A810P - Uncategorized OpenVox - Free user manual and instructions

Product Type	VoIP Analog Gateway
Model	A810P
Brand	OpenVox
Analog Ports	8 FXS ports (RJ11)
Network Interface	1 x 10/100 Base-T Ethernet (RJ45)
Supported Protocols	SIP (RFC3261), IAX2
Voice Codecs	G.711 (A-law/μ-law), G.729, G.723.1, G.726, GSM
Dimensions (W x D x H)	440 x 230 x 44 mm (1U rackmount)
Weight	2.5 kg
Power Supply	AC 100-240V, 50/60Hz (external adapter DC 12V 2A)
Power Consumption	< 15W
Operating Temperature	0°C to 40°C
Storage Temperature	-10°C to 60°C
Humidity	10% to 90% non-condensing
Management	Web GUI, CLI (Telnet/SSH), SNMP
Advanced Features	Caller ID, Call Waiting, Call Transfer, Call Forwarding, Three-way Conference, Do Not Disturb
Security	TLS/SRTP for SIP, HTTPS web access
Certifications	CE, FCC
Maintenance	Restore factory defaults via web GUI or reset button
Spare Parts	Power adapter (DC 12V 2A), rackmount brackets

Frequently Asked Questions - A810P OpenVox

How do I connect the A810P to my PBX or VoIP provider?

Connect an Ethernet cable from the A810P's LAN port to your network. Access the web GUI (default IP: 192.168.1.20) via a browser. Configure SIP trunks under 'Gateway > SIP Configuration' with your provider's server details. Set FXS port parameters under 'Port Settings' and assign extensions.

Can the A810P be used with Asterisk or FreePBX?

Yes, the A810P is fully compatible with Asterisk and FreePBX. Configure it as a SIP peer. Ensure your PBX and gateway are on the same subnet and use matching codecs (e.g., G.711).

How do I reset the A810P to factory defaults?

Use a paperclip to press and hold the Reset button (located on the rear panel) for 10 seconds while the device is powered on. Release when the LEDs blink. Alternatively, go to 'System > Factory Reset' in the web GUI.

What is the default IP address and login credentials?

Default IP: 192.168.1.20. Username: admin, Password: admin. Change these upon first login for security.

How many simultaneous calls can the A810P handle?

The A810P supports up to 8 simultaneous calls (one per FXS port). Call capacity depends on network bandwidth and codec choice.

Does the A810P support fax over IP (FoIP)?

Yes, it supports fax passthrough using G.711 (T.38 not natively supported). Enable fax mode on the desired FXS port in the web GUI for reliable fax transmission.

How do I configure caller ID on the A810P?

In the web GUI, go to 'Port Settings > FXS Port X'. Enable 'Caller ID Type' (FSK or DTMF) and set 'Caller ID Name' and 'Number' as needed. Also ensure your VoIP provider delivers caller ID.

What is the maximum cable length for analog phones connected to the A810P?

FXS ports support loop lengths up to 1 km (using 0.5 mm copper wire) for standard analog phones. Use quality twisted-pair cable for best results.

How do I update the firmware on the A810P?

Download the latest firmware from the OpenVox website. In the web GUI, go to 'System > Firmware Upgrade'. Upload the .bin file and click 'Upgrade'. Do not power off during the process.

Can I use the A810P with a 48V analog phone system?

No, the FXS ports provide -48V DC (standard telco voltage) but are designed for analog phones only. Do not connect to digital phone systems; use with standard analog devices.

USER MANUAL A810P OpenVox

Green PCI card with multiple Ethernet ports and a red circuit board (no visible text or symbols)

AE810E

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 A810E/AE810E 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 10

4. Reference 14

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 A810E/AE810E

1. Overview

1.1 What is A810E/AE810E

A810E is an independent research and development modular analog telephony interface product by OpenVox Communication Co. LTD, and AE810E is A810E with an EC module, they are designed to build SMB PBX. A810E/AE810E 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["Operating System: Nordic FeedSD"] --> B["Open Source Devices: Asterisk®, Zipid, DrillRuff, eSUN and ISDN+ISD Protocols: SIP (AX, SST, MDCP1323, R2 and store... Applications: IVR, CRM, FortCassal, CalCenter, Filing and your application."]
    B --> C["Telcom"]
    C --> D["Asterisk VOIP PBX"]
    D --> E["Switch"]
    E --> F["SIP Phone"]
    D --> G["Analog Phone"]
    G --> H["Analog Card"]
    H --> I["BRI Card"]
    H --> J["PRI Card"]
    H --> K["GSM Card"]
    H --> L["V100 Card"]
    D --> M["PC+SoftPhone"]

Figure 1 Topology

2. Hardware setup

The following matters need your attention before using A810E/AE810E, please check that:

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

Port1-8 OpenVox SMEC V1.2 Power Supply

Figure 2 Hardware setup

1. Please plug PSTN line into FXO port and normal telephone line corresponds to FXS port.

3. Software installation and configuration

A810E supports DAHDI software device driver on Linux. To make full use of A810E, 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

<h1 id="wget-httpdownloadsopengvoxcnpubdriversdahdi-linux-completeopenvox_dahdi-linux-complete-currenttargz">wget http://downloads.opengvox.cn/pub/drivers/dahdi-linux-complete/openvox_dahdi-linux-complete-current.tar.gz</h1>
<h1 id="tar-xvzf-openvox_dahdi-linux-complete-currenttargz">tar -xvzf openvox_dahdi-linux-complete-current.tar.gz</h1>

