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ATCOM® IPPBX IP-2G4A Product Guide

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1. Operator The Operator Extension is the extension which will be dialed Chioce Extension when a caller presses 0 to exit Voicemail It is also available as Voice Menu option Ring Timeout Number of seconds to ring a device before sending to the user s Time 20 Voicemail Box Call Record Dir Call Record Dir Str tmp Call Record Call Record Format Choice gsm Format 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 Languages Description default Languages The Language setting allows the user to specify the default prompts language for phone to phone inbound and outbound calls installed then the pack will be downloaded from Digium If a soundpack selection is made but not already Chioce English Change Password Change Password Description Enter New Password Str Retype New Password Str Factory reset Factory reset Description Reset to defaults include network settings Reset to defaults but keep network settings www atcom cn 30 ATCOM Chapter 5 an Application Case of IP 2G4A Public IP Address 118 37 xxx xxx Public IP Address 116 120 xxx xxx IP04 MP IP Address 172 16 1 5 AT 610 IP Address 172
2. NAT Can Reinvite DIMF Mode EEC2333 ivi insecure very V r Other Options 3 Way Calling In Directory Call Waiting Fer Is Agent Enable Call Record Pickup Group ww Q Cancel Mi Update You must pay attention to the red ellipse frame on Analog Station drop down list and choose the port which you need At last please click on Update button and click on Apply Changes button in up right corner of the main page Attention then you must reboot your IP 2G4A The configuration can go into effect 5 1 3 Register a SIP user 6001 in AT 620 After logging into the web page of IP Phone AT 620 please select VOIP option I register the 6001 as the following illustration SIP1 Configuation Basic Setting Register status Server Address 192 168 1 130 anm mmm EE Enable Register Phone Number Display Name APPLY Advanced Set After configuring please click on the APPLY button You can see the Register status is Registered if you do not register successfully 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 2G4A Now you can call each other directly between user 6001 and 6005 www atcom cn 34 gg ATCOM 5 2 How to Communicate with Outside through PSTN In order to communicate with outside through PSTN with IP 2G4A you need an analog
3. 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 www atcom cn 24 QT ATCOM either begin or end the conference altogether Pin Code set an optional pin code Ex 1234 that must be Str entered in order to access the Conference Bridge Admin PinCode Defining this option sets a PIN for Conference Str Administrators Play music for Checking this option causes Asterisk to play Hold Check box unCheck the first caller Music to the first user in a conference until another user has joined the same conference Close conference Close the conference bridge when the last marked Check box unCheck for the list caller user logs out of the conference call exit Enable call menu Checking this option allows a user to access the Check box unCheck Conference Bridge menu by pressing the Asterisk key on their dialpad Announces Checking this option announces to all Bridge Check box unCheck callers participants the joining of any other participants Quiet mode Do not play enter leave sounds Check box unCheck Waitfor marked Prevent conference participants from hearing each Check box unCheck user other until the
4. ta Home Felcome to YoIPtel CE Please login Asterisk Configuration Engine Username Password Please input the default Username admin Password atcom in the presented screen above When you login successfully you can get the configuration web page as below Apply Changes E Logout manage related features p VolPtel Uptime 04 14 57 up 10 min load average 0 04 0 06 0 02 Trunks Port Hostname IP Extensions 9 busy unavailable Check Voicemails Dial by Names 5 1 2 Add up Users from Web Page of IP 2G4A 1 Add up a DialPlan Before you add up user you have to add up a DialPlan please click on Dial Plans New DialPlan Iadd up a DialPlan like the following www atcom cn 32 ATCOM Create New DialPlan DialPlan Name DialPlanl Include Outgoing Calling Rules You do not have any calling Rules defined click here to manage calling rules Include Local Contexts v default parkedcalls Miconferences lvlringgroups Vlvoicemenus lvlqueues lvlvoicemailgroups M directory QCancel M Save After configuring please click on Save button and click on Apply Changes button in up right corner of the main page 2 Add up SIP user 6001 After logging into the web page of IP 2G4A please click on Users gt Create New User I configure user 6001 like the following Create New User r General Extension 6
5. video mode ATCOM 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 4 15 Follow Me If A calls B B does not answer the call will be transferred to C who is set up in follow me Name Description Type default Status Enable Disable FollowMe for this user Choice Disable Music On Hold Music On Hold class that the caller would hear while Choice Default Class tracking the user DialPlan DialPlan that would be used for dialing the Choice FollowMe numbers By default this would be the same dialplan as that of the user Destinations List of extensions numbers that would be dialed to Destinations reach the user during FollowMe New FollowMe Add a new FollowMe number which could be a Dial Local Number Local Extension or an Outside Number The Extension Dial selected dialplan should have permissions to dial any Outside outside numbers defined Number Dial Order This is the order in which the FollowMe destinations Ring after Ring after are dialed to reach the user Trying Trying previous previous extension num
6. 1 Music on hold Dir persistent sounds moh www atcom cn 17 2 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 3 Music on hold after holding status Status busy 4 Music on hold non rtp stream 4 9 Call Queues ATCOM Please select the Call Queues option from the vertical menu on the left of the main page then you can get the following screen Name Extension Description Extension for call queue may be dialed to reach the call queue Type Int default 6500 Name Name for call queue Str Strategy Strategy this option sets the ringing strategy for this queue the options are 1 Ring all ring all available agents simultaneously until one answers 2 RoundRbin Take turns ringing each available agent 3 LeastRecent Ring the agent which was least recently called 4 FewestCalls Ring the agent with the fewest completed calls 5 Random Ring a Random agent 6 RRmemory RoundRobin with Memoryn Remember where it left in the last ring pass ringall Ro undrobin le astrecent F ewest calls Rand om Rrmom ery ring all Music On Hold Select the Music on Hold Class for this Queue Music on Hold classes can be managed from the the Music On Hold panel on the left Choice default LeaveWhen Empty This option cont
7. Directory defaultldefaultlef 4 17 Call Features Feature Codes and Call parking preferences Features Codes Name Description Type default Features Codes Blind Transfer default is Check box amp amp Int Disconnect default is Check box amp amp Int Attended transfer Check box amp amp Int Call Parking Packing a call Check box amp amp Int Call Parking Preferences Description default 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 Application Map Description default Application Map Add an application for PBX Dial Options www atcom cn 27 ATCOM Name Description Type default Dial Options t Option Allow the called party to transfer the calling party by Check box Uncheck sending the DTMF sequence defined on the Feature Codes page T Option Allow the calling party to transfer the called party Check box Uncheck by sending the DTMF sequence defined on the Feature Codes page h Option Allow the called party to hang up by sending the Check box Uncheck DTMF sequence defined on the Feature Codes page H Option Allow the calling party to hang up by sending the Check box Uncheck DTMF sequence defined on the Feature Codes page k Option Allow the called party to enable parking of the call Check box Uncheck by sending
