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iCallDroid User Manual

1. The following figure shows an example of SIP extension 6010 settings Create New User General Extension 6010 callerlD Name 6010 D DialPlan DialPlant se D Internal CallerlD 6010 CO Calleri Number 6010 D Enable voicemail far this User D Vniceldail Access PIM code D Email Address Technology air Oe CO Analog Station D fash D nilash D Codec Preference First ulaw Second osm w Third None Fourth None w FifthiNone w VOIP Settings MAC Address D Line Number D Linekeys D SIPJAX Password D lax Require Call Token D LA Max Call Numbers D NAT D Can Reinvite DTMF Mode RFc2833 D insecure no D Other Options 3 Way Calling analogi D lin Directory D Call Waiting analogi D C ADA User D is Agent D Pickup Group Cancel Update 3 8 2 Create Analog Users Creating an analog user is similar with SIP user Please fill in Extension CallerID Name DialPlan and CallerID Number in general item and choose port 2 for Analog Station in technology component Also there are other optional settings for WWW Openvox cn you please configure them based on your fact After setting please remember to update and apply your changes Create New User General Extension 6011 CallerID Name 6011 D DialPlan niara v DD Internal CallerID 6011 D CallerID Number 6011 d Enable Yoicemail for this User VoiceMail Access PIN code D
2. 0 0 SIP User General ERES EEE 22 Configure Hardware 6001 6001 M 0 0 SIP User 6002 6002 0 0 6003 6003 0 0 6004 6004 0 0 SIP User 0 0 0 0 0 DRE SIP User Hostname aa Trunks SIP Zak ser ome openvox cn 88 Outgoing Calling Rules OS Version Linux f2home openvox cn 2 6 28 10 2 Tue May 22 13 50 41 CST 2012 armv6l unknown aa Dial Plans 6005 6005 SIP User 6006 6006 SIP User SIP User aa Ring Groups 6007 6007 6008 6008 Bum 6009 6009 essages 0 0 SIP User sterisk 1 6 2 11 6088 6088 Messages 0 0 Analog User Port 1 Asterisk GUI version SVN r5209 No Extension assigned Check Voicenails VoiceMailMain SIP User Asterisk Build On Hold aa Voice Menus 88 Time Intervals Server Date amp Tinezone Mon Jul 9 11 12 33 CST 2012 88 Incoming Calling Rules xNo Extension assigned Dial by Names Directory aa Yoicemail Uptime 11 12 29 up 1 14 Load Average 4 22 2 73 2 29 aa Conferencing aa Options aa Asterisk Logs aa Call Detail Records As you can see there are ten default SIP extensions and one analog and if connect your telephone with iCallDroid the status of analog extension 6088 is free which means you can make calls After registered a SIP user successfully by SIP software such as 3CXPhone on your PC the SIP status will change to free and then you are able to make inbound and outbound calls 2 2 Telnet access by putty
3. OpenVox iCallDroid User Manual Copyright Copyright 2012 OpenVox Inc All rights reserved No part of this document may be reproduced without prior written permission Confidentiality Information contained herein is of a highly sensitive nature and is confidential and proprietary to OpenVox Inc No part may be distributed reproduced or disclosed orally or in written form to any party other than the direct recipients without the express written consent of OpenVox Inc Disclaimer OpenVox Inc reserves the right to modify the design characteristics and products at any time without notification or obligation and shall not be held liable for any error or damage of any kind resulting from the use of this document OpenVox has made every effort to ensure that the information contained in this document is accurate and complete however the contents of this document are subject to revision without notice Please contact OpenVox to ensure you have the latest version of this document Trademarks All other trademarks mentioned in this document are the property of their respective owners WWW Openvox cn Contents Chapter By Ci VW coca dian rr EHI Ur E EE 4 WSI 4 1 2 Physical Comme ction seen 5 Chapter 2 Access iCallDroid cios iso Pane Eye Hera EU FEXR EAE eH s PaENEAERU FK S ER Sa EU sa eS ERE eH sra EROR UE 6 2L Web Page Access Dy ISTO W Sel aeuo 55 902 0921559854536085262 08055 80 260 9445428 0868510890 42240509050 082424
