ATCOM® IPPBX IP4G Product Guide Version: 1.
Content CONTACT ATCOM ............................................................................................................................... 3 CHAPTER 1 THE INTRODUCTION OF IP4G ........................................................................................... 4 CHAPTER 2 ACCESS TO THE IP4G ........................................................................................................ 6 2.1 WEBPAGE ACCESS BY BROWSER .................................................................
5.2 HOW TO COMMUNICATE WITH OUTSIDE ................................................................................................ 34 5.2.1 Create an Analog Trunk ......................................................................................................... 34 5.2.2 Create an Outgoing Calling Rule ........................................................................................... 35 5.2.3 Selected the Outgoing Calling Rules in a Dial Plan ................................................
Contact ATCOM The Introduction of ATCOM Founded in 1998, ATCOM technology has been always endeavoring in the R&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.
Chapter 1 the Introduction of IP4G 1. Overview of the IP4G The IP4G 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.
5．Applications SOHO/SMB telephony system Hosted service IVR system 6. Interface 1 X RJ45 port. 1 X Power port. 1 X RS232 port. 4 X GSM channels. www.atcom.
For the usage of IP4G in VoIP field, users can refer to the following network topology. Chapter 2 Access to the IP4G You need a PC to access to the IP4G, there are four ways for you to access the IP4G: 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 IP4G by the first three ways, Users have to check that if your network connection between IP4G and PC is OK.
2.2 Support SSH protocal Logging into IP4G by SSH, users can configure IP4G by Linux command. 2.3 Console Port Access to IP4G If it does not connect between IP4G and PC, users can try to access to IP4G by console port. Please try to do as the following steps: 1. Connect the console port of IP4G to your PC’s console port with RS232 console cable. 2. Run the 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.
Chapter3 General Operation of IP4G 1. Backup When users log in the web of IP4G, Click on Backup Create New Backup, then they can Backup the current system. 2. System Update When users log in the web of IP4G, click on Options Advanced Options Show Advansed Options , After click on Show Advanced Options in the web, users can see the advanced options in the vertical menu on the left of the main page. Click on Firmware update , users can see the following parameters in the table.
Please click save button in the page to save the setting and reboot the IP4G. Attention: users 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 users needn’t configure it. www.atcom.
Chapter 4 Configure IP4G by Web GUI 4.1 System Status In the system status screen, it displays the functions users configured, such as: trunks, extensions, conference and so on. The following table is the options description of trunks. Name Status Description 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 1.
Name Tone Region Description Select the tone region according to your Type ComboBox Default United country, if it does not have your country’s Status/North name in the dropdown list, please ask your America service operator which kind of tone region is used in your area Module Name The name of Module Textbox wctdm24xxp Opermode Specifies On Hook Speed, Ringer Impedance, ComboBox USA Ringer Threshold, current Limiting ,TIP/RING voltage adjustment, minimum Operational Look Current and so on.
4.3.1 Create Analog Trunks Analog trunk is associated with FXO port, and it will call outside by PSTN line. Click on New Analog Trunk , then users can see the parameters which are in the following table in the web. Name Channels Description Display the FXO or GSM modules Type selected Default 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.
1.Trunk name: unique label to help users identify the trunk when listed in outgoing calling rules and incoming calling rules. 4.3.2 VoIP Trunks A VoIP service provider (VSP) that users have signed up with is also a trunk. Via the VoIP trunk users can dial via the VoIP service to reduce their cost when making international calls. Users can set up the VoIP trunk to make calls to the PSTN or other VoIP network. Users also can use the VoIP trunk to link headquarter and branch offices for free internal calls.
Name Calling Rule Name Description The name of the Calling rule Type Textbox Default Null Pattern The dialing rule Textbox Null Send to Local If this option is checked and Destination is selected no select Destination defined, calls matching the specified pattern ComboBox Null ComboBox Null Textbox Null Textbox Null selected no select ComboBox ComboBox may be sent to a local extension.
are placed onto analog lines and the PSTN, so one should strip 1 digit from the front before the call is placed. 3. The way of outgoing calling rules works: Every time you dial a number, asterisk will do the following in strict order: • Examine the number you dialed. • Compare the number with the pattern that you have defined in your first outgoing rule and if matches, it will initiate the call using that trunk.
Voicemail for account selected this User VoiceMail Voicemail Password for this user Textbox Null Mailbox Voicemail Mailbox for this user Textbox Null Email Address The e-mail address for this user Textbox Null SIP Check this option if the User or Phone is using SIP or selected selected selected selected Null Access PIN code is a SIP device IAX Check this option if the User or Phone is using IAX or is an IAX device Analog If this user is attached to an analog port on the system, Com
3. Attension: in the textbox of Extension, the value users set is limited to a range, they can adjust the range in the Options option to meet their requirement. 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 called Hunt groups.