8. extension nu ber Ring mbe along with previous extension num ber Follow Playback the unreachable status message if we ve run Check box Uncheck me Option out of steps to reach the or the callee has elected not to be reachable Playback the unreachable status message if we ve run Check box Uncheck out of steps to reach the or the callee has elected not to be reachable Playback the unreachable status message if we ve run Check box Uncheck out of steps to reach the or the callee has elected not to be reachable 1 General config file etc asterisk followme conf www atcom cn 26 ATCOM 4 16 Directory 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 Name Description Type default Directory Extension to dial for accessing the Name Directory Int Extension Also read the In addition to the name also read the extension Check box Uncheck extension number to the caller before presenting dialing options number Use first name Allow the caller to enter the first name of a user in the Check box Uncheck instead of last directory instead of using the last name name 1 Directory application
9. mailbox number gt password name email mailbox number the number you use in extension conf for VoiceMail command and to register a user in sip conf or iax conf password the pass used to register a user in sip conf or iax conf name the name which to be associated with the mailbox email where a notification for the voicemail will come 3 IPPBX Max messages data 150M a Email Settings for Voice mails Name Description Type default Send messages If this option is set then voicemails will not be Check box unCheck by e mail only checkable using a Phone Messages will be sent via e mail only Note You need to have an smtp server configured for this functionality Attach recordings This option defines whether or not voicemails are Check 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 asterisk y www atcom cn 23 Voicemail Emails Qr ATCOM ourcompan y null Subject New voicemail from VM CALLERID for VM MAILBOX Template Variables X TAB VM NAME Recipient s firstname and lastname VM DUR The duration of the voicemail message VM MAILBOX The recipient s extension VM CALLERID The caller id of the person who left the message VM MSGNUM The message number in your mailbox VM DATE The date an
10. D Max Len 0 1 O Auto Fill J D Auto Pause Report Hold Time KeyPress Events None Agents Q iz agnes 6001 V Peter 6002 Rose 6008 Q cancel Mi Update Then 6008 have registered can call 6500 then 6001 6002 are ringing together Of course if you want to configure it in detail you can refer to 4 9 5 10 Follow Me Here 6001 6002 6005 have already registered and they can communicate with each other First please click Follow Me Edit on 6001 you can configure the options like the following screenshots Status Enable O Disable Music On Hold Class D defaut DialPlan Destinations Add FollowMe Number Q Cancel Mi Save Then click the button of Add FollowMe Number you can see the screen like the following you can configure it like this here I selected the Dial Local Extension number is 6005 Then click on www atcom cn 49 ATCOM Add button then please click on Update button and click on Apply Changes button in up right corner of the main page Status Enable Disable Music On Hold Class D defaut DialPlan D DialPlan1 Destinations D New FollowMe Number Dial Local Extension Dial Outside Number 6002 Peter for 30 Seconds Dial Order D Ring after Trying previous extension number Ring along with previous extension number Cancel Add Now when 6008 dial
11. 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 Pass the number to the appropriate trunk to make the call 4 5 Dial Plans A 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 Name Description Type Default DialPlan Name The name of DialPlan which is a unique label to Textbox DialPlanl help you identify the dial plan Include Outgoing Select outgoing call rule which you use selected Not seclect Calling Rules Include Local Local context is used for general using check box Select all Contexts configuration 4 6 Users Users component is used to add or remove Analog SIP IAX extension Click on Create New User button in the web of IP 2G4A you can create SIP IAX User and Analog User Name Description Type Default Extension The numbered extension Textbox 6001 Name A character based name for this user Textbox Null DialPlan DialPlans are sets of calling rules and can be managed ComboBo Null form the V Dial Plans panel X CallerID The Caller ID CID string used when this user calls Tex
12. Incoming Calling Rules Click Incoming Calling Rules gt New Incoming Calling Rules you can configure it like this Hew Incoming Rule ES Trunk analog V Time Interval None no TimeIntervals matched Pattern S O Cancel v Update Then when others call you through the analog1 they can here the IVR and do the operation which they need www atcom cn 45 ATCOM 5 7 Conference In order to realize the conference option the users which will attend to the conference must have registered Here I use 6001 6002 6005 Now please click Conferencing gt New conference Bridge you can see the screen like the following screenshots New Conference Bridge Extension 6300 Marked Admin user Extension 6322 r Password Options Pin Code 123 Admin PinCode 456 r Conference Room Options Play hold music for first D Close conference when last marked user caller exits MEO Enable caller menu nnounce callers MEO y Quiet Mode Wait for marked user O Cancel Update Then please click on Update button and click on Apply Changes button in up right corner of the main page Here I configure it like the screenshots above Then 6001 dial 6300 and input Pin Code you can here a voice promt means you are the fist user and wait oters then you can here the music 6002 does the same operation 6005 dial 6322 and input Admin PinCode Now a
13. Save button and click on Apply Changes button in up right corner of the main page Now you can call from 6030 to 6001 and 6005 by dialing with prefix 4 You can communication between IP04 and IP 2G4A 5 5 Voicemail You can configure Voicemail in Users Jor example 6005 which we have configured in 4 1 2 please click on Users Edit on 6001 you can see the configuration in the following picture especially pay attention to the configuration in the red ellipse frame Then when you want to listen to a message you can dial 6750 or the Mailbox 6001 Edit User Extension 6001 General Extension 500 Q gane agnes DialPlan DiaiPian wi CallerID agnes outBound CallerID agnes Enable Voicemail for this User D VoiceMail Access PIN code 123 D Mailbox 6001 Email Address ayi atcom com cn r Technology v sip v zax Analog Station None vi flash rxflash Q Codec Preference First u taw w Second zen ei Third ens w Fourth Nene iw Fifth Nene w r VoIP Settings SIP IAX Password HAT Can Reinvite O DTMF Mode EECH insecure very r Other Options L1 3 Way Calling L n Directory CO Caii Waiting L ert O rs Agent Fl eane Call Record O Pickup Group w O Cancel Mi Update 5 6 How to realize the IVR IVR is Interactive Voice Response Voice Menus allow for more efficient routing of calls f
14. and Check 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 messages 10 25 100 25 messages per that a user may have in any of their folders 200 500 10 folder 00 Max message This select box sets the maximum duration of a 1 minute 2 minutes time voicemail message in seconds Message recording will 2 not occur for times greater than this amount minutes 5 minutes 15 minutes 30 minutes um limited Min message This select box sets the minimum duration of a no 1 seconds time voicemail message in seconds Messages below this minimum threshold will be automatically deleted seconds 2 seconds 3 seconds 4 seconds 5 seconds Say message If this option is enabled the Caller ID of the party that Check box Check Caller ID left the message will be played back before the voicemail message begins playing Say message If this option is set the duration of the message in Check box unCheck duration mintues will be played back before the voicemail message begins playing Play envelope Turn on off playing introductions about each message Check box unCheck when accessing them from the voicemail application Allow users to Checking this option allows the caller to review their Check box Check 1 Voice mail application Voicemail ARG u 2 Automatically generated configuration file etc asterisk voicemail conf