4. cancel M Save You should input a name for the DialPlan Name and select outgoing call rule and local context that you want to use 3 8 Users Users is a shortcut for quickly adding and removing all the necessary configuration components for any new phone In default there are 10 SIP and 1 analog extensions SIP extensions are from 6000 to 6009 whose password are all 8088 analog extension is 6088 Please attention that all extension number is limited and should be between 6000 and 6299 in factory settings Create New User Modify Selected Users List of User Extensions Where to Buy Extension DialPlan OutBound CID DialPlan1 Edit XDelete DialPlan1 Edit X Delete DialPlant Edi X Delete DialPlan1 Edit X De DialPlan1 Edit X Delete DialPlan1 Edit X Delete DialPlan1 Edit Delete DialPlan1 Edit X Delete DialPlan1 Edit X Delete DialPlan1 Edit Delete DialPlan1 Edit Delete E Ei Ei F www openvox cn 1 D If you want to creat other users please click Create New User button 3 8 1 Create SIP Users When create a SIP user you should fill in extension CallerID Name DialPlan in General component and choose SIP in Technology To get more information about other parameters please put your cursor on D label and confi gure them Create an AX user is similar to SIP and just remember to choose JAX instead of SIP
5. 1 Please run your putty software and input the iCallDroid IP address like the following figure WWW Openvox cn fae PuTTY Configuration Category 5 Session Basic options for your PuTTY session Logging Specify the destination you want to connect to Terminal ner er Host Name or IP address Port Bell 192 168 1 245 23 i Features Connection type Window Raw Telnet Rlogin SSH Appearance Behaviour Translation Saved Sessions Load save or delete a stored session Selection Colours Default Settings Load Connection Data Proxy Telnet Delete Rlogin SSH Serial Save Close window on exit Always O Never Only on clean exit Aot _Open j Cave 2 Login by your putty WWW Openvox cn Chapter 3 Configure iCallDroid By Web GUI 3 1 System Status In the system status screen it shows the functions you configured such as trunks extensions system info and so on like that Apply Changes so System Status System Status Please click on a panel to manage related features e Trunks 4e Conference Rooms PorHostnameslP openvox Analog Ports 2 k Parking Lot l 1 de Extensions t1 MER Analog Free 9 Ringing Busy Unavailable No Parked Calls CL tension Name Label s 0 0 6000 6000 Messages SIP User System Info 8001 8001 Messages 010 SIP User General CELLES 6002 6
6. The following are example settings Feature Codes amp Call Parking Preferences Feature Options Feature Digit Timeout 2000 irillisecands call Parking Extension to Dial ta Park a Call 700 Extensions for Parked Calls 701 720 Ex 701 720 Parked Call Timeout in secsi 45 Feature Map Blind Transfer ge default is 3F Disconnect default is Attended Transfer 2 Call Parking 1 Feature Digital Timeout when timeout the time you set between two feature digital itis unable to go on making this call and should dialed again Call Parking When you are busy or inconvenient to answer calls you can hold on calls for a period of time After finishing works you go to answer the call 3 18 VoiceMail Groups Define VoiceMail Groups to leave a voicemail message for a group of users by dialing extension number The following figure realizes that dialing 6600 to leave messages for user 6000 6001 6002 Hew Voice Mail Group YoiceMail Group s Extension 6600 Label greetings User Mailboxes Ml ooo M oot M oo2 15003 amp Cancel Save WWW Oopenvox cn 2 5 3 19 Voice Menu Prompts This component is used for recording custom voice menu There is a default Voice Menu prompts named chinese ivr wav Now please follow me to record a new voice menu prompt First please select Voice Menu Prompts option from the vertical menu on the left of the main page then click Record a ne
7. e ell m c m kek The follow me extension 6001 in the above figure means when someone dials 6001 if 6001 responses anything the call will forward to extension 6002 and 1f 6002 also WWW Oopenvox cn doesn t answer the call it will forward to 6088 Finally if all extensions in the Follow Me do not answer the call it will hang up Let s take 6003 for an example to illustrate how to create a follow me 1 Click Edit button after 6003 1n the above screen 2 Make Status enable select a type for Music On Hold Class and choose a dialplan you have set before Status D Enable Disable Music On Hald Class D DialPlan Cb Destinations D Add FollowMe Humber amp Cancel F Save 3 Please click Add FollowMe Number button select new follow me numbers from the drop down list then click t Aa button to add it add all numbers you need by the aforementioned way Finally do not forget M save and _Apply Changes Status D Enable Q Disable Music On Hald Class D i DialPlan D Destinations D 6000 30 seconds New FollowMe Number D Dial Lacal Extension D Dial Outside Number 6001 6001 for SO Seconds Dial Grder D Ring after Tring previous extension number Ring along with previous extension number WWW Openvox cn 2 3 17 Call Features By call features you are able to set some functions such as Call Parking Feature Map
