A2410E - Telephony OpenVox - Free user manual and instructions

USER MANUAL A2410E OpenVox

Close-up of a green printed circuit board with multiple electronic components and connectors (no visible text or symbols)

A2410E

Date: 19/07/2011

Version: 1.2

OpenVox

深圳开源通信有限公司

OpenVox-Best Cost Effective Asterisk Cards

OpenVox Communication Co.Ltd.

Address: F/3, Block No.127, Jindi Industrial Zone,

Shazui Road, Futian district, Shenzhen, Guangdong 518048, China

Tel:[phone], Fax:[phone]

E-Mail: [e-mail] [e-mail]

M for Technical Support: [e-mail]

Business Hours: 9:00AM-18:00PM from Monday to Friday

URL: www.openvox.cn

Thank You for Choosing OpenVox Products!

Content

1. Overview....4

1.1 What is A2410E/AE2410E .... 4
1.2 What is asterisk....4

2. Hardware setup....5

3. Software installation and configuration ....6

3.1 Download....6
3.2 Installation 6
3.3 Configuration 7
3.4 Call test 11

4. Reference 16

Test environments

CentOS-5.6

Kernel version: 2.6.18-238.12.1.el5

DAHDI: dahdi-linux-complete-2.4.0+2.4.0

Asterisk: 1.8.4.4

Elastix 2.0.4

Hardware: OpenVox A2410E/AE2410E

1. Overview

1.1 What is A2410E/AE2410E

A2410E is an independent research and development modular analog telephony interface product by OpenVox Communication Co. LTD. AE2410E is A2410E with an EC module, they are designed to build SMB PBX. A2410E/AE2410E must be made up with FXO-400 and FXS-400 together to build a workable system.

1.2 What is asterisk

The Definition of Asterisk is described as follows:

Asterisk is a complete PBX in software. It runs on Linux, BSD, Windows (emulated) and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in four protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, IAX, SIP, H323 (as both client and gateway), MGCP (call manager only) and SCCP/Skinny (voip-info.org).

graph TD
    A["Telecom"] --> B["Asterisk VOIP PBX"]
    C["PC+SoftPhone"] --> B
    D["SIP Phone"] --> B
    E["Analog Phone"] --> B
    F["Analog Card"] --> G["BRI Card"]
    F --> H["PRI Card"]
    F --> I["GSM Card"]
    F --> J["V100 Card-"]
    K["Operating System: Linux and FreeBSD\nOpen Source Drivers Asterisk®; Zapid, Bitfall®, mSDN and ISO/ISO\nProcode: SIP JAX, SS7, MDCP1E02, P2 and more\nApplications: VR, CRM, FastCode, CallCenter, Billing, and your application."] --> B
    B --> L["Switch"]
    L --> M["SIP Phone"]

Figure 1 Topology

2. Hardware setup

The following matters need your attention before using A2410E, please check that:

1. Power supply: Plug 12V power line into the connector according to figure showed.

Module 1 Module 2 Module 3 Module 4 Module 5 Module 6 312.0mm 106.8mm Channel 1 to 4 Channel 5 to 8 Channel 9 to 12 Channel 13 to 16 Channel 17 to 20 Channel 21 to 24 SMEC V1.1 Power Supply (12V) OpenMax Copyright© 2010 ASHUP Vol 1.1

Figure 2 Hardware setup

1. Pin assignment: There are up to 6 FXS-400/FXO-400 modules on every A2410E board, a module corresponds to a RJ45 port which A2410E takes 2 of 8 pins for a pair connect to your 2-wire telephone line, so each RJ45 socket is divided into 4 telephone lines by a splitter.

Tel 1 Tel 1 Tel 2 Tel 2 Tel 3 Tel 3 Tel 4 Tel 4 TIP/RING of tel 1 TIP/RING of tel 2 TIP/RING of tel 3 TIP/RING of tel 4 NC means not connect

Figure 3 Pin assignment

1. A2410E splitter: It can divide RJ45 port into four ordinary telephone lines, please plug PSTN line into FXO port and normal telephone line corresponds to FXS port.