15. d Memory Usage Total Memory quantity Used Used of Memory Free Free of Memory Shared Shared of Memory Buffers Buffers quantity 4 21 Backup Backup and Restore are two of the mandatory functions of any application IP 2G4A is no exception Customers can backup all the files under the etc asterisk directory and restore them Description Type default Backup Create new backup Download from Unit Restore Previous config 4 22 Active Channels The channels which are in communication status will be displayed in this component Refresh Now Description Status Upload message for asterisk channels Hangup hang up channel Transfer transfer channel 4 23 Options This component is used for administrator to manage the system it includes the following modules General Preferences Description default Global OutBound This is default global CallerID that is used for all outgoing calls Int CID when no other CallerID is defined that has a higher priority www atcom cn 29 When making outgoing calls the following rules are used to determine which CallerID will be used if they exist The first CallerID used is a CallerID set for the user making the call defined in the Users tab The second CallerID is the one that is set in the VoIP Trunks configuration if applicable The last CallerID used for outgoing calls is the Global CID defined in the Options tab ATCOM
16. 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 PEE www atcom cn 54 ATCOM Reference http atcom cn download html http www asteriskguru com http www openippbx org index php title Main_Page http www atcom cn 3J www atcom cn 55
17. internal system dialtone and to place calls from it as if they were placing a call from within the switch h Response Timeout set response timeout i Macro macronamelargllarg2 Executes a macro using the context macro macroname j Play Sound Plays back given file k Ringing Indicate ringing tone Set MusicOhHold Class select a music on hold m SayAlpha Say each character in the string including letters numbers and other characters one by one n SayDigits Say the digits one by one o SayNumber Say a number e g Six thousand five hundred and seventy two p Wait Pause dialplan execution for a specified number of seconds q WaitExten Wait for the user to enter a new extension for a specified number of seconds r To Destination go to destination S Set Language set language English Spanish French t To Directory go to directory u Dial an external Number Place a call outside the pbx using the selected trunk v AGI Executes an AGI compliant application w User Event Send an arbitrary event to the manager interface x Hangup Hang up the calling channel 3 Allow keypress events Must be voice menus have application Background file e x Background a music when keypress events 4 Advance edit Change dialplan for voice menus e x include 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 Backgroun
18. trunk an outgoing calling rule a dial plan a incoming calling rule and a user Here I will give the simple configuration steps which show how to make a call to outside for detail configuration you can refer to chapter 3 5 2 Create an Analog Trunk After logging into the web page of IP 2G4A please click on Trunks Analog Trunks I configure an analog trunk like the following Channels El n Trunk Name analogl Advanced Options Busy Detection Fes Busy Count Ring Timeout 2 CID Signalling 5 Answer on Polarity Switch D Hangup on Polarity Switch Progress Zone Use CallerID Fesi Caller ID Start D Caller Pulse Dial D Flash Timing Receive Flash Timing G Cancel Mi Ada You should hook on the Channels you need input the name of the trunk Others are default At last please click on Update button and click on Apply Changes button in up right corner of the main page Then you must restart the IP 2G4A Please pay attention to the red ellipse frame in the screenshot above channels five and seven are used for GSM 5 2 2 Create an Outgoing Calling Rule After logging into the web page of IP 2G4A please click Outgoing Calling Rules gt New Calling Rule I configure an outgoing calling rule like the following www atcom cn 35 Res CallingRule Calling Rule Name outi Pattern 2 5 r L Send to Local Destination Destina
19. 001 Q wane Agnes DialPlan CallerID Agnes D outBound CallerID Agnes Enable Voicemail for this User Voicelail Access PIN code 1234 Mailbox 6001 D Email Address ayi atcom com cn r Technology v siP v tax Analog Station EK E flash Q 750 rxflash OF Codec Preference First u z w Second zen ei Third deier Fourth r VoIP Settings M C Address D Line Number i SIP IAX Password NAT Can Reinvite DIMF Mode insecure very Other Options Enable Call Record Pickup Group Cancel IV Update At last please click on Update button and click on Apply Changes button in up right corner of the main page 3 Add up an Analog user 6005 After logging into the web page of IP 2G4A please click on Users Create New User I add a user 6005 like the following www atcom cn 33 Create New User H r General Extension 6005 Name agnes DialPlan DisiFlan M CallerID agnes OutBound CallerID agnes E Enable Voicemail for this User VoiceMail Access PIN code 1234 Mailbox 6005 Email Address ayi atcon com cn r Technology M sip v Tax nalog Station Fort 3 flash Q 750 rxflash 1250 Codec Preference First eiss w Second zen ei Third Sens Fourth xone Fifth None r VoIP Settings SIP IAX Password
20. 16 1 6 Extension 6020 AT 610 IP Address AT 620 192 168 1 130 172 16 1 2 IP Address Extension 6030 192 168 1 209 Extension 6001 t t i AT 620 AT 620 Ss IP Address IP Address Analog Phone 192 108 L 210 192 168 1 206 Extension 6005 Extension 6002 Extension 6008 Figure Network Topology In the network topology above user 6020 6001 6002 6008 will be registered to IP 2G4A analog phone 6005 is connected to FXS port of IP 2G4A After configuration it will realize the following function 1 The internal user 6005 and user 6001 or user 6002 and user 6001 can call each other directly 2 6005 and 6001 can dial out through IP 2G4A to PSTN 3 6005 and 6001 can get incoming calls from PSTN by IP 2G4A 4 6001 6002 6008 or 6005 are all communication with the mobile phone by IP 2G4A 5 User 6001 and 6030 can call each other through VoIP trunk although they are registered to different IP PBX 6 User 6020 6005 and 6001 can call each other directly although they are not in the same network segment 7 Voicemail 8 IVR 9 Conference 10 Ring Groups 11 Agents 12 Follow me 13 Call pickup www atcom cn 31 ATCOM 5 1 How to Make Internal Calls through IP 2G4A 5 1 1 Access to the Web Page of IP 2G4A by Browser After connecting IP 2G4A to LAN please open your browser of PC with windows OS and input the IP Address of IP 2G4A the default IP address is 192 168 1 100 then you can get the following screen
21. 4 2 Call from IP04 to IP 2G44 ii a rte na Raa cihn ney n n k ch vik a ARA AF Re Ea RARE V e aa Eae RE AR 5 5 VOICEMAIL casino e rens 5 6 HOW TO REALIZE THE IVR eene 5 6 1 Upload Voice Menu Prompts 5 6 2 Create Voice Menu 5 6 3 Add Incoming Calling Rules 7 CONFERENCE eet dee pee verde est eege ebe ge estate de ege BS RING GROUPS i e 46 EE EE 47 5 9 1 Create Users ds e EE 47 5 9 2 Create a Call QUEUES ici 1 5 i4sansi al haka calanes ya diver tend vai dd vec Hd a ven n V s D ED aes 48 5 TO Geif Ted ERR 49 5 1 1 GROUP CALL PICKUP T 50 AGRONY IMIS wicrectilccconcesacucsescccnsecasececscncecavacenccsencesccsnesesachoncdsadsusvesccasagessedseceucesadeles RR OR UU gan DDR UE ENTRA MERE 52 GLOSSARY EE 54 REFEREN ec M kana kak 55 www atcom cn 2 sulle ATCOM Contact ATCOM The Introduction of ATCOM Founded in 1998 ATCOM technology has been always endeavoring in the R amp D and manufacturing of the internet communication terminals The product line of ATCOM includes IP Phone USB Phone IP PBX VoIP gateway and Asterisk card Contact Sales Address District C east of 2nd floor 3 Crown industry buildings Chegongmiao Industry area Futian district Shenzhen China Tel 86 755 23487618 Fax 86 755 23485319 E mail sales atcomemail com Contac
22. 6001 but nobody pick up it after 30 seconds 6002 is ringing 5 11 Group Call Pickup This allows you to collect a call from any ringing phone that is in the same pickup group as you There are two kinds of methods one is that the phone itself has the function of pickup The other is that we can configure it in the GUI of IP 2G4A You can create users like the following www atcom cn 50 Edit User Extension 6001 r General Extension 6001 D Name Ton DialPlan DieiPien jw CallerID Ton outBound CallerID Tom m L Enable Voicemail for this User VoiceMail Access PIN code D Mailbox 6001 Email Address r Technology v szP v Tax Analog Station Y flash rxflash OP Codec Preference First eiss y Second csx el Third xone w Fourth Sens w r VoIP Settings SIP IAX Password NAT Jk Can Reinvite O DTMF Mode RFC2833 v insecure m Other Options O 3 Way Calling D O mm Directory D O cali Waiting O et O rs agent D C Enable Call Record Pickup Group v O Cancel bi Update please pay attention to the red ellipse frame all the users must in the same group Here I have created 6001 6002 which are both in the group 1 I have also created 6008 it can be in groupl also can not Then 6008 dial 6001 but we do not answer it at the same time 6002 dial 8 then 6002 can connect with 6008 Now we co
