to make sure selected no select Trunk that a call goes through an alternate route when the primary trunk is busy or down If Use Failover Trunk is checked and Failover trunk is defined then calls that cannot be placed via the regular trunk may have a secondary trunk defined Ifa user s primary trunk is a VoIP trunk but one wants calls to use the PSTN when the VoIP trunk isn t available this option Fail over trunk Choose the trunk ComboBox ComboBox Pattern X Any Digit from 0 9 Z Any Digit from 1 9 N Any Digit from 2 9 12345 9 Any Digit in the brackets in this example 1 2 3 4 5 6 7 8 9 Wildcard matches anything remaining i e 9011 Matches anything starting with 9011 excluding 9011 itself Wildcard causes the matching process to complete as soon as it can unambiguously determine that no other matches are possible For example the extension NXXXXXX would match normal 7 digit dialings while _1NXXNXXXXX would represent a three digit area code plus phone number proceeded by a one Strip Allows the user to specify the number of digits that will be stripped from th
43. o agents logged in or if all logged in agents are unavailable The default option is Strict After a caller has left the queue a caller will hear a busy tone and advance to the next calling rule after attempting to enter the queue yes strict No strict JoinEmpty This option controls whether callers can join a call queue that has no agents There are three options Yes Callers can join a call queue with no agents or only unavailable agents No Callers cannot join a queue with no agents Strict Callers cannot join a queue with no agents or if all agents are unavailable yes strict No no the Queue tries to ring the next Agent TimeOut How many seconds an Agent s phone will ring before Time 15 Wrapup Time Agent will have before the Queue can ring them with a new call The default is 0 which is no delay How many seconds after the completion of a call anTime Max Len How many calls can be queued at once This count does not include calls that have been connected with Agents it only includes calls that have not yet been connected Default is 0 which is no limit When the limit has been reached a caller will hear a busy tone land advance to the next calling rule after attempting to enter the queue Int Auto full Defining this option causes the Queue when multiple calls are in it at the same time to push them to Agents simultaneously Thus instead of completing one call to an Ag
44. oming INVITE messages 3 413 SMTP Setting PBX Management system SMTP Settings For Emails Smtp server smtp gmail com Pot 25 Use TLS Authentication V Username oxxx Password eeees STMP server es IP address or hostname of an SMTP serverStr that your box may connect to without authentication in order to send e mail notifications of your voicemails i e Port The port number on which the SMTP server isStr running generally port 25 Use TLS Use TLS Transport Layer Security wheniCheck box unCheck communicating with the SMTP server Authentication Does the SMTP Server requite authentication Check box junCheck Username The username of a valid account on the STMPStr Password The password of a valid account on the STMPStr 1 Config file etc ssmtp ssmtp conf 2 Note Firmware after that starts support Gmail 3 414 Voicemail Setting When users call someone who does not answer the call users can leave a voice message for the called party if the called party supports voice mail 40 www houyuanhk com HOUYUAN PBX Management system Extensions General Voice Mail Settings General Settings Email Settings for Voicellails Extension for checking messages Direct Voicemail Dial Max greeting in seconds Dial 0 for Operator Message Options Maximum messages per folder 25 Max message time 2 minutes e Min mess
45. ot from SIP endpoints through itself selecte Enabling this option d causes asterisk to attempt to negotiate the endpoints to route the media stream directly bypassing asterisk It is not always DTMF Mode Set default dtmfmode for sending DTMF info SIP ComboB rfc2833 INFO messages inband Inband audio requires Ox 64 kbit codec alaw ulaw auto Use rfc2833 if offered 3 Way Calling Check this option if the User or Phone should have selected Not select 3 Way Calling capability In Directory Check this option if the user is to be listed in the selected Not select system telephone directory Call Waiting Check this option if the User or Phone should have selected Not select Call Waiting capability Is Agent Check this option if this User or Phone is a Call selected Not Queue selected Member Agent Pickup Group If a user called A and another user called B in the selected Not SEAMS selected group A can pick up the phone taking the place of 1 Analog Station When users want to create Analog Users please choose the FXS ports 2 Codec Preference Support g711u law g711a law g729 FXO FXS www houyuanhk com HOUYUAN 3 Attension in the textbox of Extension the value users set is limited to a range they can adjust the range in the Options option to meet their requirement 3 3 PBX features 3 311 Outgoing Calling Rule Outgoing calling rules is used to route an outgoing call when users make an external cal