Close-up of a black USB connector with attached cable, no visible text or symbols

Figure 4 A2410E splitter

3. Software installation and configuration

A2410E/AE2410E support DAHDI software device driver on Linux, to make full use of A2410E/AE2410E, you should download, compile, install and configure DAHDI and asterisk.

3.1 Download

Download DAHDI package to the directory of /usr/src/ from openvox official website http://downloads.openvox.cn/pub/drivers/dahdi-linux-complete/openvox_dahdi-linux-complete-current.tar.gz.

# wget http://downloads.opengvox.cn/pub/drivers/dahdi-linux-complete/openvox_dahdi-linux-complete-current.tar.gz

tar -xvzf openvox\_dahdi-linux-complete-current.tar.gz

3.2 Installation

1. Detect hardware by execute command: lspci -vvvv Check the outcome and confirm your system has recognized A2410E. If identified, outputs are like that:

01:01.0 Communication controller: Unknown device 1b74:2410 (rev 01)
Subsystem: Unknown device 1b74:0001
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr-Stepping- SERR- FastB2B-
Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort-
<MAbort- >SERR- <PERR-
Latency: 64, Cache Line Size: 16 bytes
Interrupt: pin A routed to IRQ 50
Region 0: Memory at dcd00000 (32-bit, non-prefetchable) [size=1M] 

Figure 5 Hardware detection

2. Modify the environment variables

Edit the file named modules under /etc/dahdi/. You are able to comment out drivers unnecessary to load, add opvxa24xx.

<h1 id="x100p-single-port-fxo-interface">X100P - Single port FXO interface</h1>
<h1 id="x101p-single-port-fxo-interface">X101P - Single port FXO interface</h1>
<h1 id="opvxa1200-comment-out-the-unnecessary-driver">opvxa1200    #comment out the unnecessary driver</h1>
<h1 id="ystdm8xx">ystdm8xx</h1>
<h1 id="ystdm16xx">ystdm16xx</h1>
... ...
<h1 id="rhino-481224-channel-analog-pci-interface-card">Rhino 4/8/12/24 Channel Analog PCI Interface Card</h1>
<h1 id="rcbfx">rcbfx</h1>
opvxa24xx    #add opvxa24xx driver 

Figure 6 Modules modification

3. Compile

Unzip and change directory to dahdi-linux-complete-XX, perform command below one by one.

<h1 id="cd-usrsrcdahdi-linux-complete-xx">cd /usr/src/dahdi-linux-complete-XX</h1>
<h1 id="make">make</h1>
<h1 id="make-install">make install</h1>
<h1 id="make-config">make config</h1>

If there is something wrong after "make", please refer to http://bbs.openvox.cn/viewthread.php?tid=1557&extra=page%3D1

If successfully, reboot your PC please.

3.3 Configuration

1. Load opvxa24xx driver

<h1 id="modprobe-dahdi">modprobe dahdi</h1>
<h1 id="modprobe-r-opvxa24xx">modprobe -r opvxa24xx</h1>
<h1 id="modprobe-opvxa24xx-opermodechina">modprobe opvxa24xx opermode=CHINA</h1>

openvox_dahdi-linux-complete 2.2.0 or higher versions allow users to adjust IRQ per millisecond. You are able to modify IRQ by the following way:

# modprobe opvxa24xx opermode=CHINA ms_per_irq=2

ms_per_irq=2 means every 2 milliseconds initiate once IRQ. You may select a valid value of ms_per_irq from 1, 2, 4, 8, 16 according to requirement, the default value is 1. While you download DAHDI from digium official website:

http://downloads.asterisk.org/pub/telephony

DAHDI version above dahdi-linux-complete-2.4.0+2.4.0 supports IRQ adjustment function, and the same method to modify interrupt as described before. After IRQ adjustment, please execute command "dmesg" to check whether you have made the EC module worked. The following figure means EC module has been detected.