23. ATCOM ATCOM IPPBX IP 2G4A Product Guide Version 1 0 2010 10 14 Content CONTACT AT COM 3 CHAPTER 1 THE INTRODUCTION OF IP 2G44A ee eene eee enne nnne nnne nnn nnn aa ness aate senes ERR E eee s enin 4 CHAPTER 2 ACCESS TO THE IP 2GA4 A 1e ege on care oio dayan k s sno nera dara dak k saye n OR kanal SUR RR eka WA en U SR e ee 6 2 1 WEBPAGE lee E dE ie TEE 6 2 2 SUPPORT SSHPROTOCA EE H 2 3 CONSOLE PORT ACCESS TO IP 2G4A ee 7 CHAPTER3 GENERAL OPERATION OF IP 2G4A ccsscccossseccccsssccccsssecccessecccnssecccesseeccesseeceseeeceesseeeeens 8 CHAPTER 4 CONFIGURE IP 2G4A BY WEB GUI ee 10 A SYSTEM STATUS orerar iaa E aE a AEE EO EEE AAAA Ea a aa 10 4 2 CONFIGURE HARDWARE tats KAKA KK eite usas ss KE KA KA dena E KA k EEEE ka A k ass eae k kk 10 PES S STN N ee 11 4 3 1 Create Analog Trunks uuu Lusa enean anas nasse nita daas assess WE KE a daas ku Ku ku sa ku ku Wu assa 12 Re EE 13 A AS OUTGOING CALLING RULES 13 AS DIAEPLAN S 15 LN Ui nM Tcr 15 4AT RING GROUP m N l_ _ r r rl TV rn 17 MT eee 17 beke li er VE TEE 4 10 VOICE MENUS 4 11 TIME NTERVALS 4 12 INCOMING CALLING RULES 4 13 VOICEMAIL ek kk e e 4 14 CONFERENCING AVS FOLOWM cT EET 26 AAG DIRECTORY EE 27 ALU CALL FEATURES EE 27 Delen 28 4 19 VOICE
24. 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 j 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 Extension 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 4 2 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 you boot the IP 2G4A which will detect the FXO and FXS modules automatically the analog hardware component displays the modules which are detected correctly www atcom cn 10 Name Tone Region Description Select the tone region according to your country if it does not have your country s name in the dropdown list please ask your service operato
25. General Extension D Name agnes DialPlan DialPlan1 CallerlD agnes D OutBound CallerlD agnes V Enable Voicemail for this User VoiceMail Access PIN code 123 D Mailbox 6001 Email Address ujiayi atcom com cn r Technology dap Max Analog Station None D flash O ndlash Codec Preference First u law Second Gsm Third None Fourth None Fifth None r VoIP Settings MAC Address Line Number 1 SIP IAX Password NAT V Can Reinvite C D DTMF Mode nrcze33 insecure very r Other Options 3 Way Calling C In Directory C Call Waiting O CH D VI Is Agent Enable Call Record D Pickup Group 1 O Cancel I Update Like this I have also created 6002 6008 Then you must click System Status then you can see the following screenshots 6001 6008 Login Login Click the button of Login so that all the Agents have logined Then refresh the web you can see the page that all the agents have logined like the following screenshots 5 9 2 Create a Call Queue Please click Call Queues gt Create New Queue then you can configure the options like this screenshots www atcom cn 48 Extension agnets Strategy ringall Music On Hold defaut D LeaveWhenEmpty strict JoinEmpty no D Queue Options TimeOut 15 Wrapup Time 0
26. MENWIPROMPTS m R KEK E 28 4 20 SYSTEMIIINFO WE 28 7 21 BACKUP deen ee a e 29 2 22 ACTIVE CHANNELS 3 nino k e EI n s k dikele d k n a d k a eeni eia h k ke ed k w kek k a a e De R h kd kela b a kek b n kaw 29 A D3 ee 29 CHAPTER 5 AN APPLICATION CASE OF IP 2G4A ee 31 5 1 HOW TO MAKE INTERNAL CALLS THROUGH P 2G4 xxx eeeee kk kk kk ke keka ke ka k KAKA ka A KA KA KAKA KA KA KA KA KA KA KA KA 5 1 1 Access to the Web Page of IP 2G4A by Browser 5 1 2 Add up Users from Web Page of IP 2G4A 5 1 3 Register a SIP user 6001 in ATOI O nennen ener da nayan na A danl ness sess sensns sana www atcom cn 1 5 2 HOW TO COMMUNICATE WITH OUTSIDE THROUGH PSTN En 35 5 2 1 Create an Analog TEE 35 5 2 2 Create an Outgoing Calling RD 35 5 2 3 Selected the Outgoing Calling Rules in a Dial PION esses 36 5 2 4 Create WERE 36 5 2 5 Create INCOMING Calling Rules asarana k ran n cika nan dah AAEE ES ra nada daas ases Walk 37 5 3 HOW TO CALL EACH OTHER DIRECTLY FROM DIFFERENT NETWORK SEGMENT eene 37 5 4 HOW TO CALL THROUGH VOIP TRU NK 5 7o nasek ke e hurt b ke gavan kak R k ke dln Sak RE k ka Sek kese SE are walk 39 5 4 1 Call from IP 2G4A to IPO nissenana kanza inta nal Wiha aran xi kan kn as l nia nana Wa a daas asas sa k na 39 5
27. and default gateway of WAN port I configured a static IP Address 172 16 1 1 Subnet Mask 255 255 0 0 Default Gateway 172 16 1 254 DHCP option and so on Configure Port Range Forwarding you can use IAX2 you can configure Port Range Forwarding as 4569 IP address is 192 168 1 130 the IP Address of IP 2G4A Here I use IAX2 so I create a IAX2 user named 6020 2 Add an IAX user 6020 in IP 2G4A After logging into the web page of IP 2G4A please click on Users Create New User I configure 6020 like the following www atcom cn 37 ATCOM Edit User Extension 6020 r General Extension 6020 Name 6020 DialPlan DialPlani CallerID 6020 OutBound CallerID 6020 rr Enable Voicemail for this User VoiceMail Access PIN code D Mailbox 5020 Email Address r Technology M sip v zax Analog Station None Y flash rxflash Codec Preference First u law w Second GSM Third None ze Fourth None e Fifth None V None 88 m VoIP Settings MAC Address Line Number 1 ze SIP IAX Password 6020 NAT Can Reinvite O DTMF Mode RFC2833 w insecure very Y r Other Options 3 Way Calling Om Directory CT call Waiting O ert Is Agent Pickup Group 1 Q Cancel bi Update At last please click on Update button and click on Apply Changes button in up rig
28. ch other among 6020 6001 and 6005 directly 5 4 How to Call through VoIP Trunk 5 4 1 Call from IP 2G4A to IP04 In order to call from IP 2G4A to IP04 I will create a user in IP04 for the SIP IAX trunk in IP 2G4A create a SIP IAX trunk an outgoing call rule and a dial plan in IP 2G4A But pay a attention that at the same time a port of the router where the IP04 in must be directed to the IP04 1 Add an user 6200 it will be used as SIP trunk in IP 2G4A in IP04 after logging into the web page of IPO4 please click on Users Create New User I add the user 6200 like the following PEE www atcom cn 39 Create Hew User en r General CallerID trunk utBound CallerID r L Enable Voicemail for this User VoiceMail Access PIN code D Mailbox 6007 Email Address r Technology v srP v rax Analog Station None v flash 750 rxflash Q 1250 Codec Preference First u law wi Second csx ei Third diene zi Fourth yons w Fifth yons iv r VoIP Settings MAC Address Line Number v SIP IAX Password NAT Can Reinvite O DTMF Mode RFC2E33 v insecure very r Other Options E 3 Way Calling O In Directory O call Waiting L ert O Is Agent L Enable Call Record Pickup Group se O Cancel V Update At last please click on Update button and click on Apply Changes button
29. d Hunt groups Please select the Ring Groups option from the vertical menu on the left of the main page then you can get the following screen insecure invite Name Description Type default Ring Group Ring group name use in pbx Str Name Extension for Ring group No dial the No if you wantto join change Int 6400 this ring group boundary value in options Ring Group The ring group of numbers EXT1 EX Members T2 EXT3 eat Available The entire Users EXT1 EX Users T2 EXT3 is Strategy Ring all simultaneously Ring in Ring in sed vtl Order ting Order all Extensions Seconds to Seconds to ring each member Time 20 ring each member If not If not answered go to hang up hang up the calling channel Hang up Hang up answered Goto Operator Go to operator o Extension a call to user Operator Voicemail Go to IVR Conference join a conference 3l Call queue Go to a call queue 1 ring group application Dial channel type EXTEN l channel type EXTEN 120i 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 4 8 Music on hold Music On Hold lets you customize audio tracks for different queues parked calls etc Name Description Type default Upload an 8 KHz Support codec g711a g711u Upload Mono Music file New music on hold Add a new music on hold
30. d fpm sunshine exten s 8 Voicemail 6002 u exten 1 1 Goto voicemenu custom 2lsl1 exten 2 1 Voicemail 6002 u exten 5 1 Goto voicemenu custom 3lsl1 Want to control music on hold play time www atcom cn 20 ATCOM include 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 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 2IsI1 exten 2 1 Voicemail 6002 u exten 5 1 Goto voicemenu custom 3lsl1 4 11 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 Name Description Type default Time A name for the time interval Str Interval Name By day of Choice an available day of week for the time interval Mon Tue week Wed Thu Fri sat Sun By Days of Choice some available days of month for the time Dateof a 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 00Imo