8. interval Dy day of week Men M to Fi v CO By Days of a Month Date Month Time Entire Day Start Time 09 00 AM End Time 06 00 FM amp Cancel iv Update Settings in the above picture mean that incoming calls from 09 00 AM to 06 00 PM of every Monday to Friday work normally and calls not in this time segment will not Work WWW Openvox cn 2 3 13 Incoming Calling Rules In factory settings all incoming calls route to FXS port You can create modify prioritize and delete incoming call rules by setting this option Hew Incoming Rule Trunk Time Interval Pattern D ds Cancel iv Update Trunk Select a trunk created before from the drop down list for incoming call use Time Interval Determine the time when the incoming call rule works Destination Set a destination extension or group to response the incoming calls Finally please click on Update and Apply Changes button at the up right corner of the main page to make settings effective 3 14 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 Extension for checking messages When you dial the number here I set it as 6600 you will hear the message other people left for you Max greeting Set the maximum number of seconds for voicemail greetings Maximum message per folder This select box sets
9. the maximum number of messages that a user may have in any of their folders Max message time This select box sets the maximum duration of a voicemail message in seconds Message recording will not occur for times greater than this amount Min message time This select box sets the minimum duration of a voicemail message in seconds Messages below this threshold will be automatically deleted WWW Openvox cn 2 1 Extension for checking messages D BOO Direct Voicemail Dial D Max greeting in seconds D Dial O for Operator D Kessage Options Maximum messages per folder ao D Max message time D 1 minute Min message time 1 second Playback Options sav message Caller ID D say message duration D Flay envelope Allow users to review D G Cancel Iv Save About other options please put your cursor on the D label to get detail information Example in the above figure means that when you dial 6600 you will hear the message anyone else left for you but the message duration should less than 20 seconds 3 15 Conferencing The conferencing function of Asterisk 1s 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 Please select conferencing option and then create a new conferencing bridg
10. three basic parameters you should configure Calling Rule Name It is unique to identify the outgoing calling rule when listed in dial plans Here I name it outgoing Pattern It acts as a filter for numbers pass through it means any number you dial out with prefix 9 will use this outgoing call rule In default settings outgoing calling rule pattern is _9X which means you make outbound calls with 9 as prefix Use Trunk Assign a trunk to carry traffic for outgoing calling rules Please put your cursor on the D label to get information about other parameters Finally do not forget 898 and www openvox cn 1 3 7 Dial Plans A Dial Plan is a collection of Outgoing Call Rules that can be assigned to one or more users Please click Dialplans then you will get the following figure As you can see from the figure there is a default dialplan named DialPlan 1 you can edit or delete it Calling Rules Options openvox default parkedcalls conferences ringgroups voicemenus queues voicemailgroups director ra p s s P s s ggroups s q s groups Ys Bait X Delete pagegroups page an extension Click New DialPlan button to add a new dialplan Create New DialPlan DialPlan Name DialPlan2 Include Outgoing Calling Rules openvox Include Local Contexts v default V parkedcalls V conferences Mringgroups Mvoicemenus v queues Mivoicemailgroups v directory Vl pagegroups V page an extension
11. 002 i ages 0 0 SIP User SIP User Hostname SIP User f2home openvox cn SIP User A OS Version SIP User i Linux fZhome openvox cn 2 6 28 10 4 Fri Jun 29 17 43 27 CST SIP User 2012 armv6l unknown SIP User SIP User Asterisk Build Analog User Port 1 Asterisk 1 6 2 11 business Ring Group Asterisk GUI version SVN r5209 No Extension assigned Check Voicemails VoiceMailMain Server Date amp Tim z ne No Extension assigned Dial by Names Directory Wed Jul 11 11 57 32 CST 2012 Uptime 11 57 32 up Z min Load Average 1 99 1 04 0 41 3 2 Configure Wan Port In the web GUI you are able to configure WAN port by your needs There is an example in the following figure for WAN port settings WAM Settings Original MAC WAN Ethernet MACUU Manual Setting WAN Part IP Assignment D Static IP DHCP MTUCD 1500 IP Address D 182 158 1 200 Subnet Mask D 255 255 255 0 Default Gateway D 172 168 0 1 Set DNS Server b Manually Automatically Primary OMS Serer Secondary DNS Semer kancel Changes FI Update Settings WWW openvox cn 3 3 DDNS DDNS is for your dynamic Domain Name Service configuration If necessary please set DDNS like the following figure Finally do not forget to update and apply your changes DONS Settings DONS Server Type WNL O Tay COM or DONS Username Qpenvox Voip DONS Password apenvax atico Hostname to register openvox vo