Open Vox A2410E version: 1.3 Module 0: Installed -- AUTO Module 1: Installed -- AUTO Module 2: Installed -- AUTO Module 3: Installed -- AUTO Module 4: Installed -- AUTO Module 5: Installed -- AUTO Module 6: Installed -- AUTO Module 7: Installed -- AUTO Module 8: Installed -- AUTO Module 9: Installed -- AUTO Module 10: Installed -- AUT Module 11: Installed -- AUT Module 12: Installed -- AUT Module 13: Installed -- AUT Module 14: Installed -- AUT Module 15: Installed -- AUT Module 16: Installed -- AUT Module 17: Installed -- AUT Module 18: Installed -- AUT Module 19: Installed -- AUT Module 20: Installed -- AUT Module 21: Installed -- AUT Module 22: Installed -- AUT Module 23: Installed -- AUT VPM450: echo cancellation s VPM450: echo cancellation f VPM450: hardware DTMF disab VPM450: Present and oper span (s)

Figure 7 EC module detections

2. Check configuration files

Run command "vim /etc/dahdi/genconf_parameters". If the hardware is AE2410E, please set echo_can to none as following:

echo_can none

While it is A2410E, just ignore that step and keep default.

Execute those commands:

# dahdi_genconf

# dahdi_cfg -vvvv

[root@localhost ~]# dahdi_cfg -vvvv
DAHDI Tools Version - 2.4.0
DAHDI Version: 2.4.0
Echo Canceller(s):
Configuration

================

Channel map:
Channel 01: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 03)
Channel 04: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 04)
Channel 05: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 05)

...
Channel 20: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 20)
Channel 21: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 21)
Channel 22: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 22)
Channel 23: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 23)
Channel 24: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 24)

24 channels to configure.

Setting echocan for channel 1 to none
Setting echocan for channel 2 to none
Setting echocan for channel 3 to none
Setting echocan for channel 4 to none
Setting echocan for channel 5 to none

...
Setting echocan for channel 20 to none
Setting echocan for channel 21 to none
Setting echocan for channel 22 to none
Setting echocan for channel 23 to none
Setting echocan for channel 24 to none 

Figure 8 Channel map

The command dahdi_genconf will automatically generate files /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf. Confirm dahdi-channels.conf is included in chan_dahdi.conf, otherwise, run command:

# echo "#include dahdi-channels.conf" >> /etc/asterisk/chan_dahdi.conf

FXO ports use FXS signaling, while FXS ports adopt FXO signaling. A part of system.conf, which is the basic channel configuration file of A2410E, is displayed.

<h1 id="span-1-opvxa24xx24-openvox-ae2410e-board-25-master">Span 1: OPVXA24XX/24 "OpenVox AE2410E Board 25" (MASTER)</h1>
fxoks=1
fxoks=2
fxoks=3
fxoks=4
...
fxks=21
fxks=22
fxks=23
fxks=24

<h1 id="global-data">Global data</h1>

Loadzone= us
defaultzone= us 

Figure 9 A part of system.conf

In order to match your country pattern, you need to change parameters loadzone and defaultzone to your country. For example, your system is in CHINA, then, you would like them change to:

loadzone = cn
defaultzone = cn 

Meanwhile, you also need to modify another parameter, which is in file /etc/asterisk/indications.conf:

country=cn 

A part of file /etc/asterisk/dahdi-channels.conf is showed as below. (Modification, if it is not agree with the hardware setup)

; Span 1: OPVXA24XX/24"OpenVox A1610 Board 25" (MASTER)
;; line="1 OPVXA24XX/24/0 FXOKS"
signalling=fxo_ks    //FXS ports use FXO signaling
callerid="Channel 1" <4001>
mailbox=4001
group=5
context=from-internal
channel => 1
callerid=
group=
context=default

;; line="2 OPVXA24XX/24/1 FXOKS"
signalling=fxo_ks
callerid="Channel 2" <4002>
mailbox=4002
group=5
context=from-internal
channel => 2
callerid=
group=
context=default

......