31. d time the message was left Hello VM NAME you received a message lasting VM DUR at VM DATE from VM_CALLERID This is message VM_MSGNUM in your voicemail Inbox b SMTP Settings Name Description Type default STMP server The IP address or hostname of an SMTP server that Str your box may connect to without authentication in order to send e mail notifications of your voicemails i e mail yourcompany com Port The port number on which the SMTP server is Str running generally port 25 Use TLS Use TLS Transport Layer Security when Check box unCheck communicating with the SMTP server Authentication Does the SMTP Server requite authentication Check box unCheck Username The username of a valid account on the STMP server Str Password The password of a valid account on the STMP server Str 1 Config file etc ssmtp ssmtp conf 2 Note Firmware after that starts support Gmail 4 14 Conferencing 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 Name Description Type default Extension This is the number dialed to reach this Conference Int 6300 Bridge Marked Admin If the conference bridge is to have marked users or Int user Extension
32. dard 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 El 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 opposed 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
33. een reached a caller will hear a busy tone and advance to the next calling rule after attempting to enter the queue Auto full Defining this option causes the Queue when multiple calls checkbox are in it at the same time to push them to Agents simultaneously Thus instead of completing one call to an Agent at a time the Queue will complete as many calls simultaneously to the available Agents Auto pause Enabling this option pauses an agent if they fail to answer checkbox acall 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 login extension Report Hold Enabling this option causes Asterisk to report to the checkbox Time Agent the hold time of the caller before the caller is connected to the Agent KeyPress If a caller presses a key while waiting in the queue this choice Events setting selects which voice menu should process the key press Agent This selection shows all Users defined as Agents in their checkbox Userconf Checking a User here makes them a member of the current Queue 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 4 10 Voice Menus Like most organization we would like to redirect all of the incoming calls automa
34. empt to negotiate the endpoints to route the media stream directly bypassing asterisk It is not always possible for asterisk to negotiate endpoint to endpoint media routing DTMF Mode Set default dtmfmode for sending DTMF info SIP ComboBo rfc2833 INFO messages inband Inband audio requires 64 kbit x codec alaw ulaw auto Use rfc2833 if offered inband otherwise 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 Queue selected Not Member Agent selected Pickup Group If a user called A and another user called B in the same selected Not group A can pick up the phone taking the place of B selected 1 Analog Station When you want to create Analog Users please choose the FXS ports 2 Codec Preference Support g711u law g711a law g729 GSM www atcom cn 16 ATCOM 3 Attension in the textbox of Extension the value you set is limited to a range you can adjust the range in the Options option to meet your requirement 4 7 Ring Groups Define Ring groups to dial more than one extension simultaneously or to ring more than one phone sequentially This feature may also be calle
35. ermanent IP address for IP 2G4A you can try to set itin web GUI for detail steps please refer to chapter 3 www atcom cn 7 ATCOM Chapter3 General Operation of IP 2G4A 1 Backup When you log in the web of IP 2G4A Click on Backup you can see the button of Create New Backup then you can Backup the current system 2 System Update When you log in the web of IP 2G4A click on Options gt Advanced Options gt Show Advansed Options After click on Show Advanced Options in the web you can see the advanced options in the vertical menu on the left of the main page Click on Firmware update you can see the following parameters in the table Parameter Name Description Type Default HTTP URL The http path of the firmware file Textbox Null TFTP Server The IP address of TFTP Server where the Textbox Null firmware file in File Name Specify the name of your ulmage md5 Textbox Null firmware file make sure to use md5 version only Reset Configs Select this box if you wish to reset to factory Selected Not selected defaults This will ensure a clean update and is highly recommended If you want to upload sound file upload backup files and so on you can refer to the link http www atcom cn downloads IPPB X ATCOM 20IPPBX 20Series 20Product 20Uper ade 20Guide V 1 0 EN pdf 3 Network After click on Options gt Advanced Options gt Show Advanced Options please select Network Set
36. ete 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 _INXXNXXXXX would represent a three digit area code plus phone number proceeded by a one 3 Note You ll 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 you have multiple SIP trunks from the same provider you ll want to set this pattern to whatever you specified as Contact Extension 4 13 Voicemail When you call someone who does not answer the call you can leave a voice message for the called party if the called party supports voice mail Name Description Type Default Extension for defines the extension that Users call in order to access NO 6750 checking their voicemail accounts messages Direct Check this to enable direct voicemail dial For instance Check box unCheck Voicemail if John s extension is 6001 you would be able to Dial directly dial into John s voicemailbox by dialing 6001 to leave him a message www atcom cn 22 ATCOM review message before it is submitted as a new voicemail message Max greeting Set the maximum number of seconds for a User s No 30 in seconds voicemail greeting Dial 0 for Enable Callers to exit the voicemail application
37. ew step you can see the screen display like the following screenshots then you can select your own www atcom cn 43 ATCOM voice prompt Here I use the voice prompt named 04 You can upload the voice prompt like 5 6 1 Edit YoiceNenu voicemenu custorm 1 Name IVR Advanced Edit Extension 7001 CO G allow Dialing Other Extensions Actions Answer the call Plays back g 04 D Allow key 1foram 2 forpm 1 yes 2 no CHANGES asterisk core en 1 4 13 CHANGES asterisk core en 1 4 19 CHANGES asterisk extra en 1 4 8 CREDITS asterisk core en 1 4 13 Third hook on the option Allow KeyPress Envents then you can configure the operation from 0 to which you need Please click on save button and click on Apply Changes button in up right corner of the main page Here I configure that press 0 then call 6001 press 1 then call 6002 press 2 then call 6008 Of course 6001 6002 6008 have registered BE IKEDZJIDIZGZZGZDZF NIBI OLILIIDA I www atcom cn 44 Edit YoiceMenu voicemenu custon 1 Name IVR Advanced Edit Extension 7001 O Allow Dialing Other Extensions Actions D Answer the call Play 04 amp Donot Listen for KeyPress events Add new Step Select an Option E v Allow KeyPress Events Goto User 6001 Goto User 6002 Goto User 6008 5 6 3 Add Incoming Calling Rules After configure the Voice Menu you must configure the
38. fox instead of IE because there are compatible issues Then input the default Username admin Password atcom www atcom cn 6 ATCOM 2 2 Support SSH protocal Logging into IP 2G4A by SSH you can configure IP 2G4A by Linux command 2 3 Console Port Access to IP 2G4A If you do not have network connection between IP 2G4A and PC you can try to access to IP 2G4A by console port Please try to do as the following steps 1 Connect the console port of IP 2G4A to your PC s console port with RS232 console cable 2 Run your HyperTerminal and set up the console port like the following Bits per second 115200 Data bits 8 Parity None Stop bits 1 Flow control None 3 Change the IP Address by HyperTerminal The default IP address of IP 2G4A is 192 168 1 100 Your network may have a different IP address segment such as 192 168 10 xx In this situation you cann t access to IP 2G4A by putty and browser if you do not change the IP 2G4A IP address So you have to change the IP address for IP 2G4A by HyperTerminal to make it in the same network segment as your LAN After you have accessed to IP 2G4A by HyperTerminal please use the following command to change the IP address for IP 2G4A root gt ifconfig ethO 192 168 1 151 the IP address what you want to set for IP 2G4A By this way the IP address you set for IP 2G4A is temporary it will recover to the original default IP address after rebooting If you want to give a static and p
