iAG840 - Telephony OpenVox - Free user manual and instructions

USER MANUAL iAG840 OpenVox

IAG840/IAG880 User Manual

White OpenVox network device on a blue abstract background with flowing blue light effects (no text or symbols on device)

Address: 10/F, Building 6-A, Baoneng Science and Technology Industrial Park, Longhua New District, Shenzhen, Guangdong, China 518109

Tel: [phone]

Fax: [phone]

Business Contact: [e-mail]

Technical Support: [e-mail]

Business Hours: 09:00-18:00 (GMT+8) from Monday to Friday

URL: www. openvox. cn

Version1.0 (2016-03-03)

Copyright

Copyright ^® 2013 OpenVox Inc. All rights reserved. No part of this document may be reproduced without prior written permission.

Condenality

Informaon contained herein is of a highly sensitive nature and is condenal and proprietary to OpenVox Inc. No part may be distributed, reproduced or disclosed orally or in wrien form to any party other than the direct recipients without the express wrien consent of OpenVox Inc.

Disclaimer

OpenVox Inc. reserves the right to modify the design, characteriscs, and products at any me without nocaon or obligaon 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 eort to ensure that the informaon contained in this document is accurate and complete; however, the contents of this document are subject to revision without noce. Please contact OpenVox to ensure you have the latest version of this document.

Trademarks

All other trademarks menoned in this document are the property of their respective owners.

Table of Contents

1. Overview....1

What is iAG840/880 ? 1

Sample Applicaon 2

Product Appearance 2

Main Features 4

Physical Informaon 4

Soware 5

2. System 6

Status 6

Time 6

Login Sengs 7

General, Tools and Informaon 8

Language Sengs 8

Scheduled Reboot 9

Reboot Tools 9

Informaon 11

3. Analog....12

Channel Sengs 12

Dial Matching Table 13

Global Sengs 14

4. SIP....18

SIP Endpoints 18

Main Endpoint Sengs 18

Advanced: Registraon Opons 21

Call Sengs 22

Advanced: Signaling Sengs 22

Advanced: Timer Sengs 23

Media Sengs 24

Batch SIP Endpoint 24

Advanced SIP Sengs 25

Networking 25

NAT Sengs 25

Advanced: NAT Sengs 26

Parsing and Compatibility 27

Security 28

Media 29

5. Network, Advanced and Logs....30

Network 30

Network Sengs 30

OpenVPN Sengs 32

DDNS Sengs 33

Toolkit 34

Advanced 34

Asterisk API 34

Asterisk CLI 36

Asterisk File Editor 37

Logs 38

1. Overview

What is iAG840/880?

This document is to explain the quad-FXS module of analog gateway.

OpenVox Analog Gateway is an open source asterisk-based Analog VoIP Gateway soluon for SMBs and SOHOs. With friendly GUI and unique modular design, users may easily setup their customized Gateway. Also secondary development can be completed through AMI (Asterisk Management Interface).

There are four models with Analog Gateway, the 4FXS, 8FXS, 4FXO and 8FXO, and there are 4/8 ports in iAG840/880. The Modular Design Analog Gateways are developed for interconnecng the PSTN networks with a wide selecon of codecs and signaling protocol, including G.711A, G.711U, G.729, G.722, G.723, ILBC to quickly reduce communicaon expenses and maximize cost-savings.

The Analog gateway use standard SIP protocol and compatible with Leading IMS/NGN plaorm, IPPBX and SIP servers, support most of the VoIP operang plaorms such as Asterisk, Elasx, 3CX, FreeSWITCH, Broadso etc.

Sample Application

graph TD
    A["IP PBX"] --> B["IP Switch"]
    C["PSTN"] --> D["iAG FXO Gateway"]
    B --> D
    D --> E["LAN"]
    F["iAG FXS Gateway"] --> G["IP Phone"]
    F --> H["FAX"]
    I["IP Phone"] --> G
    J["IP Phone"] --> H
    K["IP Phone"] --> L["IP Phone"]

Figure 1-2-1 Topological Graph

Product Appearance

The picture below is appearance of Analog Series Gateway.