46. outside makes a incoming call it will be sent to user 6000 through the first channel Of course users can communicate with other use FXO FXS by wireless For example The writer uses the channel 1 and the number is 158xxxxxxx2 Incoming Calling Rules be pointed to 6000 Then The writer can dial a mobile phone number with prefix 5 others can dial 158 xxxxxxx 2 to connect us www houyuanhk com HOUYUAN How to Call through VoIP Trunk Call from IP PBX 08 to IP PBX 08 In order to call from IP PBX 08 to IP PBX 08 The writer will create a user in IP PBX 08 for the SIP IAX trunk in IP PBX 08 create a SIP IAX trunk an outgoing call rule and a dial plan in IP PBX 08 But pay a attention that at the same time a port of the router where the IP PBX 08 in must be directed to the IP PBX 08 1 Add an user 6200 it will be used as SIP trunk in IP PBX 08 in IP PBX 08 Then Add a user 6030 in IP PBX 08 for IP PHONE the way is the same as adding 6001 2 Add a VoIP trunk in IP PBX 08 Type S iv Provider Name outsidePBX Username 6200 Fromuser Fromdomain Password 6200 Contact Ext S Insecure Type very Cancel amp Aad 3 Create an outgoing calling rule in IP PBX 08 Calling Rule Name outsidePBX Patem 5 Destnaton Use Trunk 1 FXO Trunki wj Record Calls Strip 1 igs from front and Prepend these dits betore dealing ver Trunk fail over Trunk Iv Strip pits from tront and
47. ouyuanhk com HOUYUAN Create Voice Menu PBX Management system Create New VoiceMenu Advanced Edit General Key Press Events Name vm Extension 7000 Allow Dialing Other Extensions Actions Answer the call Play if u know ext dial amp Donot Listen for KeyPress events Selected the option Background on the Add new step then click the Add new step Users can see the screen display like the following screenshots then select their own voice prompt Here the writer use the voice prompt named 04 Users can upload the voice prompt Add new Step Play Sound a t Plays back gt 04 D Allow Key for am 2 for pm 1 yes 2 no CHANGES asterisk core en 1 4 13 CHANGES asterisk core en 1 4 19 CHANGES aster sk extra en 1 4 8 CREDITS asterisk core en 1 4 13 Hook on the option Allow KeyPress Envents then users can configure the operation from 0 to which their need Please click on save button and click on Apply Changes button in up right corner of the main page Here the writer configures that press 0 then call 6001 press 1 then call 6002 press 2 then call 6008 Of course 6001 6002 6008 have registered W Allow KeyPress Events Goto User 600 Goto User 6002 Goto User 6003 Add Incoming Calling Rules After configure the Voice Menu users must configure the Incoming Calling Rules Click Incoming Calling Rules New Incoming Calling Ru
48. ows the user to specify the Chioce English default prompts language for phone to phone inbound and outbound calls If a soundpack selection is made but not already installed then the pack will be downloaded from Digium Change Password Enter New Password Retype New Password Str Factory reset Factory reset Reset to defaults include network settings Reset to defaults but keep network settings 47 HOUYUAN 3 418 IP table Firewall PBX Management system Add New Rule Rule Type Normal Action ACCEPT Direction INPUT Protocol NONE S Port Source Destination 3 419 Backup Backup and Restore are two of the mandatory functions of any application IP PBX 02 04 08 is no exception Customers can backup all the files under the etc asterisk directory and restore them PBX Management system Backup Restore Configurations List of Previous Configuration Backups backup 2012jul05 180242 Jul 05 2012 Backup Create new backup Download from Unit Restore Previous config 3 5 Diagnostics 3 51 Active Channels The channels which are in communication status will be displayed in this component www houyuanhk com PBX Management system Channel Management Refreshing Active Channels in 7 Seconds No Channels Open Status Upload message for asterisk channels Hangup hang up c
49. port access by RS232 console cable In order to access to IP PBX 02 04 08 by the first three ways Users have to check that if your network connection between IP PBX 02 04 08 and PC is OK If it does not connect between IP PBX 02 04 08 and PC users can try to use the last way to access to IP PBX 02 04 08 and change the IP address for IP PBX 02 04 08 LIII 8 www houyuanhk com HOUYUAN 2 11 Web WEB URL 192 168 1 167 Username admin Default Password admin PBX Configuration Engine mme Ssword eeeee EH 2 12 SSh WEB URL 192 168 1 167 Username admin Default Password uClinux Logging 3 Teminal Keyboard Bell Features Window Appearance Behaviour Translation Selection Colours Connection Data Proxy Telnet Rlogin SSH Serial www houyuanhk com Basic options for your PuTTY session Specify the destination you want to connect to or IP address Port 192 168 1 167 22 Connection E Raw Telnet Rlogn SSH Serial Load save or delete a stored session Saved Sessions Default Settings WinSCP temporary session Close window on exit Always Never Only on clean exit Open HOUYUAN 2 13 RS232 Console Post or minicom 1 Connect the console port of IP PBX 02 04 08 to your PC s console port with RS232 console cable II Wo p r i 2 Run the HyperTerminal and set up the console port like the following Bits per second 115200 Data bits 8