12. 1 200 Tunnel password FER Part Eoso Use Outbound Proxy server Required by some VoIP Providers Specify IP or name EE Perform provisioning From Following URL ito Advanced settings Cancel Sep 3 Register another SIP user 6001 by the same way before Then the two SIP users can call each other There are ten default SIP extensions 6000 6009 so you just need to register one or more SIP users of these ten While if want to use other SIP number you have to create SIP users and please refer to Create WWW Openvox cn 3 2 SIP Users for creating method 4 1 2 SIP to Analog Phone Step 1 Register a SIP user like aforementioned way Step 2 Connect your analog telephone with FXS port Then the two users can call each other In default settings FXS port has been set as analog extension 6088 If you want this port working please just plug your telephone on iCallDroid s FXS port Also you can change this analog extension number please refer to Create Analog Users for creating information 4 2 Make External Calls Step 1 Register one or more SIP extensions please refer to SIP to SIP Calls for how to create a SIP user Step 2 Plug your telephone to FXS port Step 3 Plug PSTN line to FXO port Step 4 Create an outgoing calling rule about it please refer to Outgoing Calling Rules Step 5 Create dial plans about it please refer to Dial Plans for how to create dialplans Then you can make internal and exter
13. 5588 6 22 ee 7 Chapter 3 Configure iCallDroid By Web GUI ee eere eene eA 9 SMS I ESI RO mcm 9 3 2 Gres M wan oh 9 DON S TT T P 10 CONS EAN I NITET 10 SNB CP TT UU 11 33A Create Analog Trans 11 E S AR TE M TUKS orrena T 12 3 6 Qutcoing Callina RTS a 13 A D ES A VIG RR E GO 15 SM BI NR eem 15 361 Create SIP RR UU 16 702 Create Analog Ne 16 iu 17 310 Music On Hold MERE UNT 18 3 11 Voice Uu 19 SN Pe sw 20 3 15 Incoming Calling Rule8 ae 21 JOIN 21 3 19 GI OMS RET mem 22 ED 23 NEG IUNII RTT 25 16 Voice NE GTOUPS ee Se 25 3 19 Voice Menu romnhts a Eai 26 Un lw 20 LAN Ce xi E C riders 21 PUA A 28 SUUS ISI NET T E E E ein 28 Wee Canl Doiii l RECO NN T 30 S PS NAT E RO TT 30 Chapter 4 Typical Application Case wisssissscssssesstsesasencassacesnnnntenwansssesntecsvacesessoseaunners 32 MAR MAO ARS ae a a des M ee 32 WWW openvox cn Noi ec Em 32 2 1 7 OU TO AIO PONG oxide diim Oed EID M MO Or 33 RE 33 Appendix A Default settings and Specifications eee e eere eene 34 Appendix B Typical Application e eee eee ee eee ee e eee eee eee esee eese ee eee eee 36 WWW Openvox cn Chapter 1 Overview 1 1 What is iCallDroid OpenVox iCallDroid is a small smart and ope
14. Email Address Technology SIP D TAX D Analog Station Port 2 D flash D TSO rxflash D 1250 Codec Preference First wlaw Second GSM Third Hone Fourth None Hone ww VoIP Settings MAC Address Line Number n D Linekeve t vl D SIP IAX Password D TAR Require Call Token D ite Nex call Nenbese U0 NAT D Can Reinvite D DIMF Mode D insecure Other Options 3 Way Calling analog D In Directory D Call Waiting analog D ADA User D Cl Is Agent D Pickup Group ii v G Cancel F Update 3 9 Ring Groups Define RingGroups to dial more than one extension simultaneously or to ring more than one phone sequentially This feature may also be called Huntgroups WWW Openvox cn 1 n Hew EingGroup Ringsroup Mame business Extension for this ring group 6400 Ring Group Kenbers Available Users 6000 SIP 6000 a BO02 SIP 6002 6001 SIP 6001 6003 SIP 6003 nalogPort 2 ra B n4 SIF 6004 ES e005 SIF 6005 E006 SIF 6006 BOOT SIP BOUT 6008 SIP 6008 6003 SIP 6009 MEE Ring Group Options Strategy seconds to ring each member of If not answered Goto Ignore redirections G Cancel b Save Please fill in RingGroup Name Extension for this ring group and select from Available Users to Ring Group Members Decide what kin
