ATCOM AX-400P Install guide
Contents
1. Add Zap Trunk Use the SSH tool to open the file etc asterisk zapata auto conf You will see there are five channels here Just like Span 1 WCFXO 0 Wildcard X100P Board 1 RED signalling fxs_ks Note this is a trunk Create a ZAP trunk in AMP for Channel 1 context from zaptel group 0 channel gt 1 Span 2 WCTDM 0 Wildcard TDM400P REV E F Board 1 signalling fxs_ks Note this is a trunk Create a ZAP trunk in AMP for Channel 2 context from zaptel group 0 channel gt 2 signalling fxs_ks Note this is a trunk Create a ZAP trunk in AMP for Channel 3 context from zaptel group 0 channel gt 3 signalling fxo_ks Note this is an extension Create a ZAP extension in AMP for Channel 4 context from internal group 1 channel gt 4 signalling fxo_ks Note this is an extension Create a ZAP extension in AMP for Channel 5 context from internal group 1 channel gt 5 To use these five channels we can use the separate via 1 2 3 or use them as a group via g0 group 0 Go to FreePBX gt Set up gt Trunks and add a new Zap trunk Outbound Caller ID Specify your caller ID when making outbound calls No use for FXO lines 13 32 www datcom cn Maximum Channels Maximum available simultaneous outbound calls Outbound Dial Prefix 9 Trixbox will add a prefix to the number you dial and send to the trunk Zap Identifier g0 the three FXO ports are grouped as g0 s2. can input extension number and voice mail password to enter the corresponding extension s voicemail box 9 32 www datcom cn 3 Make outbound call To make an outbound call we need to add trunk first There are many types of trunk SIP trunk Zap Trunk and so on We use AX 100p and A 400p to make zap trunks here Install AX 100p and AX 400p to the PC Hardware 1 Power off Trixbox and pull out the power adapter 2 Insert AX 100p card to the PCI slot of your PC and firm it with screw 3 Insert AX 400p card to the PCI slot please see below picture for the configure Module Three Module Module Four Module Two sti we Set up Insert AX 110X FXO module to module 1 and module 2 port 1 and port 2 will then be configured as FXO port ui Insert AX 110S FXS module to module 3 and module 4 port 3 and port 4 will then be configured as FXS port Bee ui If you have insert AX 110S module to the TOE o AX 400p because the FXS port need to provide AU 110S FXS AU 110X FXO signal to the normal phone you also need to power the AX 400p with your PC via the power port Otherwise AX 400p won t have sufficient voltage to drive the normal phone connected What are FXO and FXS FXS Foreign eXchange Station is an interface which drives a telephone or FAX 10 32 www atcom cn machine FXS interfaces get phones plugged into them delivery battery and provide ringing FXS interfaces are signalled wi
3. Server with Trixbox installed we also install AX 100p and AX 400p in this server we install two AX 110S FXS modules and two AX 110X modules so you can connect three PSTN lines and two normal phones So there are five analog ports on the Trixbox server one FXO port from AX 100p two FXO port from AX 400p two FXS ports from AX 400p 2 Normal phone connect to the AX 400p FXS ports 3 Working PC x lite runs on this computer act as a sip extension 4 AT 530 IP phone register to Trixbox to act to as SIP or IAX2 extension We want to provide below function in our simple application 1 free internal call 2 Voice Mail 3 make out bound call 4 IVR system We will implement these functions step by step 5 32 www atcom cn 2 Install Trixbox and make internal calls Install Trixbox The set up of Trixbox is simple Just download the ISO file from www trixbox org and record it in the CD to make an auto boot CD PUT this installation CD and reboot your computer then it will format your system and install the CENTOS and Trixbox server on it I am using Trixbox2 0 version on this article If you have question in install you can refer the www trixbox org for the install guide Ways to Access Trixbox There are two ways to access the Trixbox server gt Use Web browser to open the IP of the Trixbox server to connect to the GUI of Trixbox After open the Trixbox web interface you need to switch the access mode from use