White OpenVox device with indicator lights and ports (no readable text or symbols beyond branding)

Front view of a network switch device showing ports, connectors, and ports (no visible text or labels)

Figure 1-3-1 Product Appearance

(○) 2 3 1 2 3

Figure 1-3-2 Front Panel

1 : System LED
2 : Network interface LED
3: Power Indicator

1 5 6 7 8 1 2 3 4 2 5 6 7 8 1 2 3 4 3 4 End DC-12V Reload 6 7

Figure 1-3-4 Back Panel

1 : Analog Telephone Interface (8)
2 : Channel indicator (8)
3: USB Interface(1)
4 : Ethernet ports (1)
5: Fan vent
6 : Power socket
7: Reload buon

Main Features

Modular design
Based on Asterisk OR
▶ Editable Asterisk ^OR conguraon le
Support T.38 fax relay and T.30 fax transparent, can connually fax mulple page
Echo cancellaon and Stac jier buer
Wide selecon of codecs and signaling protocol
DTMF relay
➢ Ring cadence and frequency seng
MWI(Message waing indicator)
DHCP, DNS/DDNS, NAT Network
VAG and CNG
All hot-swap
➢ Stable performance, exible dialing, friendly GUI
▶ Two-year me warranty

Physical Information

Table 1-5-1 Description of Physical Information

Weight	580g
Size	21cm*21cm*3.6cm
Temperature	-20~70°C (Storage)
	0~40°C (Operaon)
Operaon humidity	10%~90% non-condensing
Power source	12V DC/4A
Max power	16W
LAN port	1

Software

Default IP: 172.16.99.1

Username: admin

Password: admin

Please enter the default IP in your browser to scan and congure the module you want.

Authentication Required The server http://172.16.8.125:80 requires a username and password. The server says: Openvox-Analog- Gateway. User Name: admin Password: ***** Log In Cancel

Figure 1-6-1 LOGIN Interface

2. System

Status

On the "Status" page, you will see Port/SIP/Network informaon and status.

Port	Type	Caller ID	Sip Account	Port Status
1	FXS	305	305	OnHook
2	FXS	306	306	OnHook
3	FXS	8003		OnHook
4	FXS	8004		OnHook
5	FXS	8005		OnHook
6	FXS	8006		OnHook
7	FXS	8007		OnHook
8	FXS	8008		OnHook

Endpoint Name	User Name	Host	Registration	SIP Status
305	305	172.16.99.97	client	Registered
306	306	172.16.99.97	client	Registered

Network Information
Name	MAC Address	IP Address	Mask	Gateway	RX Packets	TX Packets
LAN	A0:98:05:01:15:67	172.16.9.125	255.255.0.0	172.16.0.1	15175	1019

Figure 2-1-1 System Status

Time

Table 2-2-1 Description of Time Settings

Opons	Denion
System Time	Your gateway system me.
Time Zone	The world me zone. Please select the one which is the same or the closest as your city.
POSIX TZ String	Posix me zone strings.
NTP Server 1	Time server domain or hostname. For example, [me.asia.apple.com].
NTP Server 2	The rst reserved NTP server. For example, [me.windows.com].
NTP Server 3	The second reserved NTP server. For example, [me.nist.gov].
Auto-Sync from NTP	Whether enable automacally synchronize from NTP server or not. ON is enable, OFF is disable this funcon.
Sync from NTP	Sync me from NTP server.
Sync from Client	Sync me from local machine.

For example, you can congure like this:

Time Settings System Time: 2013-9-11 17:46:41 Time Zone: Chongqing POSIX TZ String: CST-8 NTP Server 1: us.pool.ntp.org NTP Server 2: 64.236.96.53 NTP Server 3: time.windows.com Auto-Sync from NTP: ON Sync from NTP Sync from Client

Figure 2-2-1 Time Settings

You can set your gateway me Sync from NTP or Sync from Client by pressing dierent buons.