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 Please select the Call Queues option from the vertical menu on the left of the main page, then users can get the following screen: Name Extension Name Strategy Music On Hold LeaveWhen Empty JoinEmpty www.atcom.
TimeOut Wrapup Time Max Len Auto full Auto pause Report Hold Time KeyPress Events Agent How many seconds an Agent's phone will ring before the Queue tries to ring the next Agent How many seconds after the completion of a call an Agent will have before the Queue can ring them with a new call. The default is 0, which is no delay How many calls can be queued at once. This count does not include calls that have been connected with Agents, it only includes calls that have not yet been connected.
1. 2. 3. 4. Menus allow for more efficient routing of calls from incoming callers. Also known as IVR (Interactive Voice Response) menus or Digital Receptionist. 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.
include = default exten = s,1,NoOp(Incoming DID) exten = s,2,Answer() exten = s,3,Background(record/GreetingNew) exten = s,4,Background(record/MakeYourSelection) exten = s,5,Set(TIMEOUT(absolute)=8) exten = s,6,Background(fpm-sunshine) exten = s,7,Set(TIMEOUT(absolute)=60) exten = s,8,Voicemail(6002,u) exten = 1,1,Goto(voicemenu-custom-2|s|1) exten = 2,1,Voicemail(6002,u) exten = 5,1,Goto(voicemenu-custom-3|s|1) 4.
Name Description Type Trunk Choice the trunk for the incoming rule Time Interval Choice the time interval for the incoming rule {analog, server provider, voip} Choice Pattern Pattern of the incoming rule Destination Incoming to destination Dialplan matched {users, voice mail, ring group…} default Non timeinterval matched S IVR 1. A trunk support a number of this time intervals, to support a number of Destination 2. Pattern: All patterns are prefixed by the "_" character.
Max greeting (in seconds) Dial '0' for Operator Maximum messages per folder Max message time Set the maximum number of seconds for a User's voicemail greeting Enable Callers to exit the voicemail application and connect to an operator extension. The operator extension must be defined from the 'Options' panel This select box sets the maximum number of messages that a user may have in any of their folders This select box sets the maximum duration of a voicemail message in seconds.
Voicemail Emails Subject Template Variables: \t : 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 and time the message was left ourcompan y.
Pin Code Admin PinCode Play music for the first caller set an optional pin code, Ex: "1234" that must be entered in order to access the Conference Bridge Defining this option sets a PIN for Conference Administrators Checking this option causes Asterisk to play Hold Music to the first user in a conference, until another user has joined the same conference Close the conference bridge when the last marked user logs out of the conference call Str* Str* Check box unCheck Check box unCheck Checking this opt
'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.
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’.
Name Dial Options Description Type Check box default Uncheck (T-Option) Allow the calling party to transfer the called party by sending the DTMF sequence defined on the Feature Codes page. (h-Option) Allow the called party to hang up by sending the DTMF sequence defined on the Feature Codes page. Check box Uncheck Check box Uncheck (H-Option) Allow the calling party to hang up by sending the DTMF sequence defined on the Feature Codes page.
Version Details: asterisk/GUI/Firmware version for PBX Server Date & TimeZone: time now for PBX Hostname:name for PBX b) c) Network: network message for PBX Eth0:9----- fill back IP for PBX(vlan IP) 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 d) Memory Usage Total：Memory quantity Used： Used of Memory Free： Free of Memory Shared：Shared of Memory Buffers
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.
Chapter 5 an Application Case of IP4G Figure: Network Topology In the network topology above: user 6020,6001,6002,6008 will be registered to IP4G, After configuration, it will realize the following function: 1) The internal user 6002 and user 6001 can call each other directly. 2) 6001, 6002, 6008 can communicate with outside through IP4G by GSM. 3) User 6001 and 6030 can call each other through VoIP trunk, although they are registered to different IP PBX.
5.1 How to Make Internal Calls through IP4G 5.1.1 Access to the Web Page of IP4G by Browser After connecting IP4G to LAN, please open your browser of PC with windows OS and input the IP Address of IP4G (the default IP address is 192.168.1.100), then users can get the following screen: Please input the default Username: admin; Password: atcom in the presented screen above. When users login successfully, they can get the configuration web page as below: 5.1.
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 IP4G, please click on Users Create New User, the writer configure user 6001 like the following: At last, please click on Update button, and click on Apply Changes button in up right corner of the main page. 5.1.
After configuring, please click on the APPLY button. Users can see the “Register status” is Registered, if user 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 IP4G. Users can register the 6002, 6008 like this. Now users can call each other directly between user 6001, 6002 and 6008. 5.
Users should hook on the Channels their 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. Users can configure the other channels like this. Then users must restart the IP4G. 5.2.2 Create an Outgoing Calling Rule After logging into the web page of IP4G, please click Outgoing Calling Rules New Calling Rule, the writer configure an outgoing calling rule like the following: www.atcom.