15. d of strategy for this ring group Ring all simultaneously When someone calls the ring group all members of the ring group will ring at the same time Ring in order When someone calls the ring group the member will ring in order If not answered Goto Choose a destination from the drop down list if no one in the ring group answers the call If you want your ringgroup to work you should set your destination is ring group in incoming calling rules After setting please Misave and LAPPY Changes 3 10 Music On Hold Music On Hold lets you customize audio tracks for different queues parked calls etc As you see it 1s able to upload files what you want WWW Openvox cn 1 8 Manage Music on Hold Classes New MOH class Bi X Delete Manage Music On Hold Classes manage MOH class default Upload an 8 KHz Mono Music file Choose file to Upload List of Sound Files manolo camp morning coffee wav X Delete 3 11 Voice Menus Menus allow for more efficient routing of calls from incoming callers It also is known as IVR Interactive Voice Response menus or Digital Receptionist Create Hew VYoiceNenu Name volcemenn Advanced Edit Extension TOOO D F D Allow Dialing Other Extensions Actions Answerthe call Play demo instruct amp Donat Listen far KeyPress events Hangup call Add new Step Allow KevPress Events 0 Goto RingGroup business 1 foto Ope
16. der to sort by that column Click on row to display full record Start time Duration Source Destination Caller ID Disposition LO ert TBI 5221 ANSWERED 2 2012 07 10 03 54 34 NO ANSWER 3 2012 07 10 03 48 25 NO ANSWER 3 23 2 Active Channels It displays current active channels on the PBX with the options to hangup or transfer WWW Openvox cn 30 Refresh Now Refresh Now Active Channels 3 Refreshing Active Channels in 10 Seconds Seconds Application Channel DAHDII 1 Up undefined Transfer SIP 6000 00000001 undefined 59 Transfer 1564 System icommandp Transfer Localfexecutecommand asterisk_guitools 69a5 2 undefined There are other advanced options such as SIP Settings IAX settings Asterisk CLI www openvox cn Hangup Hangup Hangup 31 Chapter 4 Typical Application Cases 4 1 Make Internal Calls 4 1 1 SIP to SIP Calls Step 1 Run your SIP software to create two SIP users Step 2 Register a SIP user 6000 like that ccount settings X Account name soo Caller ID administrator Credentials Enter our SIP account credentials Extension sono ID soon Password eee My location Specify Ehe IP of your PBX SIP server I am in the office local IP fisz 158 1 245 af PBX f Lam out of the office external IP of FPEX Use 3CX Tunnel Eliminates Firewall configuration Requires 3Cx Phone System Far Windows Local IP of remote PBs 1568
17. e WWW Openvox cn 2 2 Hew Conference bridge Extension ASOO D Marked Admin user Extension Password Options Pin Code 1213 D Bone Pincede 1415 ai Conference Room Options D Play hold music for first C D Close conference when last marked user caller exits 0 M Enable caller menu nnounce callers 10 Quiet Mode Wait for marked user G Cancel Update The example in the above figure achieves that the conference number is 6300 common members type the pin code 1213 to enter conference and administrator types 1415 to enter the conference Enabling play hold music for first caller option and announce callers option so the first member who enter the conference will hear music and the online members will be informed when someone enter the conference At last please click on Update button and click on Apply Changes button in up right corner of the main page 3 16 Follow Me If A calls B B does not answer the call will be transferred to C who ls set up in follow me FollowMe Preferences for Users FollowMe Options Extension Follow Me Follow Order 6000 Enabled 6001 6002 6088 6001 Enabled 6002 6088 6002 Enabled 6088 6001 6003 Disabled Not Configured 6004 Disabled Not Configured 6005 Disabled Not Configured 6006 Disabled Not Configured 6007 Disabled Not Configured 6008 Disabled Not Configured 6009 Disabled Not Configured 6088 Disabled Not Configured m e
18. ip eicp net cancel Changes E Update Settings 3 4 Configure Hardware Analog Hardware FES Forts FEO Ports Tone Kegion D 3 United States North America software echo canceller QD iw mg F Advanced Settings permade D a law override D fxs honor mode boostringer D fastringer D lowpower D ring detect D MUI mode D Eee 1 og cidbuflen D 0 78 7080 70 7878777 line ub fixedt D NA Cancel Changes W Update Settings WWW Openvox cn 1 There are three items which are Analog Hardware Tone Region and Advanced Settings Analog hardware When you boot iCallDroid FXS and FXO ports will be detected automatically And also you are able to choose and update their signaling type when click edit button Kewl Start and Loop Start are available for each port Tone Region You should select your tone region and software echo canceller according to your situation if your country name is not in the dropdown list please ask your service operator which kind of tone region is used in your area Advanced Settings Please set every option by your fact For example if you would like a law override is able please tick off and select configuration like that 3 5 Trunks Trunks are outbound lines used to allow the system to make calls to the real world Trunks can be VoIP lines or traditional telephony lines Please select Trunks from the vertical list o