;; line="13 OPVXA24XX/24/12"
signalling=fxs_ks    //FXO ports use FXS signaling
callerid=asreceived
group=0
context=from-pstn
channel => 13
callerid=
group=
context=default

;; line="14 OPVXA24XX/24/13"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 14
callerid=
group=
context=default 

Figure 10 A part of dahdi-channels.conf

Check automatically generated files information is agree with your hardware setup, if not, you should modify to your requirements.

After you done works above, reboot your PC please.

3. Start asterisk by executing command: asterisk -vvvvvvvvgc

If asterisk is already activate, run "asterisk -r" instead.

After entering CLI, run command "dahdi show channels". If DAHDI channels are found, it means DAHDI channels have been loaded into asterisk.

3.4 Call test

1. Log in Elastix

Type IP address of Elastix operation system in browser, next come to “Welcome to Elastix” interface, type your username and password. Elastix login interface is like that

> Welcome to Elastix Please enter your username and password Username: Password: Submit Elastix is licensed under GPL by PaloSanto Solutions. 2006 - 2011.

Figure 11 Elastix login interface

elastix System Agenda Email Fax PBX IM Reports Extras Addons My Extension Security Dashboard Network User Management Shutdown Hardware Detector Updates Backup/Restore Preferences System Resources CPU Info: GenuineIntel Intel(R) Celeron(R) D CPU 3.06GHz Uptime: 2 min CPU usage: 38.09% used of 3,066.76 MHz Memory usage: 12.33% used of 1,010.14 Mb Swap usage: 0.00% used of 2,047.99 Mb Processes Status Telephony Service (Astanisk): Running Instant Messaging Service (OpenFire): Service Not Activated Fax Service (Pyifax): Running Email Service (Postfix): Running Database Service (MySQL): Running Web Server (Apache): Running Elastic Call Center Service (Dialer): Not installed N/A N/A N/A Hard Drives Partition Name: /dev/mapper/VolGroup00-LogVol00 Capacity: 70.14GB Usage: 3% Mount point Used Free Performance Graphic Simultaneous calls, memory and CPU Use, calls CPU usage (C) New, usage (NE) News Communication Activity

Figure 12 Elastix interface

2. Hardware detection

Click “system” option, then you will see “hardware detection”, choose it you will see the following outcome.

Span # 1: OPVXA24XX/24 "OpenVox A2410 Board 25" (MASTER) You can set the parameters for these ports here. Configuration of Span FWS 1 FWS 2 FWS 3 FWS 4 FWS 5 FWS 6 FWS 7 FWS 8 FWS 9 FWS 10 FWS 11 FWS 12 Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected BY Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisk Defected by Asterisks

Figure 13 A2410E hardware detection

3. Add SIP extensions

1) Click PBX, extension, choose Generic SIP Device, and finally submit it. You also can refer to the following figure.

elastix® FREEDON TO COMMUNICATE System Agenda Email Fax PBX IM Reports Extras PBX Configuration Operator Panel Voicemail Monitoring Endpoint Configurator Conference Batch of Extensions Apply Configuration Changes Here Basic Extensions Feature Codes General Settings Outbound Routes Trunks Inbound Call Control Inbound Routes Add an Extension Please select your Device below then click Submit Device Device Generic SIP Device Submit

Figure 14 Add a SIP

2) Configure "User Extension", "Display Name", "Secret" these three options, keep others default, and submit your configurations.

Basic Extensions Feature Codes General Settings Outbound Routes Trunks Inbound Call Control Inbound Routes Zap Channel DIDs Announcements Blacklist PIN Sets Paging and Intercom Parking Lot System Recordings Add SIP Extension Add Extension User Extension 6000 Display Name 6000 CID Num Alias SIP Alias Extension Options This device uses sip technology. secret 6000 dtmfmode rfc2833

Figure 15 SIP extension parameters

3) After successfully adding, click "Apply Configuration Changes Here" button to take your configurations effect. Also you are able to add another SIP by click "Add Extension".