4. eesessossesoesossesoosoesescoscesossossesossossesossssse 21 US 21 Link outbound Poute 22 6 Remote register through IAX2 protocol cccssccssssccssssscsssccsssscsssssscssssseseees 23 Add TAX extensionen nunne a e E a ee eE aE EE ERER 24 Add Port Forwarding On router a ww 24 Ts Echo Cancelation sscessiesicesss ivssacsnsnsnsces cusvesssnsicassassdas veusecsavaseuveasesecdsessccitesseensieste 27 Use ztmonitor to ttace the Ch ows iii sa iendsdacies avevee tetas Mares encanto cee ees aie 27 Use FXOTUNE tool to configure the line impedance pp 27 Install the MG2 echo cancellation cccccecccessceeseecssceceseeeseceeeescecsseecssceceseeeseseeeneesseeeseneeees 28 Octware echo cancellation ea 28 8 Install AX 4S in the TrixboX e ssseseessssesessessossossesossossesossosseecosoesoesossesossossessssssse 29 Configure the jumper and switch of the card 4 29 Install the MISDN Cri Vers sccsscelecosensdvasscussiess cas tvasauesliies ae e en EESE KESE e aea a niie iieii 29 Modify misdn init conf and misdn conf file es 29 Add custom trunk ii iissids tsi diesen betas i Gave ddinaaeeele tne aesdcdum ane aera dude ade 30 Add inbound TOUS cenia E ER E E T 30 Auto load the AX 4S card after system StartQp 3 31 9 The WO vases sase scensheccecaracacecaivendeteneuahegasenavtuescacicausaush auecedeunceeditehechasveaieedseasteeedaeans 32 2 32 www atcom cn 3 32 www atcom cn 1 Introduce This article
5. forwarded to IP Address 0 to 0l Both 192168110 O 1921681 2 then all HTTP m requests from outside users will 0 lto 0 Both 19216810 Oo be forwarded to 192 168 1 2 It The IAX2 protocol use the 4569 port as it register port and voice communicate port So I forward port 4569 to the public IP Then all data incoming from the internet via port 4569 will be sent to my Trixbox server 192 168 1 14 Go to the Linksys status page 24 32 www dtcom cn LINKSYS A Division of Cisco Systems Inc Firmware Version 1 04 17 Etherfast Cable DSL Router BEFSR41 V3 St atus Applications Setup Security amp Gaming Administration Router J Local Network Router Status Firmware Version 1 04 17 Now 26 2003 This screen provides the MAC Address 00 14 BF AC 7A F8 Router s current status information in a read only format Login Type PPPoE LODE Di t This field shows the Internet Login Status Connected login status When you choose PPPoE RAS PPTP or HBS as Internet IP Address 121 35 127 209 the login method you can click the Connect button to log in If Static DNS1 202 96 128 86 you click the Disconnect Static DNS2 202 96 134 133 button the Router will not dial up again until you click the Static DNS3 0 0 0 0 Connect button I can see that my public ip is 124 35 127 209 so I can use my ATA AG 188 to register to my Trixbox server via internet VOIP Udltew
6. is a guide to use Trixbox and ATCOM products to build a simple application on small office application Through this article we hope that reader can build the IP PBX solution for small enterprise Related Hardware and software Trixbox2 0 Asterisk based Trixbox enables even the novice user to quickly set up a voice over IP phone system Trixbox can be configured to handle a single phone line for a home user several lines for a small office or several T1s for a million minute a month call center AX 100p Asterisk PCI card with One FXO port AX 400p Asterisk PCI card with four FXO FXS interchangeable ports four ports can be configured as FXO or FXS port individually AX 110S FXS module of AX 400p AX 110X FXO module of AX 400p AT 530 Stand along IP phone can be configured as SIP IAX2 extensions X lite softphone run on the PC can be configured as SIP extensions Normal phone connect to AX 400p act as ZAP extensions 4 32 www dtcom cn System set up Simple structure for small enterprise Three PSTN Lines Zap Group J ley Normal Phone Zap Extension 2002 Router IP 192 168 1 1 G TrixBox Server AX 100p AX 400p IP 192 168 1 14 N Working PC With x lite installed SIP Extension 2003 A cal Area Networ a AT530 IP Phone ATS530 IP Phone SIP Extension 2004 SIP Extension 2005 Normal Phone Zap Extension 2001 We use following devices in our set up l