Login Settings

Your gateway doesn't have administraon role. All you can do here is to reset what new username and password to manage your gateway. And it has all privileges to operate your gateway. You can modify both your "Web Login Sengs" and "SSH Login Sengs". If you have changed these sengs, you don't need to log out, just rewring your new user name and password will be OK.

Table 2-3-1 Description of Login Settings

Opons	Denion
User Name	Dene your username and password to manage your gateway, without space here. Allowed characters"-+_+. <>&0-9a-zA-Z". Length: 1-32 characters.
Password	Allowed characters "-+_+. <>&0-9a-zA-Z".Length: 4-32 characters.
ConrmPassword	Please input the same password as 'Password' above.

Web Login Settings User Name: admin Password: ...... Confirm Password: ...... SSH Login Settings Enable: ON User Name: admin Password: admin Port: 12345

Figure 2-3-1 Login Settings

Notice: Whenever you do some changes, do not forget to save your conguraon.

General, Tools and Information

Language Settings

You can choose dierent languages for your system. If you want to change language, you can switch "Advanced" on, then "Download" your current language package. Aer that, you can modify the package with the language you need. Then upload your modified packages, "Choose File" and "Add", those will be ok.

Language Settings Language: English Advanced: ON Download: Download selected language package. Download Delete: Delete selected language. Delete Add New Language: New language Package: 浏览... Add

Figure 2-4-1 Language Settings

Scheduled Reboot

If switch it on, you can manage your gateway to reboot automacally as you like. There are four reboot types for you to choose, "By Day, By Week, By Month and By Running Time".

Scheduled Reboot Enabled: ON Reboot Type: By Day By Day By Week By Month By Running Time : 0 Minute: 0 Running Time: Save

Figure 2-4-2 Reboot Types

If use your system frequently, you can set this enable, it can helps system work more ecient.

Reboot Tools

On the "Tools" pages, there are reboot, update, upload, backup and restore toolkits. You can choose system reboot and Asterisk reboot separately.

Reboot Tools The page at 172.16.179.1 says: Are you sure to reboot your gateway now? You will lose all data in memory! Reboot the gateway and all the current calls will be Reboot the asterisk and all the current calls will be OK Cancel System Reboot Asterisk Reboot

Figure 2-4-3 Reboot Prompt

If you press "Yes", your system will reboot and all current calls will be dropped. Asterisk Reboot is the same.

Table 2-4-1 Instruction of reboots

Opons	Denion
System Reboot	This will turn o your gateway and then turn it back on. This will drop all current calls.
Asterisk Reboot	This will restart Asterisk and drop all current calls.

We oer two kinds of update types for you, you can choose System Update or System Online Update. System Online Update is an easier way to update your system.

Update Firmware New system file: 浏览... System Update New system file is downloaded from official website and update system. System Online Update

Figure 2-4-4 Update Firmware

If you want to store your previous conguraon, you can rst backup conguraon, then you can upload conguraon directly. That will be very convenient for you.

Upload Configuration New configuration file: Choose File No file chosen File Upload Backup Configuration Current configuration file version: 1.0.1 Download Backup

Figure 2-4-5 Upload and Backup

Sometimes there is something wrong with your gateway that you don't know how to solve it, mostly you will select factory reset. Then you just need to press a buon, your gateway will be reset to the factory status.

Restore Configuration This will cause all the configuration files to back to default factory values! And reboot your gateway once it finishes. Factory Reset

Figure 2-4-6 Factory Reset

Information

On the “Informaon” page, there shows some basic informaon about the analog gateway. You can see soware and hardware version, storage usage, memory usage and some help informaon.