50. roup s Extension Str User MailBoxes The entire user Mailboxes Check boxs www houyuanhk com HOUYUAN 3 316 Music on Hold Music On Hold need users customize audio tracks for different queues parked calls etc PBX Management system Manage Music On Hold Classes Upload MOH File Delete Delete Selected default Manage Music On Hold class default No files found in this class Use the above upload form to add music files to this class Upload an 8 KHz Support codec g711a g711u Upload Mono Music file New music on hold Add a new music on hold 2 Music on hold Dir persistent sounds moh 3 Sounds LICENSE asterisk moh freeplay ulaw LICENSE asterisk moh freeplay ulaw fpm world mix ulaw fpm world mix alaw fpm sunshine ulaw fpm sunshine alaw fpm calm river ulaw fpm calm river ulaw 4 Music on hold after holding status Status busy 5 Music on hold non rtp stream www houyuanhk com HOUYUAN 3 317 Call Queues Please select the Call Queues option from the vertical menu on the left of the main page then users can get the following screen PBX Management system New Queue Queue Options Extension 6500 TimeOut 15 Wrapup Time 15 Auto Fill 1 Auto Pause KeyPress Events None Name QCalls Strate Il Sy 5 Enable initial Anouncement Music On Hold default Wait Before 2 WaitAfter LeaveWh
51. rvice provider VSP that users have signed up with is also a trunk Via the VoIP trunk users can dial via the VoIP service to reduce their cost when making international calls Users can set up the VoIP trunk to make calls to the PSTN or other VoIP network Users also can use the VoIP trunk to link headquarter and branch offices for free internal calls Click on New SIP IAX Trunk the following table is the parameter of VoIP trunk x You can select SIP or IAX type to meet your ComboBox SIP need Provider Name A unique label to help you identify this trunkTextbox Null when listed in outbound rules incoming rules etc Hostname The IP Address of the server which you want Textbox Null to connect Username the username that your service provider Textbox Null configured Fromdomain The domain of the server which you want to Textbox Null connect www houyuanhk com HOUYUAN Password the password that your service provider Textbox Null configured for the user Contact Ext Textbox S Insecure Type The insecure type of the trunk transferringComboBox very data 1 Notice Provider Name must be unique label especially do not the same with Username 2 Insecure Type insecure very To allow registered hosts to call without re authenticating insecure port Allow matching of peer by IP address without matching port number insecure invite removes the requirement for authentication of inc
52. s is the number dialed to reach this nt 6300 Marked Admin user Extension If the conference bridge is to have marked users or admin users then those users should enter the conference bridge using a separate extension Admin conference users can lock and unlock the conference and can kick the most recent conference participant Marked users are special users whose entrance and exit if the Wait for Marked user or Close conference when last marked user exits can either begin or end the conference altogether Int Pin Code set an optional pin code Ex 1234 that must be entered in order to access the Conference Admin PinCode Defining this option sets a PIN for Conference Administrators user each other until the marked user has joined Play music for theChecking this option causes Asterisk to playCheck box unCheck first caller Hold Music to the first user in a conference until another user has joined the same Close conference Close the conference bridge when the lastCheck box unCheck for the list callermarked user logs out of the conference call exit Enable call menu Checking this option allows a user to accessCheck box unCheck the Conference Bridge menu by pressing the Asterisk key on their dialpad Announces Checking this option announces to all BridgeCheck box unCheck callers participants the joining of any other articipants Quiet mode Do not play enter leave sounds
53. sed to U law The result is a less fuzzy sound as sampling artifacts are better supressed Pick up the ability to pull a ringing call to the phone you are currently on There are two main types a Group call pickup this allows you to collect a call from any ringing phone that is in the same pickup group as you if there were more than one phone ringing then you would have no control over which call you collected b Directed pickup this allows you to pickup a call at a specific extension maybe you re in another office and you hear a phone ringing and wonder if it s yours You dial the pickup number and your extension and the call will only transfer if it is your extension Group call pickup is typically invoked by dialing 8 or 8 from another phone in the call pickup group Syslog Syslog is a standard for logging program messages It allows separation of the software that generates messages from the system that stores them and the software that reports and analyzes them It also provides devices which would otherwise be unable to communicate a means to notify administrators of problems or performance Time Zone A Time Zone is a region on Earth more or less bounded by lines of longitude that has a uniform legally mandated standard time usually referred to as the local time www houyuanhk com
