AXE4DL + EC128L. ATCOM Digital Card AXE4DL User Manual Version: 1.0 2013-07-03



Similar documents
AXE1DL + EC32L. ATCOM Digital Card AXE1DL User Manual Version:

AX400P User Manual. ATCOM Analog Card AX400P User Manual Version:

Content CONTACT ATCOM... 2 CHAPTER 1 THE INTRODUCTION OF AX-2G4A... 3 CHAPTER 2 HARDWARE INTRODUCTION... 5 CHAPTER 3 SOFTWARE INSTALLATION...

Table of Contents. Overview Features Applications Hardware requirement Card dimensions Software Installation...

Content CONTACT ATCOM CHAPTER 7 REFERENCE

OpenVox Communication Co. LTD.

OpenVox Communication Co. LTD.

OpenVox Communication Co.Ltd. OpenVox-Best Cost Effective Asterisk Cards

OpenVox DE210E/DE410E User Manual

Product Guide Version: 1.0

OpenVox D110P/D110E User Manual

This manual contains product information for the GSM Series cards. The manual is organized in the following manner:

Cost Effective Tapping and Call Recording using the Sangoma T116 Card. November 2013

DAHDI User Guide. Schmooze Com Inc.

How to use IP-0x to connect to Skype

SIP Trunking with Elastix. Configuration Guide for Matrix SETU VTEP

Mini IP-PBX PBX PBX120

Merging Old and New Telephony with Asterisk

Cost Effective Tapping and Call Recording using the Sangoma T116. April 2013

IPPBX FAQ. For Firmware Version: V2.0/V

Configuration Guide for connecting the Eircom Advantage 4800/1500/1200 PBXs to the Eircom SIP Voice platform.

VoIP-PSTN Interoperability by Asterisk and SS7 Signalling

PRImaGate Switch RACK 3U

Sangoma Webinar Series Telecom Tapping Solutions. F.Dickey / N.Corbic December 14, 2010

Allo PRI Gateway and Elastix Server

Software Based VoIP Lab A step by step guide to setting up and configuring an IP-PBX. Donal O Connor DNET 4 donaloconnor@gmail.com

OpenVox Communication Co. LTD. OpenVox-Best Cost Effective Asterisk Cards

Internet Telephony Terminology

Link Gate SIP. (Firmware version 1.20)

Open Source Telephony Projects as an Application Development Platform. Frederic Dickey Director Product Management

Mediatrix 3000 with Asterisk June 22, 2011

Overview of Asterisk (*) Jeff Gunther

Configuration Notes 290

Open source VoIP Networks

Dynamix IP PBX-100. User's Manual

Using Asterisk with Odin s OTX Boards

PCI BASED ISDN INTERFACE CARD

EZLoop IP-PBX Enterprise SIP Server

Internet Telephony PBX System

Kerio Operator. Administrator s Guide. Kerio Technologies

2N NetStar. 2N NetStar as a PBX Booster. Quick guide. Version 1.00

Basic configuration of the GXW410x with Asterisk

NCC Blade Network Communication Controller

IP-PBX Quick Start Guide

By Numan Khan

OpenVox A400P/A400E User Manual

IP PBX. SD Card Slot. FXO Ports. PBX WAN port. FXO Ports LED, RED means online

Internet telephony Asterisk system.

Applications between Asotel VoIP and Asterisk

BlueGate SIP. VoIP GSM Gate. Quick Installation guide v 1.0

Digium Communications Hardware

GSM GATEWAY PRI / E1 RACK 3U

MAGIC TH6. System Configuration SW Version 2.000

Setup Reference guide

Micronet VoIP Solution with Asterisk

Configuration Notes 283

Installation and setup guide V 1.0

Quick Start Guide. Vpacket 5100/6100 Voice/Data Router SIP with PRI/ISDN Procedure Release 2.1.1

Setup Guide: on the MyNetFone Service. Revision History

AudioCodes Mediant 1000 Configuration Guide

Trunks User Guide. Schmooze Com Inc.