Model Name:	YS-AGU-E2M0800
$software Version:	1.1.0
Hardware Version:	1.0.0
Slot Number:	1
Storage Usage:	1.7M/63.5M (3%)
Memory Usage:	59.067 % Memory Clean
Build Time:	2016-03-01 17:26:17
Contact Address:	10/F, Building 6-A, Baoneng Science and Technology Industrial Park, Longhua New District Shenzhen, Guangdong, China 518109
Tel:	[phone]
Fax:	[phone]
E-Mail:	[e-mail]
Web Site:	www.openvox.cn
System Time:	2016-3-3 14:32:39
System Uptime:	0 days 03:59:26

Figure 2-4-7 System Information

3. Analog

You can see much informaon about your ports on this page.

Channel Settings

Port	Type	Caller ID	Sip Account	Port Status	Actions
1	FXS	301	301	OnHook
2	FXS	8002	None	OnHook
3	FXS	8003	None	OnHook
4	FXS	8004	None	OnHook
5	FXS	8005	None	OnHook
6	FXS	8006	None	OnHook
7	FXS	8007	None	OnHook
8	FXS	8008	None	OnHook

Figure 3-1-1 Channel System

On this page, you can see every port status, and click acon buon to congure the port.

Port 1 ▼ General Port type: FXS Rx gain: 0.0 Tx gain: 0.0 Ring timeout: 8000 Sip Account: None ▼ ▼ Caller ID Caller ID: 8001 Full name: Channel 8001 CID signalling: bell ▼ Save Cancel

Figure 3-1-2 Port Configure

Dial Matching Table

Dialing rules is used to eecvely judge whether the received number sequence is complete, in order to mely end receiving number and send out number

The correct use of dial-up rules, helps to shorten the turn-on me of phone call

_01[358]XXXXXXXXX
_010XXXXXXXXX
_02XXXXXXXXX
_0[3-9]XXXXXXXXX
_11[02-9]
_111XX
_9[56]XXX
_100XX
_10[1-9]
_12[0-24-9]
_1[358]XXXXXXXXX
_[235-7]XXXXXXXXX
_[48][1-9]XXXXX
_[48]0[1-9]XXXXX
_[48]00XXXXXXXXX
_#XX
_*XX
##
_X.
Dial Matching rule may be numbers, letters, or combinations thereof. If an rule is prefixed by a '_' character, it is interpreted as a pattern rather than a literal. In patterns, some characters have special meanings:
X - any digit from 0-9
Z - any digit from 1-9
N - any digit from 2-9
[1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
! - wildcard, causes the matching process to complete as soon as ;it can unambiguously determine that no other matches are possible
For example, the rule _NXXXXXXXX would match normal 7 digit dialings, while _1NXXXXXXXX would represent an area code plus phone number preceded by a one. 

Figure 3-2-1 Port Configure

Global Settings

General Tone duration: 100 Dial timeout: 180 Codec: Ulaw Impedance: Fcc Echo cancel tap length: 512 VAD/CNG: Flash/Wink: ✓ Max flash time: 400 "#" as Ending Dial Key:

Figure 3-3-1 General Configuration

Table 3-3-1 Instruction of General

Opons	Denion
Tone duraon	How long generated tones (DTMF and MF) will be played on the channel. (in milliseconds)
Dial meout	Species the number of seconds we aempt to dial the specied devices.
Codec	Set the global encoding : mulaw, alaw.
Impedance	Conguraon for impedance.
Echo cancel tap length	Hardware echo canceler tap length.
VAD/CNG	Turn on/o VAD/CNG.
Flash/Wink	Turn on/o Flash/wink.
Max ash me	Max ash me.(in milliseconds).
“#”as Ending Dial Key	Turn on/o Ending Dial Key.

Caller ID The pattern of sending CID: send CID after first ring Waiting time before sending CID: 100 Send polarity reversal(DTMF Only): Start code(DTMF Only): Stop code(DTMF Only):

Figure 3-3-2 Caller ID

Table 3-3-2 Instruction of Caller ID

Opons	Denion
The paern of sending CID	Some countries(UK) have ring tones with dierent ring tones(ring-ring), which means the caller ID needs to be set later on, and not just aer the rst ring, as per the default(1).
Waing me before sending CID	How long we will waing before sending the CID on the channel.(in milliseconds).
Sending polarity reversal(DTMF Only)	Send polarity reversal before sending the CID on the channel.
Start code(DTMF Only)	Start code.
Stop code(DTMF Only)	Stop code.