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 IP4G, please click on Dial PlansEdit DialPlan1, then selected the name of the outgoing calling rules like the following: At last, please click on Save button, and click on Apply Changes button in up right corner of the main page. 5.2.
5.2.5 Create Incoming Calling Rules In order to get an incoming call from outside with IP4G, users need set Incoming Calling Rules. Of course the precondition is that users have set up a trunk, a destination which include Voice Menu、Voice mail、a User Extension etc and a Time Interval.
At last, please click on Update button, and click on Apply Changes button 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.
is. After configuring, please click on Add button, and click on Apply Changes button in up right corner of the main page. Attention : the option of Fromuser : default is null. 3) Create an outgoing calling rule in IP4G, after logging into the webpage of IP4G, please click on Outgoing Calling RulesNew Calling Rule, the writer configures an outgoing call rule like the following: After configuring, please click on Save button, and click on Apply Changes button in up right corner of the main page.
2) Create a SIP trunk in IP04 named out . 3) Configure the router. 4) Create an outgoing calling rule in IP04 named toIP4G. Here I use Pattern: _4. . 5) Hook on the “toIP4G” option in DialPlan. After configuring, please click on Save button, and click on Apply Changes button in up right corner of the main page. Now users can call from 6030 to 6001 by dialing with prefix 4. Users can communication between IP04 and IP4G. 5.4 How to Call Each Other Directly from Different Network Segment.
At last, please click on Update button, and click on Apply Changes button in up right corner of the main page. 3) Set up AT-610 and register an IAX2 user 6020 Please select the VOIP option, then select the IAX2 option, the writer registers the IAX2 user 6020 as the following illustration: www.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 user must register IAX2 user instead of SIP user, because the user 6020 is not in the same network segment as IP4G.
5.6 How to realize the IVR IVR is Interactive Voice Response. Voice Menus allow for more efficient routing of calls from incoming callers. Also known as IVR menus or Digital Receptionist. 5.6.1 Upload Voice Menu Prompts If users want to configure the IVR which they need, they must upload their voice prompt.
5.6.2 Create Voice Menu Users can configure IVR like this: click on Voice Menus Create Voice Menus, then users can configure the IVR like the following pictures. First, selected the option “Answer” on the “Add new step” then click the Add new step . Second, selected the option “Background” on the “Add new step” then click the Add new step. Users can see the screen display like the following screenshots, then select their own voice prompt. Here the writer use the voice prompt named 04.
Third, hook on the option : Allow KeyPress Envents, then users can configure the operation from “0” to “*”, which their need. Please click on save button, and click on Apply Changes button in up right corner of the main page. Here the writer configures that press “0” then call “6001”, press “1” then call “6002”, press “2” then call “6008”. Of course 6001, 6002, 6008 have registered. www.atcom.
5.6.3 Add Incoming Calling Rules After configure the Voice Menu, users must configure the Incoming Calling Rules. Click Incoming Calling Rules New Incoming Calling Rules, users can configure it like this: Then when others call you through the analog1, they can here the IVR and do the operation which they need. www.atcom.
5.7 Conference In order to realize the conference option, the users which will attend to the conference must have registered. Here the writer uses 6001, 6002, 6008. Now please click Conferencing New conference Bridge, users can see the screen like the following screenshots: Then please click on Update button, and click on Apply Changes button in up right corner of the main page. Here the writer configures it like the screenshots above. Then 6001 dial 6300, and input Pin Code.
5.9 Agents You need complete the following two steps when you need the function of Agents . 5.9.1 Create Users as Agents Users can create phone 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.
Like this I have also created 6002, 6008.Then you must click System Status, then you can see the following screenshots: Click the button of “Login” so that all the Agents have logined. Then refresh the web, users can see the page that all the agents have logined like the following screenshots: 5.9.2 Create a Call Queue Please click Call Queues Create New Queue, then users can configure the options like this screenshots: www.atcom.
Then 6008(have registered) can call 6500, then 6001, 6002 are all ringing together. Of course you can refer to 4.9 in detail. 5.10 Follow Me Here 6001, 6002, 6008 have already registered and they can communicate with each other.
click on Add button, then please click on Update button, and click on Apply Changes button in up right corner of the main page. Now when 6008 dial 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 IP4G.
please pay attention to the red ellipse frame, all the users must in the same group. Here the writer haves created 6001, 6002 which are both in the group 1. 6008 can be in group1, 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 the writer completes the Group Call pickup function. www.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.
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.
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.
Reference http://atcom.cn/download.html http://www.asteriskguru.com/ http://www.openippbx.org/index.php?title=Main_Page http://www.atcom.cn/ www.atcom.