7. to make outbound calls you need to add prefix 9 to your number when dialing We have finished outbound call now the next step we will add IVR for our system 17 32 www datcom cn 4 IVR IVR means Interactive Voice Response It is very important for the enterprise application We can use the IVR to do the auto attendant job Generate the Voice file Go to FreePBX gt Setup gt System Recording gt Add Recording You can use your extension to record a voice file or Upload a voice file Record via extensions Enter an extension 2001 in the record via extension and Go to the next page System Recordings y g Add Recording Add Recording Built in Recordings Step 1 Record or upload lf you wish to make and verify recordings from your phone please enter your extension number here 2001 Go Alternatively upload a recording in wav format pl Upload system Recordings Add Recording Step 1 Record or upload Using your phone dial and speak the message you wish to record Alternatively upload a recording in way format Step 2 Verify After recording or uploading dia 99 to listen to your recording lf you wish to re record your message dial 77 Step 3 Name Name this Recording Greeting_ATCOM Click SAVE when you are satisfied with your recording Save 18 32 www dtcom cn Specify the name of this record and dial 77 in 2001 extension and you will here indicate sound and then record yo
8. 32 www dtcom cn Add IAX2 extension Go to gt Free PBX gt setup gt Extensions gt add generic IA X2 device and add a IAX2 extensions Add JAX2 Exenions User Extension 2006 Phone number of this extension Display Name Alice Caller ID Secret 2006 IAX2 Log on password Enable Voicemail Voicemail password 2006 password of your mailbox Add Port Forwarding on router Since our Trixbox server is behind the router and don t have public ip We need to do port forwarding in our router so the corresponding packets call be rend to the Trixbox server Below is the port forwarding setting in my Linksys router Linksys A Division of Cisco Systems Inc Firmware Version 1 04 17 Etherfast Cable DSL Router BEFSR41 V3 Applications amp Gaming Setup Security preen Port Range Forwarding Port Triggering 1 UPnP Forwarding Administration Status Port Range Forwarding Port Range Forwarding Port Range Port Range Forwarding can be used to set up public services Application Start End Protocol IP Address Enabled on your network When users 7 from the Internet make certain iax2 4569 to 4569 Both 1921684114 requests on your network the E Router can forward those 0 to 0 Both 1921681 0 El requests to computers equipped to handle the requests If for example you set the port v 0 to 0 Both 192 168 1 0 number 80 HTTP to be FE
9. about this utility please refer www voip info org you can find more detail of this tool 27 32 www atcom cn Install the MG2 echo cancellation If the echo is still worst or you hear some noise at the beginning of your speech then we can try to use the MG2 echo cancellation Trixbox doesn t install this soft echo cancellation in default The soft echo cancellation may give you a good result in Trixbox Download the latest 1 2 version zaptel for asterisk official website www asterisk org before install the zaptel driver we need to install the kernel source of Trixbox Use SSH to connect the Trixbox server and run Install Kernel Source yum y install kernel devel kernel yum y install kernel smp devel There is a bug in the new kernel to fix it you need to cd usr srce kernels 2 6 9 34 0 2 EL smp i686 include linux vi spinlock h search for rw_lock_t in that file and change it to rwlock_t otherwise there will be error when compile the zaptel driver Install Zaptel Download the zaptel driver cd usr sre tar zxvf zaptel 1 2 17 tar gz cd zaptel 1 2 17 vi zconfig h and locate define ECHO_CAN_MARK2 define ECHO_CAN_MARKS3 define ECHO_CAN_KBI MG2 is a version of KB1 that has some changes to it that are supposed to improve how it performs If you have echo problems try it out define ECHO_CAN_MG2 Uncomment the define ECHO_CAN_MG2 And comment define ECHO_CAN_KB1 make c