Mediatrix Gateway 440x Series Quick Configuration Guide

H.KHouyuan Technology Co.,Limited

IP Telephony Center for Small Offices and Remote Branch Offices

Personal VoIP Gateway SKG-300 User Manual

DPH-50U VoIP USB Phone Adapter Quick User Guide

Quick Installation Guide

Asterisk Fast Start. The Asterisk Fast Start course is a three-day course. The class will consist of a combination of lectures and lab exercises.

Personal USB VoIP Gateway User s Guide

LessWires Advanced IP Soft-PBX System

Asterisk: A Non-Technical Overview

How To Use Analog 410 Series Cards

Achieving optimal scalability and voice quality in open source telephony. Konrad Hammel Software Engineer Sangoma Technologies

Wildix W04FXO Whitepaper

MyIC setup and configuration (with sample configuration for Alcatel Lucent test environment)

spiderstar VoIP Interface Version 4.0 User manual

GSM VOIP GATEWAY LEVEL. User Guide. GB with GSM module Two-way converter between VoIP and GSM

TE120 Series TE120P/TE121/TE122

Cisco Unified Communications Express - Quick Configuration Tool User Guide

Extended communication server: Fax Server- Administration

A200 Analog INSTALLATION MANUAL

SIP and H.323. SIP call flow example

SmartPTT Tutorial Telephone Interconnect

Avaya IP Office 8.1 Configuration Guide

Building Robust IPTSP Based on Open Source Technology. Anowar Hasan Sabir, BDCOM Online Ltd. Bangladesh

SIP Internet Telephony Gateway

mobile uc client End user guide

ASTRIBANK USER MANUAL

Softswitch & Asterisk Billing System

SIP Trunking using Optimum Business SIP Trunk Adaptor and the Panasonic KX-NCP500 IP PBX V2.0502

Khomp KGSM-USB SPX and Elastix Server

General Guidelines for SIP Trunking Installations

White Paper. Open Source Telephony: The Evolving Role of Hardware as a Key Enabler of Open Source Telephony in the Business Market.

Extending Open Source PBX For Scalable Media Gateways

Crash Course in Asterisk

1 VoIP/PBX Axxess Server

Extend the Life of Your Legacy PBX while Benefiting from SIP Trunks. December 5, 2013

Transcoding (Voice coding G.711 A-law/u-law, G.722 G.729AB) licenses included

Mediatrix 4404 Step by Step Configuration Guide June 22, 2011

How to Config MTG1000B With T1 and Elastix

Transcription:

AXE4DL + EC128L ATCOM Digital Card AXE4DL User Manual Version: 1.0 2013-07-03

Content CHAPTER 1 THE INTRODUCTION OF AXE4DL...3 CHAPTER 2 HARDWARE INTRODUCTION...5 CHAPTER 2 TEST ENVIRONMENT... 8 CHAPTER 3 SOFTWARE INSTALLATION...9 CHAPTER 4 SOFTWARE CONFIGURATION...11 CHAPTER 5 TEST...15 CHAPTER 6 REFERENCE... 17 www.atcom.cn 1

Contact ATCOM The Introduction of ATCOM ATCOM is the leading VoIP hardware manufacturer in global market. We have been keeping innovating with customer s needs oriented, working with partners to establish a total solution for SMB VoIP with IP phone, IP PBX and Asterisk cards. With over 10 years experience of R&D, manufacturing and service in network and VoIP filed; mission of creating the biggest value for IP terminals, we commit ourselves in supplying the competitive IP phone and other terminals for IP PBX, softswitch, IMS, NGN providers and carriers; supplying the competitive total VoIP solution for SMB market. We keep improving the customer s experience and creating the bigger value with our reliable products. Until now, our VoIP products has been sold to over 60 countries and used by millions of end users. Contact sales: Address District C, east of 2nd floor, #3, Crown industry buildings, Chegongmiao Industry area, Futian district, Shenzhen, China Tel +(86)755-23487618 Fax +(86)755-23485319 E-mail sales@atcomemail.com Contact Technical Support: Tel +(86)755-23481119 E-mail Support@atcomemail.com Website address: http://www.atcom.cn/ Download Center: http://www.atcom.cn/download.html www.atcom.cn 2