Hardware gain FXS Rx gain: 0 FXS Tx gain: 0

Figure 3-3-3 Hardware Gain

Table 3-3-3 Instruction of Hardware gain

Opons	Denion
FXS Rx gain	Set the FXS port Rx gain. Range: -35, 0 or 35.
FXS Tx gain	Set the FXS port Tx gain. Range: -35, 0 or 35.

Fax Mode: Adaptive Rate: No speed limit Ecm: ✓

Figure 3-3-4 Fax Configuration

Table 3-3-4 Definition of Fax

Opons	Denion
Mode	Set the transmission mode.
Rate	Set the rate of sending and receiving.
Ecm	Enable/disable T.30 ECM (error correcon mode) by default.

Country Country: United States / North America Ring cadence: 2000,4000 Dial tone: 350+440 Ring tone: 440+480/2000,0/4000 Busy tone: 480+620/500,0/500 Call waiting tone: 440/300,0/10000 Congestion tone: 480+620/250,0/250 Dial recall tone: !350+440/100,10/100,!350+440/100,10/100,!350+440/100,10/100,350+4- Record tone: 1400/500,0/15000 Info tone: !950/330,!1400/330,!1800/330,0 Stutter tone: !350+440/100,10/100,!350+440/100,10/100,!350+440/100,10/100,!350+4

Figure 3-3-5 Country Configuration

Table 3-3-5 Definition of Country

Opons	Denion
Country	Conguraon for locaon specic tone indicaons.
Ring cadence	List of duraons the physical bell rings.
Dial tone	Set of tones to be played when one picks up the hook.
Ring tone	Set of tones to be played when the receiving end is ringing.
Busy tone	Set of tones played when the receiving end is busy.
Call waing tone	Set of tones played when there is a call waing in the background.
Congeson tone	Set of tones played when there is some congeson.
Dial recall tone	Many phone systems play a recall dial tone aer hook ash.
Record tone	Set of tones played when call recording is in progress.
Info tone	Set of tones played with special informaon messages (e.g., number is out of service.)
Stuer tone

4. SIP

SIP Endpoints

This page shows everything about your SIP, you can see status of each SIP.

Endpoint Name Registration Credentials Actions 301 client 301@172.16.8.63 Add New SIP Endpoint Delete

Figure 4-1-1 SIP Status

You can click buon to add a new SIP endpoint, and if you want to modify

existed endpoints, you can click buon.

Main Endpoint Settings

There are 3 kinds of registraon types for choose. You can choose "Anonymous, Endpoint registers with this gateway or This gateway registers with the endpoint".

You can congregate as follows:

If you set up a SIP endpoint by registraon "None" to a server, then you can't register other SIP endpoints to this server. (If you add other SIP endpoints, this will cause Out-band Routes and Trunks confused.)

Main Endpoint Settings Name: 301 User Name: Anonymous Password: Registration: None Hostname or IP Address: 172.16.8.63 Transport: UDP NAT Traversal: Yes SUBSCRIBE for MWI: No

Figure 4-1-2 Anonymous Registration

For convenience, we have designed a method that you can register your SIP endpoint to your gateway, thus your gateway just work as a server.

Main Endpoint Settings Name: 301 User Name: 301 Anonymous Password: 301 Registration: Endpoint registers with this gateway Hostname or IP Address: dynamic Transport: UDP NAT Traversal: Yes SUBSCRIBE for MWI: No

Figure 4-1-3 Register to Gateway

Also you can choose registraon by "This gateway registers with the endpoint", it's the same with

"None", except name and password.