39. g this option enabled CallerId Boolean yes detection Caller ID Start This option allows one to define the start of a ComboBox Ring 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 mode dialing Boolean No instead of DTMFE 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 in Textbox millseconds that is generated for a flash 750 operation Receive Flash Flash Time defines the time in milliseconds Textbox Timing that is generated for a flash operation 1250 www atcom cn 12 ATCOM 1 Trunk name unique label to help you identify the trunk when listed in outgoing calling rules and incoming calling rules 4 3 2 VoIP Trunks A VoIP service provider VSP that you have signed up with is also a trunk Via the VoIP trunk you can dial via the VoIP service to reduce your cost when making international calls You can set up the VoIP trunk to make calls to the PSTN or other VoIP network depends on the service you use You can also 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 Name Description Type Defa
40. ht corner of the main page 3 Setup AT 610 and register an IAX2 user 6020 Please select the VOIP option then select the IAX2 option I register the IAX2 user 6020 as the following illustration IP Phone See H ATCOM Current Status Network VOIP Advanced Dial peer Config Manage Update System Manage SIP1 IAX2 Configuation SIP2 IAX 2 IAX2 Register Status Unregistered lAX2 Server Addr 172 16 1 1 lAX2 Server Port 4569 Account Name 6020 Account Password ecos Phone Number 6020 Local Port 4569 Voice Mail Number 0 Voice Mail Text Echo Test Number Echo Test Text Refresh Time Enable Register Enable G 729 www atcom cn 38 Pain ATCOM Please pay attention to the red ellipse frame in the screenshot above it is the IP address of the router After configuring please click on the APPLY button Attention here you must register IAX2 user instead of SIP user because the user 6020 is not in the same network segment as IP 2G4A If you use SIP user you need configure the SIP Setting where is in Options gt Adance Options gt Show Advance Options you can configure the two options in the red ellipse frame in the screenshot like this General DebugNotify NAT Misc Jitter Buffer Codecs Extern Host Extern Refresh 10 Local Network Address D NAT mode GO Allow RTP Reinvite Now you can call ea
41. in up right corner of the main page Please pay attention to the Name and OutBound CallerID in the red ellipse frame if the user uses for a trunk the two options are null so that the caller ID on the phone is the calling party Then Add a user 6030 in IP04 for AT 620 the way is the same as adding 6001 2 Add a VoIP trunk in IP 2G4A after logging into the webpage of IP 2G4A please click on Trunks gt VOIP Trunks New SIP IAX Trunk I configure a SIPtrunkl like the following Create New SIP IAX trunk Type SP si Provider Name outsidePBX Username 6200 Fromuser Fromdomain Password Contact Ext Insecure Type Please pay attention to the red ellipse frame the Hostname is the public IP address where the IP04 is After configuring please click on Add button and click on Apply Changes button in up right www atcom cn 40 ATCOM corner of the main page Attention the option of Fromuser default is null 3 Create an outgoing calling rule in IP 2G4A after logging into the webpage of IP 2G4A please click on Outgoing Calling Rules New Calling Rule I configure an outgoing call rule like the following New CallingRule x Calling Rule Name D toPBX Pattern E O Send to Local Destination Destination r Send this call through trunk Use Trunk CD sese v Strip Q 1 digits from front and Prepend these digits before dialing O Use FailOver T
42. is option allows the user to specify the type of Message Waiting indicator detection to be done on trunk FXO interfaces ComboBox none 4 3 Trunks To receive calls from PSTN and make calls to the outside world you have to use trunk Please select the Trunks option from the vertical menu on the left of the main page www atcom cn 11 4 3 Create Analog Trunks we ie ATCOM Analog trunk is associated with FXO port and it will call outside by PSTN line Click on New Analog Trunk then we can see the parameters which are in the following table in the web Name Description Type Default Channels Display the FXO or GSM 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 end hang Boolean up or for detecting busy signal Yes busycount If Busy Detection is enabled it is also possible Int 3 to specify how many busy tones to wait for before hanging up Ring Timeout Thrunk FXO devices must have a timeout Int 8000 to determine if there was a hangup before the line was answered answeronpolarit If this option is enabled the reception of a Boolean yswitch polarity reversal will mark when a outgoing no call is answered by the remote party hanguponpolarit In some countries a polarity reversal is used Boolean yswitch to single the disconnect of a phone line no Use CallerID Enablin
43. lar trunk may have a secondary trunk defined trunk but one wants calls to use the PSTN If a user s primary trunk is a VoIP when the VoIP trunk isn t available this option is a good idea selected no select Fail over trunk Choose the trunk ComboBox ComboBox 1 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 INXXNXXXXX would represent a three digit area code plus phone number proceeded by a one 2 Strip Allows the user to specify the number of digits that will be stripped from the 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 calls www atcom cn 14 ATCOM are placed onto analog lines and the PSTN so one should strip 1 digit from the front before the call is placed LA The way of outgoing calling rules works Every time you dial a number asterisk will do the following in strict order
44. ling rule Type Textbox a _ aw ATCOM Default Null Pattern The dialing rule Textbox Null Send to Local Destination If this option is checked and Destination is defined calls matching the specified pattern may be sent to a local extension selected no select Destination Choose the Local Destionation User VoiceMenu Hungup ComboBox Null Use trunk Defines the Trunk that calls matching the specified pattern will be placed through ComboBox Null Strip Allows the user to specify the number of digits that will be stripped from the front of the dialing string before the call is placed via the trunk selected in VUse Trunk One might Textbox Null Prepend these digits Allows the user to specify digits that are prepended before the call is placed via the trunk Ifa 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 trunks to provide a slight delay before dialing Textbox Null Use Failover Trunk Failover trunks can be used to make sure 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 regu
45. ll the users are in the conference You can see the detail in 4 14 and configure it as your need 5 8 Ring Groups 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 You can click Ring Groups New Ring Group then you can configure it like the following screenshots Of course 6001 6002 have registered 6008 have registered Then 6008 dial 6400 you can hear 6001 6002 are ringing simultaneously If you want the users are ringing sequentially you can configure the strategy as Ring in Order 9 www atcom cn 46 New RingGroup RingGroup Name Extension for this ring group Ring Group Members 6001 SIP agnes 6002 SIP Peter ATCOM Available Users 6001 1 amp X2 agnes 6002 1AX2 Peter 6008 SIP Rose 6008 1AX2 Rose Ring Group Options Strategy Seconds to ring each member If not answered Goto 5 9 Agents Ring all simultaneously 1 20 Hangup w Cancel Mi Save When you need the function of Agents you need complete the following two steps 5 9 Create Users as Agents You can create users like 5 1 2 but hook on the option of Is Agent like the following screenshots please pay attention to the red ellipse frame www atcom cn 47 Edit User Extension 6001