10. art Default Slaves 03 Channel 04 FXO Kewlstart Default Slaves 04 Channel 05 FXO Kewlstart Default Slaves 05 5 channels configured Above message shows that AX 100p and AX 400p are already auto configured You can see that channel 1 2 and 3 are configured as FXS_KS signaling that means ports 1 2 and 3 will be act as FXO ports And for the ports 4 and 5 they use FXO_KS signaling and act as FXS ports Check AX 100p and AX 400p status Type root asterisk1 zttool prebe Zapata Telephony Interfaces ce eA N oe 4H Jalarns Span OK Wildcard TDM400P REV I Board 1 UNCONFIGURED ZTDUMMY 1 1 fitdtitttt ttt 30 Here ihe AX 100p s state is RED Fern that you haven t connected the PSTN lines to AX 100p It shows OK after the PSTN lines is connected and ready for use Regarding the AX 400p it shows OK all the time when the driver is install correctly After auto install the configure the AX 100p and AX 400p You also need to add two properties in the etc asterisk zapata conf file under the channels section busydetect yes busycount 5 This two command is to enable the busy detect on the system when you make calls to a PSTN network the other sides hangup the call you will here the busy tone If you don t enable busy detect The system will regards the busy tone as a normal talking tone so it won t hangup the line and other one can not call you 12 32 www atcom cn
11. beronet com download install misdn mqueue tar gz get mISDN and chan_misdn tar xzfv install misdn mqueue tar gz cd install misdn mqueue make make install reboot 2 oS A After reboot we have successfully install the misdn driver of the AX 4S and we can use the card now Modify misdn init conf and misdn conf file The job of etc misdn init conf file is the same as the zaptel conf I put below on this file card 1 0x4 this is for the AX 4S card te_ptmp 1 2 3 4 configure port 1 4 to TE port etc asterisk misdn conf is the interface of the card and asterisk Most important is intern 29 32 www datcom cn define your ports e g 1 2 depends on mISDN driver loading order ports 1 2 3 4 context from pstn context where to go to when incoming Call on one of the above ports msns In the misdn conf file I configure the port 1 4 as a group named intern to use these ports to make outgoing call We need to add a custom trunk Add custom trunk Go to FreePBX gt Set up gt Trunks and add a new custom trunk Configure the outgoing setting as Outgoing Settings Custom Dial String mISDN g intern OUTNUM b Submit Changes You can also use mISDN 1 SOUTNUM b to specify the outgoing call via port 1 Then add a new route to this trunk and you can make outgoing call via the AX 4S card Add inbound route According to the misdn conf file our incoming call will go to from pstn context s exte
12. dy lAX Registered Configuration sum Pr ar ao The configure method of AG 188 is the same as AT 530 TAX Server Addr 121 35 127 209 Account Name 2006 Account Password 2006 Phone Number 2006 Enable Register Enable IA X2 as default protocol 25 32 www atcom cn OK Now we can use the AG 188 to register our Trixbox server now but it is very annoy because our public IP is dynamic and will change after several hours Fortunately there is a service called DDNS Via the DDNS you can bind your dynamic public IP to a fix domain For example I am using a free DDNS service from a Chinese company and they give me an account I run the DDNS client on the Trixbox server The client will connect to the DDNS server and send my public ip to them and the server bind the ip with my register domain So I can use my domain to access the dynamic ip 26 32 www atcom cn 7 Echo Cancellation Echo is a big problem in VoIP and the most possibility echo problem in our system is the making outbound calls via an IP Phone Below is several ways to improve our voice quality Use ztmonitor to trace the echo Run ztmonitor 1 v to trace the port and you will see Audio Level Max Audio Hit HHT HEHEHEHEH HEHEHHE You can see the RX voice level and TX voice level when you are talking and if we mute one side for example mute the RX PSTN side and the there sh