Main Endpoint Settings Name: 301 User Name: 301 Anonymous Password: 301 Registration: This gateway registers with the endpoint Hostname or IP Address: 172.16.0.88 Transport: UDP NAT Traversal: Yes SUBSCRIBE for MWI: No

Figure 4-1-4 Register to Server

Table 4-1-1 Definition of SIP Options

Opons	Denion
Name	A name which is able to read by human. And it's only used for user's reference.
Username	User Name the endpoint will use to authenticate with the gateway.
Password	Password the endpoint will use to authenticate with the gateway. Allowed characters.
Registraon	None---Not registering;Endpoint registers with this gateway---When register as this type, it means the GSM gateway acts as a SIP server, and SIP endpoints register to the gateway;This gateway registers with the endpoint---When register as this type, it means the GSM gateway acts as a client, and the endpoint should be register to a SIP server;
Hostname or IP Address	IP address or hostname of the endpoint or 'dynamic' if the endpoint has a dynamic IP address. This will require registraon.
Transport	This sets the possible transport types for outgoing. Order of usage, when the respective transport protocols are enabled, is UDP, TCP, TLS. The rst enabled transport type is only used for outbound messages until a Registraon takes place. During the peer Registraon the transport type may change to another supported type if the peer requests so.
NAT Traversal	Addresses NAT-related issues in incoming SIP or media sessions.No---Use Rport if the remote side says to use it.Force Rport on---Force Rport to always be on.Yes---Force Rport to always be on and perform comedia RTP handling.Rport if requested and comedia---Use Rport if the remote side says to use it and perform comedia RTP handling.

Advanced: Registration Options

Table 4-1-2 Definition of Registration Options

Opons	Denion
AuthencaonUser	A username to use only for registraon.
Register Extension	When Gateway registers as a SIP user agent to a SIP proxy (provider), calls from this provider connect to this local extension.
From User	A username to identify the gateway to this endpoint.
From Domain	A domain to identify the gateway to this endpoint.
Remote Secret	A password which is only used if the gateway registers to the remote side.
Port	The port number the gateway will connect to at this endpoint.
Quality	Whether or not to check the endpoint's conncon status.
Qualify Frequency	How oen, in seconds, to check the endpoint's conncon status.

Call Settings

Table 4-1-3 Definition of Call Options

Opons	Denion
DTMF Mode	Set default DTMF Mode for sending DTMF. Default: rfc2833.Other opons: 'info', SIP INFO message (applicaon/dtmf-relay);'Inband', Inband audio (require 64kbit codec -alaw, ulaw).
Call Limit	Seng a call-limit will cause calls above the limit not to be accepted.
TrustRemote-Party-ID	Whether or not the Remote-Party-ID header should be trusted.
SendRemote-Party-ID	Whether or not to send the Remote-Party-ID header.
Remote Party IDFormat	How to set the Remote-Party-ID header: from Remote-Party-ID or from P-Asserted-Identity.
Caller IDPresentaon	Whether or not to display Caller ID.

Advanced: Signaling Settings

Table 4-1-4 Definition of Signaling Options

Opons	Denion
Progress Inband	Set default DTMF Mode for sending DTMF. Default: rfc2833.Other opons: 'info', SIP INFO message (applicaon/dtmf-relay);'inband', Inband audio (require 64kbit codec -alaw, ulaw).
Allow Overlap Dialing	Allow Overlap Dialing: Whether or not to allow overlap dialing. Disabled by default.
Append user=phone to URI	Whether or not to add ‘; user=phone’ to URIs that contain a valid phone number.
Add Q.850 Reason Headers	Whether or not to add Reason header and to use it if it is available.
Honor SDP Version	By default, the gateway will honor the session version number in SDP packets and will only modify the SDP session if the version number change. Turn this opon o to force the gateway to ignore the SDP session version number and treat all SDP data as new data. This is required for devices that send non-standard SDP packets (observed with Microso OCS). By default this opon is on.
Allow Transfers	Whether or not to globally enable transfers. Choosing 'no' will disable all transfers (unless enabled in peers or users). Default is enabled.
Allow Promiscuous Redirects	Whether or not to allow 302 or REDIR to non-local SIP address.Note that promiscredir when redirects are made to the local system will cause loops since this gateway is incapable of performing a "hairpin" call.
Max Forwards	Seng for the SIP Max-Forwards header (loop prevenon).
Send TRYING on REGISTER	Send a 100 Trying when the endpoint registers.
Outbound Proxy	A proxy to which the gateway will send all outbound signaling instead of sending signaling directly to endpoints.