Chapter 1 the Introduction of AXE4DL Overview of the AXE4DL AXE4DL Asterisk card is the asterisk PCI-E card which supports four ISDN PRI E1/T1/J1 ports, optional with teleco-grade hardware echo cancellation. Using AXE4DL digital PRI card, open source Asterisk PBX and stand alone PC, users can create their IP PBX telephony solution included all the sophisticated features of traditional PBX, and extended features such as voicemail in IP PBX. With low profile design, the AXE4DL is suitable for 2U server. Features AXE4DL: Four ISDN PRI E1/T1/J1 ports Support original Dahdi, Zaptel(No patch) Support Asterisk, Freeswitch, Yate Support Elastix, Trixbox, AsteriskNOW, PBX in a Flash 100% compatible with all features of Asterisk PBX Supports standard telephony and data protocols(including PRI, CAS, CCS for voice and PPP, HDLC, Cisco HDLC and Frame Relay for data modes) Supports chan_ss7 and Openr2 Optional hardware echo cancellation Select the hardware echo cancellation direction With low profile design, the AXE4DL is suitable for 2U server. Applications ISDN PRI IP PBX ISDN least cost router Calling Card Platforms IVR system Call Center Traditional Calls/VoIP Calls Conference VoIP Gateway Callback Service Optional DSP Hardware Echo Cancellation Module G.168 2002 echo cancellation in hardware 1024 taps/128 ms tail per channel on all channel densities DTMF decoding and tone recognition Voice quality enhancement: music protection, acoustic echo control, and adaptive noise reduction Does not increase the physical size of the card, and no additional slot is required www.atcom.cn 3

Voice Modes PRI CPE and PRI NET: EuroISDN 4ESS(AT&T) 5ESS(Lucent) DMS100 Hardware Requirement 1.6-Ghz Pentium IV 512 MB RAM PCI-E slot AXE4DL Dimension 120mm (Length)*64mm (height) Operating System Linux (all versions, releases and distributions from 1.0 up) www.atcom.cn 4

AXE4DL User Manual with DAHDI Chapter 2 Hardware Introduction AXE4DL The Front View of EC128L www.atcom.cn 5

LED: The LED of the four ports will be red clearly, when the driver of the card is loaded correctly and the /etc/dahdi/system.conf file is configured correctly, ; the LED will be green clearly, when the PRI line is connected correctly, and it synchronizes to the other equipment successfully,. C2: It is used for selecting E1, T1, J1 mode. In default, it will be set it up according to users requirement before shipping out. Users can set up E1, T1/J1 like the following: For E1 Mode For T1 and J1 Mode C4: It is used when there are more than one PRI card in the one server; For example: if users have two cards in the server, then turn it to 0 in the one card, and turn it to 1 in the other card; If users have four cards in the server simultaneously, then turn it to 0,1,2,3 respectively. www.atcom.cn 6

C1: It is used to select the hardware echo cancellation direction. In default, It will not use the jumper, and it will delete the echo for the far end side; if users need delete the echo for the local side, please use the jumper like the following illustration. Delete the echo for the local side Delete the echo for the far end side C3: It is not available. www.atcom.cn 7

Chapter 2 Test Environment Test Environment: Libpri-1.4.7 dahdi-linux-complete-2.6.2+2.6.2 asterisk-1.8.7.0 Centos6.0 AXE4DL+EC128L (kernel version: 2.6.32-279.22.1.el6.i686) www.atcom.cn 8