13. e 2003 Domain 192 168 1 129 IP address of your Trixbox server 2 Register IP phone AT 530 a b c d Connect the AT 530 s WAN port to the switch And it can get the ip from your router Press the sysinfo key on AT 530 to get the IP of AT 530 Put the ip on the IE of your computer and you can enter the AT 530 configure page through this ip Put the SIP extensions info on the AT 530 IP phones 8 32 www datcom cn IP Phone SIP Registered Configuration 192 168 1 129 Proxy Server Addr 5060 Proxy Server Port 2004 Proxy Username on Proxy Password 2004 Register Expire Time 2833 RFC Protocol Edition Enable Register Register Server Addr 192 168 1 129 IP address of Trixbox server Register Username 2004 Register Password 2004 Phone Number 2004 Use the same method register another at 530 to extension 2005 then you can free inbound call between these three extensions 2003 2005 Voice Mail Box Voice mail box is enabled when we create the extensions So if somebody calls you on your extension and you are unavailable he will hear the voice mail greeting message and leave voice message Retrieve voice message Dial 97 in your sip terminal and input the voice mail password of your extension then you will enter your voice mail box Dial 98 in your sip terminal and you will enter the voice mail main menu then you
14. et 2003 SIP Log on password Enable Voicemail Voicemail password 2003 password of your mailbox you need to input this password when you enter the mailbox Add Extension User Extension 2003 Display Name Fantasy Device Options secret 2003 dtmfmode rfc2833 Voicemail amp Directory Status Enabled voicemail password 2003 email address pager email address Use the same method add two other extensions User Extension 2004 Phone number of this extension Display Name Edwin Caller ID Secret 2004 Log on password Enable Voicemail Voicemail password 2004 password of your mailbox User Extension 2005 Phone number of this extension Display Name Marvin Caller ID 7 32 www dtcom cn Secret 2005 Log on password Enable Voicemail Voicemail password 2005 password of your mailbox So we have created three SIP extensions And we can use SIP device to log on the Trixbox use these extensions Register SIP device 1 Register SIP softphone Download the x lite softphone phone from couterpath website www xten com After install the x ltie right click the panel and select the SIP Account Setting and then configure it Account Voicemail Topology Presence Advanced User Details Display Name User name 2003 Password akok Authorization user name 2003 Domain 192 168 1 129 Display Name Fantasy User Name 2003 Password 2003 Authorization User Nam
15. lean make install finish the install of zaptel Then MG2 echo cancellation is successful installed in your system Octware echo cancellation Octasic provides a software echo cancellation for Asterisk it can improve the voice quality much Trixbox 2 2 has installed this software OCTWARE default You will need the license key to use it You can find it easily on the internet the price is about several US dollars This is the best choice to improve the voice 28 32 www atcom cn Install AX 4S in the Trixbox AX 4S is a four ports BRI card It doesn t use the zaptel driver so the genzapconf command doesn t work on it If you need to install this card on the Trixbox server you need to install the mISDN driver it base on the mISDN driver Configure the jumper and switch of the card First according to the AX 4S manual to set the card to correct jumper I use the card to connect the NT Plus s S T port so I configure the card s four ports as TE port The Jumper setting should be jumper S401 404 set to right side Switch 401 402 set to ON Left side And then put the AX 4S card on your PC Install the mISDN driver Trixbox doesn t have the linux kernel source in their default install You need to install the kernel source manually otherwise you will have trouble to install the oterh software 1 yum y install kernel devel kernel 2 yum y install kernel smp devel install kernel source 3 cd usr sre 4 weet
16. nsion We need to add a route how it should work for this s extensions Go to FreePBX gt SetUp gt Inbound Routes gt Add Inbound Route Edit Incoming Route DID Number z Caller ID Number Set Destination O Recordings Greeting_ATCOM v Core Hangup J IVR Welcome iv O Custom App 30 32 www atcom cn After doing that all calls to the BRI port will route to the IVR entry Auto load the AX 4S card after system startup Open the etc re d re local file and add a new line on this file to auto start the AX 4S card after system startup etc rc d rc local bin sh This script will be executed after all the other init scripts You can put your own initialization stuff in here if you don t want to do the full Sys V style init stuff touch var lock subsys local etc init d misdn init start lt add this line etc trixbox runonce usr local sbin motd sh gt etc motd usr sbin fxotune s usr sbin amportal start 31 32 www atcom cn 9 The End Thanks for your reading of this article If you have question or advise of this article please feel free to contact me at edwin atcom com cn Next update Value add service calling group FAX Skype and so on 32 32