Advanced: Timer Settings

Table 4-1-5 Definition of Timer Options

Opons	Denion
Default T1 Timer	This mer is used primarily in INVITE transacons. The default for Timer T1 is 500ms or the measured run-trip me between the gateway and the device if you have qualify=yes for the device.
Call Setup Timer	If a provisional response is not received in this amount of me, the call will auto-congest. Defaults to 64 mes the default T1 mer.
Session Timers	Session-Timers feature operates in the following three modes: originate, Request and run session-mers always; accept, run session-mers only when requested by other UA; refuse, do not run session mers in any case.
Minimum Session Refresh Interval	Minimum session refresh interval in seconds. Default is 90secs.
Maximum Session Refresh Interval	Maximum session refresh interval in seconds. Defaults to 1800secs.
Session Refresher	The session refresher, uac or uas. Defaults to uas.

Media Settings

Table 4-1-6 Definition of Media Settings

Opons	Denion
Media Sengs	Select codec from the drop down list. Codecs should be dierent for each Codec Priority.

Batch SIP Endpoint

If you want add batch Sip accounts, you can congregate this page. Look out: this is only used when "This gateway registers with the endpoint" work mode.

Port	User Name	Password	Hostname or IP Address	Codec Priority
1				G.711 u-law ▼
2				G.711 u-law ▼
3				G.711 u-law ▼
4				G.711 u-law ▼
5				G.711 u-law ▼
6				G.711 u-law ▼
7				G.711 u-law ▼
8				G.711 u-law ▼

Figure 4-2-1 Batch SIP Endpoint

Advanced SIP Settings

Networking

Table 4-3-1 Definition of Networking Options

Opons	Denion
UDP Bind Port	Choose a port on which to listen for UDP trac.
Enable TCP	Enable server for incoming TCP conncon (default is no).
TCP Bind Port	Choose a port on which to listen for TCP trac.
TCP AuthencaonTimeout	The maximum number of seconds a client has to authenticate. If the client does not authenticate before this meout expires, the client will be disconnected.(default value is: 30 seconds).
TCP AuthencaonLimit	The maximum number of unauthenticated sessions that will be allowed to connect at any given me(default is:50).
Enable HostnameLookup	Enable DNS SRV lookups on outbound calls Note: the gateway only uses the rst host in SRV records Disabling DNS SRV lookups disables the ability to place SIP calls based on domain names to some other SIP users on the Internet specifying a port in a SIP peer denion or when dialing outbound calls with suppress SRV lookups for that peer or call.
Enable Internal SIPCall	Whether enable the internal SIP calls or not when you select the registraon opon "Endpoint registers with this gateway".
Internal SIP CallPrex	Specify a prex before round the internal calls.

NAT Settings

Table 4-3-2 Definition of NAT Settings

Opons	Denion
Local Network	Format:192.168.0.0/255.255.0.0 or 172.16.0.0./12. A list of IP address or IP ranges which are located inside a NATed network.This gateway will replace the internal IP address in SIP and SDP messages with the external IP address when a NAT exists between the gateway and other endpoints.
Local Network List	Local IP address list that you added.
Subscribe Network Change Event	Through the use of the test_stun_monitor module, the gateway has the ability to detect when the perceived external network address has changed. When the stun_monitor is installed and congured, chan_sip will renew all outbound registraons when the monitor detects any sort of network change has occurred. By default this opon is enabled, but only takes eect once res_stun_monitor is congured. If res_stun_monitor is enabled and you wish to not generate all outbound registraons on a network change, use the opon below to disable this feature.