Chapter 3 Software Installation After inserting the card into the PCI slot and boot the server, please use the lspci command to check the PCI bus compatibility. From the correct output, users can see the following line: ---------------------------------------------------------------------------------------------------------------------- 03:01.0 Communication controller: Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) ---------------------------------------------------------------------------------------------------------------------- The TE410P will be found. if users can not see one line like the line above, please power off the server and try to use another PCI slot; If it does not help still, users have to check the compatibility issue between the card and the PCI-E bus. Please try to use another PCI-E slot or other server and try it again. 1. To install asterisk and dahdi, users have to use yum command to install the following prerequisite packages: yum install -y bison bison-devel zlib zlib-devel openssl openssl-devel gnutls-devel gcc gcc-c++ ncurses ncurses-devel 2. Download libpri, dahdi, and asterisk packages Notice: If uses need use the latest version, please check them on asterisk download center. [root@localhost src]# wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4.14.tar.gz [root@localhost src]# wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.6. 2+2.6.2.tar.gz [root@localhost src]# wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.8.7.0.tar.gz www.atcom.cn 9

3. Installation the packages Install libpri 1) [root@localhost src]# tar -xvzf libpri-1.4.14.tar.gz 2) [root@localhost libpri-1.4.14]# make 3) [root@localhost libpri-1.4.14]# make install Install dahdi-linux 4) [root@localhost src]# tar -xvzf dahdi-linux-complete-2.6.2+2.6.2.tar.gz 5) [root@localhost src]# cd dahdi-linux-complete-2.6.2+2.6.2 6) [root@localhost dahdi-linux-2.6.2]# make 7) [root@localhost dahdi-linux-2.6.2]# make install 8) [root@localhost dahdi-linux-2.6.2]# make config Install asterisk 1) [root@localhost src]# tar -xvzf asterisk-1.8.7.0.tar.gz 2) [root@localhost asterisk-1.8.7.0]#./configure 3) [root@localhost asterisk-1.8.7.0]# make 4) [root@localhost asterisk-1.8.7.0]# make install 5) [root@localhost asterisk-1.8.7.0]# make samples Notice: if uses run the command: make samples, all the asterisk configuration files will be set as the samples. www.atcom.cn 10

Chapter 4 Software Configuration 1. Please run the dahdi_genconf command to configure the /etc/dahdi/system.conf file and generate /etc/asterisk/dahdi-channels.conf file. [root@localhost ~]# dahdi_genconf It will not show any output when the command run successfully. After do that successfully, it will get the following configuration in the system.conf file with E1 mode. [root@localhost ~]# cat /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf do not hand edit # This file is parsed by the Dahdi Configurator, dahdi_cfg # Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31............ # Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" span=4,4,0,ccs,hdb3,crc4 # termtype: te bchan=94-108,110-124 dchan=109 echocanceller=mg2,94-108,110-124 # Global data loadzone = us defaultzone = us www.atcom.cn 11

After running dahdi_genconf successfully, the dahdi-channels.conf file will get the following configuration. [root@localhost ~]# cat /etc/asterisk/dahdi-channels.conf ; Autogenerated by /usr/sbin/dahdi_genconf on Mon Aug 5 15:02:51 2013 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 1-15,17-31 context = default group = 63............ ; Span 4: TE4/0/4 "T4XXP (PCI) Card 0 Span 4" group=0,14 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel => 94-108,110-124 context = default group = 63 2. Please add the line: #include dahdi-channels.conf into the end of the chan_dahdi.conf file. And users can run the following command to add it to the file. [root@localhost ~]# echo #include dahdi-channels.conf >> /etc/asterisk/chan_dahdi.conf 3. Please run asterisk with the following command: [root@localhost ~]# asterisk [root@localhost ~]# asterisk -vvgr [root@localhost ~]# reload 4. Please run the dahdi show channels command in the CLI. Users will get 124 channels. www.atcom.cn 12