19. kup 2012ju110_ 133433 J Cancel Backup Enter a file name and click backup button once backup process 1s completed After backup the following screen will display WWW Openvox cn 2 7 Create New Backup r List of Previous Configuration Backups 1 backup 2012ju110 133311 Jul 10 2012 Download from Unit Restore Previous Config Delete 3 22 Update This function enables to update firmware installed on the appliance Please click EL to choose firmware file to upload Update Firmware Upload a new image Choose a KRC image file Upload Image 3 23 Options This component is composed of General Preferences Language Change Password Reboot and Advanced Options Please attention Extension preferences in General Preferences there are default ranges when you create any kind of extensions also you are able to disable these ranges WWW Openvox cn 2 8 General Preferences Language Change Password Reboot Advanced Options Global jut Bound CID 3 Global Cr Dour CID Man perator Extension D Ring Timeout Enable Idle Image Display Ur Phone Digi TEN D VoIP Phone Digit Timeout Extension preferences Disable Extension Ranges User Extensions 6000 Conference Extensions ASO VoiceMenu Extensions TOOU RingGroup Extensions 6400 Queue Extensions 6500 VoiceMail Group Extensions 6600 Rese
20. n source IP PBX designed for home use With the analog ports integrated the legacy telecom equipments would be able to connect to the modern unified communication world The smart phones and other mobile devices can be one part of it Suppoting SIP and G 729 protocal and codec expansibility is strongly enhanced The Asterisk GUI interface tremendously reduces the difficulty to enter the VoIP communications Easy to use and ready to work are features shown to you when you open the package Sample applications Application Scenes De 5 el I3 E WIFI Router Scene 1 The two parties are at home and they call each other through WIFI network and iCALLDroid Calls between them are free of charge Scene 2 If the user is a VOIP user he she can make calls through iCALLDroid appliance and VOIP network which can save much costs em f WS G Scene 3 If the user wants to make calls to an overseas business partner he she can take advantage of 3G Figure 2 Sample applications Features Access via telnet web gt Call conference Www openvox cn VW Now ON ON VV NV V WV Call Detail Record CDR Call forward Call waiting Call transfer Call queues Ring group Configurable IVR menu Music on Hold PSTN analog SIP IAX trunk Voice Mail Open Source Asterisk IP PBX Firmware upgradable via web page 10 available SIP IAX2 extensions Applications SOHO SMB telephon
21. n the left of the main page and then the following screen will be displayed There is a default analog trunk named openvox for 1CallDroid Apply Changes Hanage Analog trunks Analog Trunks VOIP Trunks Trunk Analog Ports openvox Edit lelete 3 5 1 Create Analog Trunks There is a default analog trunk if you wouldn t like it you can delete or edit it to set a new onc WWW Openvox cn 1 1 Edit Analog Trunk G D Channels v 2 roups Group 1 ropenvox Trunk Hame D openvox CallerlD Normally you should nat have to adjust your analog ports beyond the initial calibration Port 2 should you still need to fine tune your audio settings please use the adjustments at the right Advanced Options Busy Detection Busy Count D Busy Pattern D 500 500 Ring Timeout Answer on Mo w Hangup on Folarity Switch Polarity Switch Call Progress D No M Progress Zone D Use CallenD CD Caller ID Start CD CallerD D As Received M Pulse Dial CID Signalling CD mailbox Flash Timing D 750 Receive Flash Timing D Cancel F Update There are many parameters to set an analog trunk and here I just configure two parameters Channels Please tick off before 2 means channel 2 1s able for the analog trunk Trunk Name Please give a unique label to identify the trunk name and I name it openvox Advanced Options Parameters in advanced options are optional If yo