17. o when you make outbound call via this trunk Trixbox will pick up the available FXO channel in gO automatically The channel are grouped by Trixbox when you use genzaptelconf you can see the group information in the file etc asterisk zapata auto conf you can also put 1 2 or 3 here to specify the separate FXO port here Add ZAP Trunk GeneralSettings Outbound Caller ID Never Override CallerlD O Maximum channels Outgoing Dial Rules Dial Rules Clean amp Femove duplicates Dial rules wizards pick one v Outbound Dial Prefix Outgoing Settings Zap Identifier trunk name g0 Submit Changes 14 32 bd www dtcom cn Add outbound routes Ok We have added the trunk already To make out bound calls we also need to specify that which trunk that our calls should route to Go to the Freepbx gt Setup gt Outbound Routes gt Add Route Add Route Route Name 9 outside Route Password PIN Set Emergency Dialing Intra Company Route O Dial Patterns gl Clean amp Remove duplicates Insert Trunk Sequence Submit Changes Route Name 9 outside Dial Patterns 9 Trixbox will cut the first number 9 if the phone number dial begins with 9 Trunk Sequence ZAP g0 all number accord with the above Dial Patterns will be sent to ZAP g0 trunk This outbound route means if we dial any phone number start with 9 for e
18. ould be total silent in the RX side In this case RX level should be 0 and if we are talking on the TX side and see there are some audio on the RX side it means there are echo in our talking The first way to reduce the echo is modify the RX and TX gain we can go to etc asterisk zapata conf and adjust the rxgain and txgain Modify and modify up to the echo is acceptable Remember you need to go to the asterisk CLI to reload the chan_zap so module after every time you modify the file Use FXOTUNE tool to configure the line impedance Every country has different PSTN line impedance If the line impedance doesn t match then you will experience more or less echo Fxotune utility is a tool to auto configure the line impedance of your PSTN line and set the corresponding echo coefficient for your line This tool is installed default in Trixbox Use the SSH tool to access the Trixbox and run asterisk vvvvvere stop now before running the fxotune You need to stop asterisk fxotune i 5 The system will test the PSTN line impedance and set the corresponding echo coefficient in the etc fxotune conf file This will take about 20 minutes for to do it Notice It will test all the FXO ports in your system if you don t have the PSTN line connect to this port it will generate wrong data in the etc fxotune conf file fxotune s to apply the setting Run the asterisk and then you will find that the voice is better than before More info
19. r to admin The default username and password is maint and password This is base configure way of Trixbox Trixbox doesn t support IE well You can use Firefox to configure it gt Use the SSH tool to access the Trixbox server in this way the use name is root and password is the password you input when install the Trixbox Most of our configure job will be done in the web interface and we will do some advance configure via SSH tool Install Trixbox modules Trixbox doesn t install many function default you need to install them manually In the web interface go to the Asterisk gt Free PBX page gt Tools gt Module Admin You can see many function modules in this page Just select all and process the install to install all these modules Notice every change on the Trixbox will show a red section ask you to apply the change via click the link Add extensions At first we need to add some extensions to make internal calls Each extension acts as an internal number There are many types of extensions we will use SIP IAX2 and ZAP extensions on this article 6 32 www datcom cn Add SIP extensions Sip extensions is an SIP account allows you to log on the Trixbox via an SIP terminal such as IP phone AT 530 and softphone x lite Go to gt Free PBX gt setup gt Extensions gt add generic SIP device Add Exenions User Extension 2003 Phone number of this extension Display Name Fantasy Caller ID Secr