5. How to check if the hardware echo cancellation works or not 1) After booting the server with the AXE4DL and the echo cancellation module, please run the dmesg command to check if the hardware is detected or not. Users will get the following information. VPM450: echo cancellation for 128 channels wct4xxp 0000:05:04.0: VPM450: hardware DTMF disabled. wct4xxp 0000:05:04.0: VPM450: Present and operational servicing 4 span(s) 2) When users are using the hardware echo cancellation module, please disable the software echo cancellation in /etc/dahdi/system.conf, and enable the parameter: echocancel=yes line in the /etc/asterisk/chan_dahdi.conf file. After do that, please restart dahdi and asterisk. 3) Run asterisk, and make a call by channel 1, users can get the following information marked with red line, if the the echo cancellation module is working. Notice: If users enable the software echo cancellation, it can get the following red lines also. So please disable the software echo cancellation when users use the hardware cancellation. localhost*cli> dahdi show channel 1 Channel: 1 File Descriptor: 17 Span: 1 Extension: 300 Dialing: no Context: from-pstn Caller ID: 900 Calling TON: 33 Caller ID name: 900 Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: ISDN PRI Radio: 0 Owner: DAHDI/1-1 Real: DAHDI/1-1 Callwait: <None> Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Busy Detection: no www.atcom.cn 13

TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps currently ON Wait for dialtone: 0ms PRI Flags: Call PRI Logical Span: Implicit www.atcom.cn 14

Chapter 5 Test 1 LED state LED: If the driver of the card is loaded correctly and the /etc/dahdi/system.conf file is configured correctly, the four ports of LED will be red clearly. If the PRI line is connected correctly, and it synchronizes to the other equipment successfully, then the LED will be green clearly. 2 Use the E1/T1/J1 cables to connect to the ports. And Check the PRI state 1) Load the asterisk [root@localhost asterisk]# asterisk vvgr *CLI> reload 2) Check the PRI state users will get the following port information, all the four ports are in up status. localhost*cli> pri show spans PRI span 1/0: Up, Active PRI span 2/0: Up, Active PRI span 3/0: Up, Active PRI span 4/0: Up, Active 3 Add the dial plan for the PBX Please edit the dial-plan in the extensions.conf file, users can refer to the following example. [from-internal] exten=>_1.,1,dial(dahdi/i1/${exten:1}) exten=>_1.,2,hangup() exten=>_2.,1,dial(dahdi/i2/${exten:1}) exten=>_2.,2,hangup() exten=>_3.,1,dial(dahdi/i3/${exten:1}) exten=>_3.,2,hangup() exten=>_4.,1,dial(dahdi/i4/${exten:1}) exten=>_4.,2,hangup() [from-pstn] exten=>s,1,answer() exten=>s,2,dial(sip/500) exten=>s,3,hangup() www.atcom.cn 15

4 Use a registered sip phone to make an outgoing call. According to the dial-plan above, if users want to make the outgoing call via the first port, users have to add a prefix 1 ; If users want to make a calling via the second port, users have to add a prefix 2. For example, the destination number is 0755-83018618, so dialing 0755-83018618, And then users can get the following output from asterisk CLI: ------------------------------------------------------------------------------------------------------------------ *CLI> == Using SIP RTP CoS mark 5 --Executing [83018618@from-internal:1] Dial("SIP/500-00000003", "dahdi/1/83018618") in new stack -- Called 1/83018618 -- DAHDI/1-1 answered SIP/500-00000003 ------------------------------------------------------------------------------------------------------------- The called party ringing, and pick up the phone, so the call is successful. 5 Please use a phone to make an incoming call. For example, the number of the PRI line is 0755-23485319, using a phone to dial 0755-23485319. The sip 500 rings, and users can get the following output from asterisk CLI: ---------------------------------------------------------------------------------------------------------------- *CLI> -- Starting simple switch on 'DAHDI/1-1' [Jul 1 11:42:53] NOTICE[20678]: chan_dahdi.c:8734 ss_thread: Got event 18 (Ring Begin)... -- Executing [s@from-pstn:1] Answer("DAHDI/1-1", "") in new stack -- Executing [s@from-pstn:2] Dial("DAHDI/1-1", "sip/500") in new stack == Using SIP RTP CoS mark 5 -- Called 500 -- SIP/500-00000004 is ringing -- SIP/500-00000004 answered DAHDI/1-1 ---------------------------------------------------------------------------------------------------------------- Users can test other three ports in the same way. www.atcom.cn 16

Chapter 6 Reference http://www.asteriskguru.com/ http://www.asterisk.org/downloads http://www.atcom.cn/ www.atcom.cn 17