54. t answered Goto Hangup Ring Group Ring group name use in pbx Str Name Extension for thisRing group No dial the No if you want to join Int 6400 ring group change boundary value in options Ring Group The ring group of numbers EXT1 EX Members T2 EXT3 aoo Available The entire Users EXT1 EX Users T2 EXT3 boc Strategy Ring all simultaneously Ring in order Ring inRing Order tingin all Order Extensions Seconds to ringSeconds to ring each member Time 20 each member www houyuanhk com HOUYUAN If not If not answered go to hang up hang up the calling Hang up answered Goto channel Operator Go to operator Extension a call to user Voicemail Go to IVR Conference conference Call queue Go to a call queue join a Operator Hang up 1 ring group application Dial channel type EXTEN channel type EXTEN 20 i 2 ring group up after please a call 3 non ring if ring group user off hook or non user registered 4 only one man can connected in coming call 3 315 VoiceMail Groups Define Voice Mail extension Groupsto leave a voicemail message for a group of users by dialing a New Voice Mail Group PBX Management system VoiceMail Group s Extension 6600 Name Vgroup User MailBoxes 7 6000 7 6001 MoiceMail Group sDefault Voicemail Group s Extension Int 6601 Extension Label The label of Voicemail G
55. t version of the protocol to be widely deployed NAT Network Address Translation DTMF Dual Tone Multi Frequency FXO FXS Global System for Mobile Communications www houyuanhk com Glossary Zaptel Zaptel refers to Jim Dixon s open computer telephony hardware driver API Zaptel drivers were first released for BSD and Jim s Tormenta series of DIY T1 interface cards Digium later produced interface cards from Jim s designs and improved the Zaptel drivers on the Linux platform Digium then added further drivers also following the Zaptel API for other telephony hardware Asterisk Asterisk is a software implementation of a telephone private branch exchange PBX originally created in 1999 by Mark Spencer of Digium Like any PBX it allows attached telephones to make calls to one another and to connect to other telephone services including the public switched telephone network PSTN and Voice over Internet Protocol VoIP services Voice Codec G 711 is a high bit rate 64 Kbps ITU standard codec It is the native language of the modern digital telephone network There are two versions A law and U law G 711 A law is indigenous to the E1 standard used in the rest of the world G 711 U law is indigenous to the T1 standard used in North America and Japan The difference is in the method the analog signal being sampled In both schemes the signal is not sampled linearly but in a logarithmic fashion A law provides more dynamic range as oppo
56. ully please pay attention to the Password in the red ellipse frame which must be the same with the SIP IAX Password of the user 6001 in IP PBX 02 04 08 Now users can call each other directly between user 6001 6002 and 6008 www houyuanhk com HOUYUAN How to Communicate with Outside In order to communicate with outside by IP 282 02104108 users need an analog trunk an outgoing calling rule a dial plan a incoming calling rule and a user Here the writer will give the simple configuration steps which show how to make a call to outside First Create an Analog Trunk After logging into the web page of IP PBX 02 04 08 please click on Trunks Analog Trunks Click New Analog Trunk And click Save the writer configure an analog trunk like the following PBX Management system Manage Analog trunks Analog Trunks VOIP Trunks T1 E1 BRI Trunks Next Create an Outgoing Calling Rule PBX Management system New CallingRule Calling Rule Name OuttoPSTN Pattem 9 send to Local Destination Destination Send this call through trunk Use Trunk FXO Trunki Record Calls Stip 0 digits from front and Prepend these digits before dialing J Use FailOver Trunk fail over Trunk Ew Stip digits from front and Prepend these digits before dialing At last please click on Save button and click on Apply Changes button in up right corner of the main page Next Add the Rule to Dia
57. w voice Choice 6001 Voice codes Choice 3 319 VoiceMenus Like most organization users would like to redirect all of the incoming calls automatically The voice menu is very handy for these sorts of things The system should allow callers to make the selection according to the voice menu www houyuanhk com HOUYUAN PBX Management system Create New VoiceMenu General Key Press Events Name vm Extension 7000 P Allow Dialing Other Extensions Actions Answer the call Play if u know ext dial amp Donot Listen for KeyPress events Name A name for the voice menus Str Extension If you want this Voicemenu to be accessible byNo 7001 dialing an extension then enter that extension number Actions A sequence of actions performed when a call entersDial plan the menu script Add new Step Add additional steps performed during the menu Dial plan script Allow Allow key press events will cause the system to listencheckbox KeyPress for DTMF input from the caller and define the actions Events that occur when a user presses the corresponding i Advance edit Advance edit for the voice menu Dial plan script 1 Menus allow for more efficient routing of calls from incoming callers Also Known as IVR Interactive Voice Response menus or Digital Receptionist 2 Step a b www houyuanhk com Answer Answer a channel if ringing Authenticate This application asks the caller to enter a give
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