46. m cn 4 5 Applications SOHO SMB telephony system Hosted service IVR system 6 Interface 1 X RJ45 port X Power port 1 X RS232 port 4 X FXO FXS ports 2 X GSM channels 7 Measurement and Weight oMo ATCOM Inner box 225 120 30mm G W unit 0 79KG Carton MEAS 456 442 362 mm Units per Carton 21 units CTN G W CTN 18 KG CTN 8 Package Item Quantity IP 2G4A 1 RS232 module 1 Power Adapter 1 Manual disk 1 For the usage of IP 2G4A in VoIP field you can refer to the following network topology www atcom cn ATCOM AT 620 x ui e a Analog Phone Chapter 2 Access to the IP 2G4A You need a PC to access to the IP 2G4A there are four ways for you to access the IP 2G4A 1 Web page access by browser 2 SSH access by putty 3 Access by browser with Fallback IP Address 4 Console port access by RS232 console cable In order to access to IP 2G4A by the first three ways you have to check that if your network connection between IP 2G4A and PC is OK If you do not have network connection between IP 2G4A and PC you can try to use the last way to access to IP 2G4A and change the IP address for IP 2G4A 2 1 WebPage Access by Browser It is the most convenient and common way to access the IP 2G4A you just need to open your browser and input the IP address of IP 2G4A WAN port the default IP address is 192 168 1 100 You d better use Fire
47. marked user has joined 1 Conferencing application MeetMe confno options pin Enters the user into a specified MeetMe conference ex MeetMe EXTEN IMsIqwxaA disable you are currently the only person in this conference message for first member a set admin mode A set marked mode b run AGI script specified in 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 i announce user join leave with review T 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 Ip allow user to exit the conference by pressing P always prompt for the pin even if it is specified Ou quiet mode don t play enter leave sounds TY Record conference records as MEETME_RECORDINGFILE using format MEETME RECORDINGFORMAT s Present menu user or admin when is received send to menu t set talk only mode Talk only no listening T set talker detectio www atcom cn 25 v
48. mpleted the Group Call pickup function www atcom cn 51 an n ATCOM Acronyms 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 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 i
49. n 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 frequency 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
50. n sumll 3 1 January February March April May June july August September October november December all time intervals timeinterval date Imon tuel Monday to Tuesday of weekl 4 12 Incoming Calling Rules 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 www atcom cn 21 ATCOM Name Description Type default Trunk Choice the trunk for the incoming rule analog server provider voip Time Interval Choice the time interval for the incoming rule Choice Non timeinterval matched Pattern Pattern of the incoming rule Dialplan S matched Destination Incoming to destination users voice IVR mail ring group 1 A trunk support a number of this time intervals to support a number of Destination 2 Pattern All patterns are prefixed by the on 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 compl
51. r which kind of tone region is used in your area Type ComboBox ATCOM Default United Status North America Module Name The name of Module Textbox wctdm24xxp Opermode Specifies On Hook Speed Ringer Impedance Ringer Threshold current Limiting TIP RING voltage adjustment minimum Operational Look Current and so on Please choose your country or your nearest neighboring country ComboBox USA a law override Specifies the codec to be used for analog line ComboBox ulaw fxs honor mode This option allows the user to determine if they would like opermode characteristics applied to trunk FXO modules only or both trunk FXO and station FXS modules ComboBox FXO modules boostringer This option allows the user to define whether they require normal ringing voltage 40v or maximum ringing voltage 89v or analog phones attached to station FXS modoules ComboBox nomal fastringer This option sometimes used in conjunction with the Low Power Option allows the user to increase the ringing speed to 25HZ ComboBox nomal lowpower This option generally used in conjunction with the Fast Ringer Option allows the user to set the peak voltage during Fast Ringer Operation to 50V ComboBox nomal ring detect This option allows the user to choose from normal ring detection or a full wave detection ComboBox standard MWI mode Th
52. rols 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 no 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 www atcom cn 18 TimeOut How many seconds an Agent s phone will ring before the Time 15 Queue tries to ring the next Agent Wrapup How many seconds after the completion of a call an Agent Time 0 Time will have before the Queue can ring them with a new call The default is 0 which is no delay Max Len How many calls can be queued at once This count does Int 0 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 b
53. rom incoming callers Also known as IVR menus or Digital Receptionist www atcom cn 42 ATCOM 5 6 1 Upload Voice Menu Prompts If you want to configure the IVR which you need you must upload your voice prompt You can click on Voice Menu Prompts you 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 You can click the button of Record a new Voice Menu prompt to record a voice prompt or you can click the button of Upload a Voice Menu prompt to upload your voice prompt 5 6 2 Create Voice Menu You can configure IVR like this click on Voice Menus Create Voice Menus then you can configure the IVR like the following pictures First selected the option Answer on the Add new step then click the Add new step Name IVR Extension 7001 O G Allow Dialing Other Extensions Actions Authenticate Background ES 7 Cancel v Wi TO Qcancel Z Save Congestion DigitTimeout DISA ResponseTimeout Macro Playback Ringing Set MusicOhHold Class SayAlpha SayDigits SayNumber Wait WaitExten Goto Destination Set Language Second selected the option Background on the Add new step then click the Add n
54. runk fail over Trunk v Strip digits from front and Prepend these digits before dialing O cancel M Save After configuring please click on Save button and click on Apply Changes button in up right corner of the main page 4 Hook on the outgoing calling rules in dial plan in IP 2G4A after logging into the webpage of IP 2G4A please click on Dial Plans Edit DialPlan and then hook on the outgoing calling rules Edit DialPlan DialPlan Name DialPlanl Include Outgoing Calling Rules v Include Local Contexts v default lvlparkedcalls V conferences Miringgroups Mivoicemenus queues vlvoicemailgroups V directory OCancel M Save After configuring please click on Save button and click on Apply Changes button in up right corner of the main page In configuration screens of 6001 and 6005 Now you can call from 6001 or 6005 to 6030 by dialing 86030 5 4 2 Call from IP04 to IP 2G4A In order to call from IPO4 to IP 2G4A I will create a SIP user in IP 2G4A for the SIP trunk in IP04 like 4 5 1 and then create a SIP trunk an outgoing call rule and a dial plan in IP04 www atcom cn 41 ATCOM 1 Adda user 6008 in IP 2G4A like 5 1 2 2 Create a SIP trunk in IP04 named out 3 Configure the router 4 Create an outgoing calling rule in IP04 named toIP 2G4A Here I use Pattern 4 5 Hook on the toIP 2G4A option in DialPlan After configuring please click on
55. t Technical Support Tel 86 755 23481119 E mail Support atcomemail com Website Address http www atcom cn ATCOM Wiki Website http www openippbx org index php title Main_Page Download Center http www atcom cn download html www atcom cn 3 sugli ATCOM Chapter 1 the Introduction of IP 2G4A 1 Overview of the IP 2G4A The IP 2G4A is a complete Asterisk Appliance with combination of GSM and Ananlog 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 ATCOM GSM 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 GSM and PSTN network 2 Hardware CPU 400MHz Blackfin 532 Chip 2 x GSM ports and four analog ports NAND flash 256 M SDRAM 64M 3 System Open Source uClinux 4 Function features PSTN GSM ISDN Support g711 g729 gsm codec Voicemail Voicemail groups 3 way Calling Conferencing Follow Me Call Feature In directory Call Waiting Call Queues Pickup Group Ring Group Is Agent Music On Hold Voice Menus Voice menus Prompts Time intervals Backup Update www atco