Apply Configuration Changes Here Add an Extension Please select your Device below then click Submit Device Device Generic SIP Device Submit Add Extension 6000 <6000>

Figure 16 SIP Apply Configuration

Once add two or more SIP phones, make them effective and registered, you are able to make the soft phones call each other fluently and conveniently.

4. Add analog phones

1) The way to add an analog phone is similar to SIP phone. The figure below will make you clear.

elastix® FREEDON TO COMMUNICATE System Agenda Email Fax PBX IM Reports Extras PBX Configuration Operator Panel Voicemail Monitoring Endpoint Configurator Conference Batch of Extensions Apply Configuration Changes Here Basic Add an Extension Extensions Feature Codes General Settings Outbound Routes Trunks Inbound Call Control Inbound Routes Please select your Device below then click Submit Device Device Generic DAHDI Device Submit

Figure 17 Add analog phones

2) After finishing works above, interface will come to "Add DAHDI Extension", please configure "User Extension", "Display Name", "channel" these three items, and keep others default, finally click the left bottom "submit".

Add DAHDI Extension Add Extension User Extension 4000 Display Name 4000 CID Num Alias SIP Alias Extension Options Device Options This device uses dahdi technology. channel 1

Figure 18 Analog extension configurations

3) Click “Add Extension” button to add more phones, and select device type by your requirement. Finally, click “Apply Configuration Changes here” to make your configurations effective.

Once add two or more analog phones, make them effective and registered, you are able to make calls fluently and conveniently.

1. Configure inbound routes

Click “Inbound Routes”, you may like to fill in “Description” which is optional, and then choose “Extensions” in “Set Destination”. After submitting settings, you are also able to select an extension number you need, submit again, finally “Apply Configuration Changes Here”.

Set Destination Extensions <4000> 4000 Submit Clear Destination & Submit

Figure 19 Inbound routes settings

1. Set outbound routes

Click “Outbound Routes”, set “Route name”, “Dialplan pattern”, “Trunk sequence” these three items to meet your requirements, finally submit changes. The following settings mean all outbound calls through g0 which is an exterior line.

Apply Configuration Changes Here Basic Extensions Feature Codes General Settings Outbound Routes Trunks Inbound Call Control Inbound Routes Zap Channel DIDs Announcements Blacklist CallerID Lookup Sources Day/Night Control Follow Me IVR Queue Priorities Queues Ring Groups Time Conditions Time Groups Internal Options & Configuration Conferences Languages Misc Applications Misc Destinations Music on Hold Add Route Route Settings Route Name: out Route CID: □ Override Extension Route Password: Route Type: □ Emergency □ Intra-Company Music On Hold? default Time Group: ---Permanent Route--- Route Position Last after out Additional Settings PIN Set: None Dial Patterns that will use this Route ( ) + | [X. / ] + Add More Dial Pattern Fields Dial patterns wizards: (pick one) Trunk Sequence for Matched Routes 0 ZAP/g0 Submit Changes

Figure 20 Outbound routes configurations

Additional function

Users should run command “cat /proc/interrupts” to check A2410E has independent interrupt. If A2410E shares interrupt with other device, it may cause some problems even cannot work normally. While A2410E allows users to modify interrupt pin during firmware upgrade for avoid conflict, please visit the following link for details:

http://downloads.openvox.cn/pub/misc/opvx-update%20user%20manual.pdf

4. Reference

www.openvox.cn

www.digium.com

www.asterisk.org

www.voip-info.org

www.asteriskguru.com

www.elastix.org

Tips

Any questions during installation and usage, please consult in our forum or look up for answers from the following websites:

http://bbs.openvox.cn/