22. nal calls In default settings FXS port has been set as analog extension 6088 If you want this port working please just plug your telephone on iCallDroid s FXS port Also you can change this analog extension number please refer to Create Analog Users for creating information WWW Openvox cn 3 3 Appendix Default settings and Specifications Access mr Puy IP 192 168 1 254 8088 192 168 1 254 23 telnet Username admin root Password admin OpenVox Recommended Browser gt Firefox gt Chrome SIP Analog Extensions Limited 6000 6299 TenSIP Extensions 6000 6009 Password 8088 gt One Analog Extension 6088 Calling Rules gt Outgoing FXO Dial Out With Prefix 9 Incoming Goto FXS Port Spec Extensions 1 Analog Phone 10 300 SIP IAX2 Extensions CPU 700MHz ROM Flash 32MB 64MB Available for OEM RAM 128MB DDR2 SDRAM Up to 256MB Available for OEM Power DC 12V FXS FXO 1 FXS 1 FXO LED 4 OS Linux Y Y Y Y VW V V WV Kernel Version 2 6 28 10 www openvox cn 34 Vv Y V V WV Size 160 L 100 W 31 8 H mm Weight 236g Operation Temperature 0 70 C Operation Humidity 10 95 Power Dissipation lt 5W 1 FXS No USB Device Www openvox cn 35 Appendix B Typical Application p o gt ou Android Phone 6 iCallDroid WIFI Router Analog Phone SIP Phone e Recommended Software Androidsip client Bria gt iphone si
23. p client Bria is phonelite zoiper gt iphone iax client zoiper Www openvox cn 36 OpenVox OpenVox Communication Co Ltd Open Vox Best Cost Effective Asterisk Cards Address F 3 Building No 127 Jindi Industrial Zone Shazui Road Futian District Shenzhen Guangdong 518048 China Tel 86 755 82535461 82535095 82535362 Fax 86 755 83823074 Business Contact sales openvox com cn Technical Support support openvox com cn Business Hours 09 00 18 00 GMT 8 from Monday to Friday URL www openvox cn Thank You for Choosing OpenVox Products Www openvox cn 3 J
24. rator ato User 6000 Hone User Extension 6000 User Extension 6001 User Extension B00 User Extension 6003 User Extension 6004 User Extension 6005S User Extension 6006 User Extension BOOT User Extension BODB User Extension 6009 User Extension 60868 Ring Group business Operator Hangup Congestion Name A name for the voice menu Extension Optional if you want this voice menu to be accessible by dialing an WWW Openvox cn 1 O extension and then enter that extension number Actions Show the actions you select from Add new Step after you choose an action from Add new Step drop list options please click T Add new Step button then it will show at Actions frame KeyPress Including digital 0 to 9 and other characters When you put your cursor after the digital there will be an orange frame and you can click at here to choose the caller 1n the drop down list and then update it After setting please save and apply your changes The above example KeyPress Events settings mean the call will goto ringgroup you have configured before if press 0 goto operator if press 1 and goto the extension 6000 if press 2 3 12 Time Intervals Time Intervals are defined ranges of time that will be used by call routing features Mew Time Interval i When you click button the following figure will display Hew Time Interval Time Interval Mame
25. t to defaults CJ Cancel v Save Advanced Options General Preferences Language Change Password Reboot Advanced Options Advanced Options Clicking the Show Advanced Options button below provides the additional menu items on the left hand sidebar Hotice Digium does not provide support for the options configurable in the Advanced menu items Digium does nat provide support for bugs Uncovered in the Advanced menu items If voaur unit becomes inoperable due to editing of the Advanced menu items Digium Technical Support will request that you reset your unit toFactory Default configuration Continue at your own risk show Advanced Options After clicking Show Advanced Options button there will be advanced options in the vertical menu on the left of the main page like the following Www openvox cn 2 9 aa Options Admin settings 22 SIP Settings 3 23 1 Call Detail Records This component provides the record of all incoming and outgoing calls including the channels used and duration of calls After click on options Advanced Options Show Advanced Options please select the Call Detail Records option from the vertical menu on the left then you will get the following screen Call Detail Report Inbound calis D Vi Outbound calls D Internal calis D V External callsD _ Show all fields Show system callsW 3 Total records Viewing 1 3 of 3 Selected Next Click on column hea
26. u don t know what they mean you can put your cursor on the D label to get detailed information about the parameter Please set these parameters according to your requirements and service provider 3 5 2 VOIP Trunks A VoIP service provider VSP that you have signed up with is also a trunk Via the VoIP trunk you can make calls by VoIP service to save much cost or even free of charge for internal calls WWW Openvox cn 1 2 Create Hew SIP IAX trunk x Type SIP Context Naming D Bazed on Provider Hame Provider Name D siptrunk Hostname Of 192 155 1 158 Username D ROLE Password Anis G Cancel Save Type You can select SIP or IAX type to meet your need Context Naming This parameter means how Asterisk GUI should determine the context name in Asterisk s conf files Provider Name It is a unique label to help you identify the trunk when listed in outgoing calling rules and incoming Hostname It is the IP address or domain name of your service provider s server Username It is your service provider configured Password Password is your service provider configured for the user Configuration in the above figure is an example of SIP trunk named 8018 whose password is 8018 and hostname is 192 168 1 168 3 6 Outgoing Calling Rules Outgoing calling rules 1s used to identify an outgoing call route when make external calls which trunk and what dial pattern calls use So an o