20. th 8 will outgoing via our SIP service gt And calls begin with 9 will outgoing via the PSTN line gt Other calls will be regards to internal calls 22 32 www dtcom cn 6 Remote register through IAX2 protocol We have built a local simple IP PBX system so far This is the total structure of my system gt Ilease an ADSL line to connect the internet gt Ihave a dynamic WAN IP dynamic IP is enough to me and static ip is expensive gt Ihave a Trixbox server behind my router gt Ican use my ip phone connect the Trixbox to act as an extension in my local area network They are at the same network as Trixbox And I want to use my Trixbox server when I am on business trip or at home So how can I do it in this case that my Trixbox server is behind a router and have a private IP The key point is the IAX2 protocol and port forwarding Simple structure for small enterprise Router Three PSTN Lines Zap Group Normal Phone Zap Extension 2002 Normal Phone Zap Extension 2001 IAX2 extension 2006 Normal Phone A 7 Tf I Linksys Router IP 192 168 1 1 TrixBox Server AX 100p AX 400p IP 192 168 1 14 N Working PC With x lite installed SIP Extension 2003 AT530 IP Phone AT530 IP Phone SIP Extension 2004 SIP Extension 2005 Above it the update structure of my system I have added a IAX2 extensions 2006 and I use AG 188 ata to register the Trixbox server remotely Below is the step 23
21. th FXO signalling FXO Foreign eXchange Office is an interface that connect to a phone line They supply your PBX with access to the public telephone network FXO interfaces use FXS signalling FXS interfaces are what allow you to hook telephones to your PBX and FXO interfaces allow you to connect your PBX to real analog phone lines Configure AX 100p and AX 400p on Trixbox There is a command genzaptelconf can generate the configure file of AX 100p and AX 400p automatically Use the SSH tool to connect Trixbox server and run root asterisk1 genzaptelconf Trixbox will then auto detect zaptel hardware and install AX 100p and AX 400p driver automatically You may see below info in this process Loading wcfxo wcfxo DAA mode is FCC Found a Wildcard FXO Wildcard X100P Detect AX 100p card Loading wctdm Freshmaker version 7 Freshmaker pass register test Module 0 Installed AUTO FXS DPO Module 1 Installed AUTO FXO FCC mode Module 2 Installed AUTO FXO FCC mode Module 3 Installed AUTO FXS DPO Found a Wildcard TDM Wildcard TDM400P REV I 4 modules Detect AX 400p card Which indicate that Trixbox detect and install AX 100p and AX 400p successfully And type root asterisk1 ztcfg vvv To see the channel state Zaptel Configuration Channel map Channel 01 FXS Kewlstart Default Slaves 01 Channel 02 FXS Kewlstart Default Slaves 02 11 32 www atcom cn Channel 03 FXS Kewlst
22. to make calls to via the voipbuster service Add SIP Trunk Go to FreePBX gt SetUp gt Trunks gt Add SIP trunk Outgoing Settings Trunk Name SIP_Internat ion PEER Details host sip voipbuster com secret passowrd type peer username ani ceman Trunk Name SIP_International Peer Details host sip voipbuster com the voipbuster SIP server address secret password your sip account password type peer can receive and make calls via this trunk username aniceman your sip account username Registration Register String aniceman password sip voipbuster com Submit Changes Register string use this string to register the voipbuseter service Format is username password sip serve ip 21 32 www datcom cn Link outbound route to SIP trunk Go to FreePBX gt SetUp gt Outbound Routes gt Add ourbound routes Route Name 8_international_callS Rename Route Password PIN Set Emergency Dialing O Intra Company Route O Dial Patterns gl Clean amp Femve duplicates Insert Pick pre defined patterns v Trunk Sequence D SIP SIP_International v Add Route name 8_international_calls Dial Patterns 8 Trunk Sequence SIP SIP_International Now we have added a new international route All number begin with number 8 will be sent to this route According to the zap trunk in the above chapter in our system gt Calls begin wi