3.2 Installation

1. Detect hardware by execute command: lspci -vvvv

Check the outcome and confirm your system has recognized A810E. If identified, outputs are like that:

01:02.0 Communication controller: Device 1b74:0810 (rev 01)
Subsystem: Device 1b74:0001
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr- Steppi
SERR- FastB2B- DisINTx-
Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort-
>SERR- <PERR- INTx-
Latency: 64, Cache Line Size: 16 bytes
Interrupt: pin A routed to IRQ 225
Region 0: Memory at ded80000 (32-bit, non-prefetchable) [size=512K]
Kernel driver in use: opvxa24xx
Kernel modules: opvxa24xx 

Figure 3 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>
#rcbfx
Opvxa24xx    #add opvxa24xx driver 

Figure 4 Modules modification

3. Compile

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

Then run "make" again, 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 requirements, the default value is 1. While DAHDI is also available 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.

OpenVox A810E version: 1.3
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Installed -- AUTO FXS/DPO
Module 3: Installed -- AUTO FXS/DPO
Module 4: Installed -- AUTO FXO (FCC mode)
Module 5: Installed -- AUTO FXO (FCC mode)
Module 6: Installed -- AUTO FXO (FCC mode)
Module 7: Installed -- AUTO FXO (FCC mode)
OpenVox VPM: echo cancellation supports 32 channels 

Figure 5 EC module detection

1. Check configuration files

Run command "vim /etc/dahdi/genconf_parameters". If the hardware is AE810E, please set echo_can to none as following:

echo_can none

While it is A810E, just ignore that step and keep default.

Execute those commands:

<h1 id="dahdi_genconf">dahdi_genconf</h1>
<h1 id="dahdi_cfg-vvvv">dahdi_cfg -vvvv</h1>

[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: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 08) 

8 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 6 to none
Setting echocan for channel 7 to none
Setting echocan for channel 8 to none 

Figure 6 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:

<h1 id="echo-include-dahdi-channelsconf-etcasteriskchan_dahdiconf">echo "#include dahdi-channels.conf" >> /etc/asterisk/chan_dahdi.conf</h1>

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

<h1 id="span-1-opvxa24xx24-openvox-a810e-board-25-master">Span 1: OPVXA24XX/24 "OpenVox A810E Board 25" (MASTER)</h1>
fxoks=1
fxoks=2
fxoks=3
fxoks=4
fxks=5
fxks=6
fxks=7
fxks=8

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

loadzone= us
defaultzone= us 

Figure 7 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 A810 Board 25" (MASTER)
;; line="1 OPVXA24XX/24/0 FXOKS (In use)"
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 (In use)"
signalling=fxo_ks
callerid="Channel 2" <4002>
mailbox=4002
group=5
context=from-internal
channel => 2
callerid=
group=
Context=default
......
;; line="7 OPVXA24XX/24/6 FXSKS"
signalling=fxs_ks //FXO ports use FXS signaling
callerid=asreceived
group=0
context=from-pstn
channel => 7
callerid=
group=
context=default

;; line="8 OPVXA24XX/24/7 FXSKS"
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel => 8
callerid=
group=
context=default 

Figure 8 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, and 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 9 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: 39.09% 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 (Astensk): Running Instant Messaging Service (OpenFire): Service Not Activated Fax Service (Hylafax): Running Email Service (Postfix): Running Database Service (MySQL): Running Web Server (Apache): Running Elastix Call Center Service (Dialer): Not installed Hard Drives Partition Name: /dev/mapper/VokGroup00-LogVol00 Capacity: 70.14GB Usage: 3% Mount point 972 Used Free Performance Graphic Simultaneous calls, memory and CPU Six, calls CPU usage (C:) New, Usage (MB) News Communication Activity

Figure 10 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 "OpenVxA810 Board 25" (MASTER) You can set the parameters for these ports here. Configuration of Span RMS 1 RMS 2 RMS 3 RMS 4 RMS 5 RMS 6 RMS 7 RMS 8 Detected by Asterisk Detected by Asterisk Detected by Asterisk Detected by Asterisk Not detected by Asterisk Not detected by Asterisk Not detected by Asterisk Not detected by Asterisk

Figure 11 A810E 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 COMMUNICATIONS 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 12 Add a SIP

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

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

Figure 13 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 <5000>

Figure 14 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® FREEPORT TO COMMUNICATIONS System Agenda Email Fax PBX IM Reports Extras PBX Configuration Operator Panel Yoicemail 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 DAHDI Device Submit

Figure 15 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". For example:

Apply Configuration Changes Here Basic Extensions Feature Codes General Settings Outbound Routes Trunks Inbound Call Control Inbound Routes Zap Channel DIDs Announcements Blacklist Misc Destinations Music on Hold PIN Sets Paging and Intercom 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 16 Analog extension configurations

3) Click “Add Extension” button to add more phones, and select device type by your requirement. Do not forget to 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.

5. 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 17 Inbound routes settings

6. 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 figure means all outbound calls through g0 which is an exterior line. Attention to “Apply Configuration Changes Here” to make your settings effective.

Apply Configuration Changes Here 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 18 Outbound routes configuration

Additional function

Users should run command “cat /proc/interrupts” to check A810E has independent interrupt. If A810E shares interrupt with other device, it may cause some problems even cannot work normally. While A810E 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 or usage, please consult in our forum or look up for answers from the following websites:

http://bbs.openvox.cn/