56. tbox Null another internal user OutBound Caller ID that would be applied for out bound calls Textbox Null CallerID from this user Note that your ability to manipulate your outbound Caller ID may be limited by your VoIP provider Enable Check this box if the user should have a voicemail Selected Not www atcom cn 15 Qr ATCOM Voicemail for account selected this User VoiceMail Voicemail Password for this user Textbox Null Access PIN code 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 or selected selected is a SIP device IAX Check this option if the User or Phone is using IAX or selected selected is an IAX device Analog If this user is attached to an analog port on the system ComboBo Null Station please choose the port number here D Codec Choose priority codec ComboBo u law GS Preference X M 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 ports Can Reinvite By default Asterisk will route the media steams from selected Not SIP endpoints through itself Enabling this option selected causes asterisk to att
57. the DTMF sequence defined on the Feature Codes page K Option Allow the calling party to enable parking of the call Check box Uncheck by sending the DTMF sequence defined on the Feature Codes page 4 18 VoiceMail Groups Define VoiceMail Groups to leave a voicemail message for a group of users by dialing an extension Name Description Type default VoiceMail Default Voicemail Group s Extension Int 6601 Group s Extension Label The label of Voicemail Group s Extension Str User MailBoxes The entire user Mailboxes Check boxs 4 19 Voice Menu Prompts This component is used for recording custom voice menu Description default Voice menu File Name Str prompts dial this User Extension to record a new voice prompt Choice 6001 Upload a Voice menu prompt Choice 4 20 System Info From this component you can easily get the basic system information it includes a General OS Version Linux version for PBX Uptime uptime for PBX www atcom cn 28 ATCOM Version Details asterisk GUI Firmware version for PBX Server Date amp TimeZone time now for PBX Hostname name for PBX b Network network message for PBX Eth0 9 fill back IP for PBX vlan IP c Disk Usage Filesystem File system of PBX 1k blocks A total of system modules Used Used of system modules Available Available of system modules Use Percentage Mounted on The specified directory
58. 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 first version of the protocol to be widely deployed NAT Network Address Translation DTMF Dual Tone Multi Frequency GSM Global System for Mobile Communications PE www atcom cn 53 alle ATCOM 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 stan
59. tically 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 Name Description Type default Name A name for the voice menus Str Extension If you want this Voicemenu to be accessible by dialing an No 7001 extension then enter that extension number Actions A sequence of actions performed when a call enters the Dial plan 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 listen for checkbox KeyPress DTMF input from the caller and define the actions that Events occur when a user presses the corresponding digit Advance edit Advance edit for the voice menu Dial plan script www atcom cn 19 Qm ATCOM 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 Answer Answer a channel if ringing b Authenticate This application asks the caller to enter a given password in order to continue dialplan execution c Background Play an audio file while waiting for digits of an extension to go to d Busy Tone Indicate the Busy condition e Congestion Indicate the congestion condition to the calling channel f Digit Timeout set digit timeout g DISA Password Allow someone from outside the telephone switch PBX to obtain an
60. tings option from the vertical menu on the left of main page You can set IP address Subnet mask Gateway DNS what you want like the following www atcom cn 8 DHCP Hostname Domain MAC IP address Subnet mask Gateway DNS NTP IP address Subnet mask DHCPD Start IP End IP Lease Time Subnet Mask Gateway no r IP2G4A 192 168 1 100 255 255 255 0 192 168 1 1 192 168 1 1 pool ntp org 192 168 10 1 255 255 255 0 yes 192 168 10 2 192 168 10 254 86400 255 255 255 0 192 168 10 1 ATCOM Please click save button in your page to save your setting and reboot the IP 2G4A Attention you need configure the IP address Subnet mask Gateway and DNS at WAN Interface so that the network connects successfully The option of LAN Interface is used for Routing functions here you needn t configure it www atcom cn ATCOM Chapter 4 Configure IP 2G4A by Web GUI 4 System Status In the system status screen it displays the functions you configured such as trunks extensions conference and so on The following table is the options description of trunks Name 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 IP Address port The register status of trunks include three kinds Unregistered Request Sent Registered 2 The type of trunks VoIP trunk including SIP and IAX
61. tion r Send this call through trunk Use Trunk D ansics v Strip D 1 digits from front and Prepend these digits before dialing ES C Use Failover Trunk D fail over Trunk v Strip digits from front and Prepend these digits before dialing Q Cancel Iv Save At last please click on Save button and click on Apply Changes button in up right corner of the main page 5 2 3 Selected the Outgoing Calling Rules in a Dial Plan After logging into the web page of IP 2G4A please click on Dial Plans Edit DialPlan1 then selected the name of the outgoing calling rules like the following Edit DialPlan DialPlan Name DialPlanl Include Outgoing Calling Rules vlouti Include Local Contexts Midefault Vlparkedcalls Miconferences ringgroups Mivoicemenus M queues lVlvoicemailgroups V directory O Cancel Ki Save At last please click on Save button and click on Apply Changes button in up right corner of the main page 5 2 4 Create a User I will use the user 6001 I created before Now I can call out with prefix 9 if the caller number is 10086 I will dial 910086 If you use GSM ports we will communicate with outside by Mobile Phone Network www atcom cn 36 m ATCOM 5 2 5 Create Incoming Calling Rules In order to get an incoming call from outside with IP 2G4A you need set Incoming Calling Rules Of course the precondition is that you have set
62. 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 atcom cn 52 ATCOM 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 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
63. ult Type You can select SIP or IAX type to meet your ComboBox SIP need Provider Name A unique label to help you identify this trunk Textbox Null when listed in outbound rules incoming rules etc Hostname The IP Address of the server which you want to Textbox Null connect Username the username that your service provider Textbox Null configured Fromdomain The domain of the server which you want to Textbox Null connect Password the password that your service provider Textbox Null configured for the user Contact Ext Textbox N Insecure Type The insecure type of the trunk transferring data ComboBox very 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 incoming INVITE messages 4 4 Outgoing Calling Rules Outgoing calling rules is used to route an outgoing call when you make an external call 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 www atcom cn 13 Name Calling Rule Name Description The name of the Cal
64. up a trunk a destination which include Voice Menu Voice mail a User Extension etc and a Time Interval After logging into the web page of IP 2G4A please click on Incoming Calling Rules New Incoming Rule I configure an incoming calling rule like the following d analogl w Time Interval None no TineIntervals natched s i Pattern D S Destination User Extension 6001 GCancel bi Update At last please click on Update button and click on Apply Changes button in up right corner of the main page Here I use analog trunk 1 you can choose you need Then when the outside makes a incoming call it will be sent to user 6001 through analog 1 you may configure the analog trunk for GSM by wireless in a similar way Attention Here if you choose the five channel Outgoing Calling Rule and Incoming Calling Rules are both use the channel 5 Then you can communication with the mobile phone For example I configure the Outgoing Calling Rule as _5 Then use the channel 5 and the number is 158xxxxxxx2 Incoming Calling Rules be pointed to 6001 Then I can dial a mobile phone number with prefix 5 others can dial 158xxxxxxx2 to connect us 5 3 How to Call Each Other Directly from Different Network Segment Take the user 6020 6005 and 6001 for example I will configure router users and IP 2G4A then the three users can call each other directly 1 Setup router Please configure the router IP address subnet mask

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