23. ur voice When hang up and Save now we have record a voice name Greeting ATCOM in the Trixbox Notice you can also upload a fair sounding voice to the system in the add recording page Add IVR entry Go to FreePBX gt SetUp gt IVR gt Add IVR Digital Receptionist Edit Menu Unnamed Delete Digital Receptionist Unnamed Change Name Welcome Timeout 10 Enable Directory Directory Context default v Enable Direct Dial Announcement Greeting_ATCOM v Change Name Welcome Announcement Greeting ATCOM 19 32 www datcom cn Add incoming route to IVR entry Go to FreePBX gt SetUp gt Inbound Routes gt Add Inbound Route Add Incoming Route Add Incoming Route DID Number Caller ID Number OR Zaptel Channel Set Destination Recordings Greeting_ATCOM i O Core Hangup v IVR Welcome Custom App Zaptel Channel 1 Recordings Greeting_ATCOM Then all PSTN incoming to channel 1 will be routed to our IVR system and they can dial any internal extension when listen the VR 20 32 www datcom cn 5 Make outbound calls via SIP Service We have added zap trunk in the before chapter But the rate for international call on the PSTN line is expensive I want to more lower rate for our international call So I apply an voipbuster account they have a low international rate and after adding a SIP trunk we can use our extension
24. www atcom cn Tour of Trixbox Edwin Version Date Author Description 1 0 2007 June 08 Edwin Creation 1 32 www atcom cn Ms TWO C AET A T 4 Related Hardware and software osise ienien Se Aee aae ri iR oori iaiia oia kiai 4 Syst m Set UP seoan a a a A a a OE 5 2 Install Trixbox and make internal calls e sosseseosoesesessoesessossesossossesessossossossesese 6 St 0 We 0 Install Trix DOX AMG CUES senere e a a a aa a a 6 Add CXtENSIONS sneissis iinit iiin e n E EREE E EREE EEE RR 6 Register SIP devices cesna ene ana aen A e e EA aa EREE Aea E IE 8 VOICE Mail BOX saceceessbasesaiess copsaceacesases cecscaceatesssenaccecsscessvesieedeees bu cgaseaseubavensaegasved iaieetecdepaseetises 9 3 Make outbound call wssicccdcsssccdvessenncccsassnvesisscasicasesnevecsissceures seuss couesdeanscceeauesccaseats 10 Install AX 100p and AX 400p to the PC Hardware cece ees eeceeeseeseceeceeeeseeseeseeeaeeas 10 Configure AX 100p and AX 400p on TFIXbOX eee eee ceeeeeeeecseseeeeseeaeeseeeeeeaetaseeeeeeeaes 11 Add 2 13 Add outbound routes ensien e anaes e i a EE EATE 15 Add Zap extensionS siseses iiinis aisis iiei ei o 16 18 Genmirat MO 18 Add IVR entity csciivceisieetsuisseeicesssestaeslund sua siedlouesevs sueadesabuasnueacussunaseuscaubdenneluaacspalvacsuteausaunandeaees 19 Add incoming route to TVR entry ee 20 5 Make outbound calls via SIP Service
25. xample 983018806 then Trixbox will cut the number first number 9 and send the number 83018806 to the g0 group it will use the available channel of channel 1 2 3 After doing above you can use SIP extensions to make outbound calls 15 32 www datcom cn Add zap extensions Remember that we have two FXS modules on the AX 400p card we can use this two fxs port to build two zap extension Go to the Free PBX gt setup gt Extensions gt add generic zap device Add Extension User Extension 2001 Display Name Annie Device Options channel a Wai S 9 H m Voicemail amp Directory Status Enabled v voicemail password 2001 email address pager email address User Extension 2001 Display Name Annie Channel 4 Enable VoiceMail VoiceMail Password 2001 the fourth channel is the FXS port Now connect a normal phone to the 3 port of the AX 400p card and pick up the phone you will hear the dial tone and you can make calls now Use the same method to add another zap extension 2002 User Extension 2002 Display Name Crystal Channel 5 Enable VoiceMail VoiceMail Password 2002 System Review So far we have the fifth channel is the FXS port Five internal extensons 2001 2005 calling between these extensions are free 16 32 www atcom cn And we have assigned voice mail boxes to each extension To expand the number of extensions you can just add IP phone or softphone Three PSTN extensions
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