27. utgoing calling rule pairs an extension pattern with a trunk used to dial the pattern In default settings all outgoing calls pass through FXO port that is openvox trunk to the external PSTN network Manage Calling Rules New Calling Rule Restore Default Calling Rules An outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern This allows different patterns to be dialed through different trunks e g local 7 digit dials through an FXO but long distance 10 digit dials through a low cost SIP trunk You can optionally set a failover trunk to use when the primary trunk fails Note that this panel manages only individual outgoing call rules See the Dial Plans section to associate multiple outgoing calling rules to be used for User outbound dialing WWW Openvox cn 1 3 Please click Outgoing Calling Rules from the vertical menu on the left of the main page and then click New Calling Rules to define a new outgoing calling rule New CallingRule Calling Rule Name outgoing Pattern CD E Caller IDD C Send to Local Destination D Destination Ex MacrofsomeMacro S EXTEN 1 send this call through trunk Use Trunk D strip D digits from front and Prepend these digits before dialing using this filter O Use FailOver Trunk fail over Trunk Strip digits from front and Prepend these digits before dialing using this filter Cancel Save There are
28. w Voice Menu prompt button you can get the a screen to set Record a new Voice Kenu prompt File Mame OpenVox Format GSM v 6000 dial this User Extension to record a new voice prompt CJ Cancel Record Here I set OpenVox as the file name and dial extension 6000 to record the format gsm new voice prompt Once click Record button your software SIP will show like that Incoming calls You answer the call and speak to the microphone to record After you finish the record please hang up the call You can refresh your webpage to see that there 1s a sound like 3 20 System Info Click System Info to get general network disk usage and memory usage WWW openvox cn 2 6 information System Information General Hetwork Disk Usage Memory Usage OS Yersion Linux f home copenvox cn z 6 28 10 4 Fri Jun 29 17 43 27 CST 2012 armwv l unknown Uptime Le smi Sa ma tes el Load Average 2 85 2 45 2 29 Asterisk Build heterisk 1 6 2 11 Asterisk GUI version SVHW r5209 Server Date amp TimeZone Tue Jul 10 12 01 54 CST 2012 Hostname fzhome opermn ecox cn 3 21 Backup Backup and Restore are two of the mandatory functions of any application and iCallDroid is not an exception You can backup all the files under the etc asterisk directory and restore them Click 5 oreste Wew Backup then you will get the following screen Create Hew Backup File Name bac
29. y system Hosted service FAX terminal IVR system 1 2 Physical Connection DC 12V gt 12V Power Supply Adapter WAN Port Network Switch FXO Port gt PSTN Analog Line FXS Port gt Analog Telephone And please attention that LAN port is unavailable Www openvox cn Chapter 2 Access iCallDroid There are two ways to access 1CallDroid and a PC is needed to access it 1 Web page access by browser Firefox and Google chrome are recommended 2 Telnet access 192 168 1 254 23 by putty 2 1 Web Page Access by Browser Default IP address 192 168 1 254 8088 Username admin Password admin It is very convenient to access by inputting the IP address in your web browser and because of compatible issues I recommend Firefox and Google chrome Before access please make sure that your PC is in the same network segment with 1CallDroid For example you set 192 168 1 253 as your PC s IP and 255 255 255 0 as subnet mask Asterisk Configuration Engine Username admin Password esse Login After login successfully you can get the configuration web page as bellow WWW Openvox cn Apply Changes Logout ag System Status System Status Please click on a panel to manage related features JdeTrunks 1 Je Conference Rooms L 1 Port Hostnane TP e Parking Lot openvox Analog Ports 2 de Extensions No Parked Calls ai ETE es Free O Ringing Busy 9 Univailable 88 DDNS 6000 6000 Messages

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