Advanced: NAT Settings

Table 4-3-3 Definition of NAT Settings Options

Opons	Denion
Start of RTP Port Range	Start of range of port numbers to be used for RTP.
End of RTP port Range	End of range of port numbers to be used for RTP.
RTP Timeout

Parsing and Compatibility

Table 4-3-4 Instruction of Parsing and Compatibility

Opons	Denion
Strict RFCInterpretaon	Check header tags, character conversion in URIs, and mulline headers for strict SIP compatibility(default is yes)
Send CompactHeaders	Send compact SIP headers
SDP Owner	Allows you to change the username led in the SDP owner string.This led MUST NOT contain spaces.
Disallowed SIPMethods	The external hostname (and oponal TCP port) of the NAT.
Shrink Caller ID	The shrinkcallerid funcon removes '(', '', ')', non-trailing '.', and '-' not in square brackets. For example, the caller id value 555.5555 becomes 5555555 when this opon is enabled. Disabling this opon results in no modicaon of the caller id value, which is necessary when the caller id represents something that must be preserved. By default this opon is on.
MaximumRegistraon Expiry	Maximum allowed me of incoming registraons and subscripons (seconds).
MinimumRegistraon Expiry	Minimum length of registraons/subscripons (default 60).
DefaultRegistraon Expiry	Default length of incoming/outgoing registraon.
RegistraonTimeout	How oen, in seconds, to retry registraon calls. Default 20 seconds.
Number ofRegistraonAempts Enter '0'for unlimited	Number of registraon aempts before we give up. 0 = connue forever, hammering the other server unl it accepts the registraon. Default is 0 tries, connue forever.

Security

Table 4-3-5 Instruction of Security

Opons	Denion
Match AuthUsername	If available, match user entry using the 'username' eld from the authencaon line instead of the 'from' eld.
Realm	Realm for digest authencaon. Realms MUST be globally unique according to RFC 3261. Set this to your host name or domain name.
Use Domain as Realm	Use the domain from the SIP Domains seng as the realm. In this case, the realm will be based on the request 'to' or 'from' header and should match one of the domain. Otherwise, the congured 'realm' value will be used.
Always AuthReject	When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password/hash instead of leng the requester know whether there was a matching user or peer for their request. This reduces the ability of an acker to scan for valid SIP usernames. This opon is set to 'yes' by default.
AuthencateOpons Requests	Enabling this opon will authenticate OPTIONS requests just like INVITE requests are. By default this opon is disabled.
Allow Guest Calling	Allow or reject guest calls (default is yes, to allow). If your gateway is connected to the Internet and you allow guest calls, you want to check which services you oer everyone out there, by enabling them in the default context.

Media

Table 4-3-6 Instruction of Media

Opons	Denion
Premature Media	Some ISDN links send empty media frames before the call is in ringing or progress state. The SIP channel will then send 183 indicang early media which will be empty - thus users get no ring signal. Seng this to "yes" will stop any media before we have call progress (meaning the SIP channel will not send 183 Session Progress for early media). Default is 'yes'. Also make sure that the SIP peer is congured with progressinband=never. In order for 'noanswer' applicaons to work, you need to run the progress() applicaon in the priority before the app.
TOS for SIP Packets	Sets type of service for SIP packets
TOS for RTP Packets	Sets type of service for RTP packets

5. Network, Advanced and Logs

Network

On "Network" page, there are "Network Setngs", "DDNS Sengs", and "Toolkit".

Network Settings

There are three types of LAN port IP, Factory, Stac and DHCP. Factory is the default type, and it is 172.16.99.1. When you Choose LAN IPv4 type is "Factory", this page is not editable.

A reserved IP address to access in case your gateway IP is not available. Remember to set a similar network segment with the following address of your local PC.

LAN IPv4 Interface: eth0 Type: Static MAC: A0:98:05:01:17:39 IPv4 Settings Address: 172.16.161.1 Netmask: 255.255.0.0 Default Gateway: 172.16.0.1 DNS Servers DNS Server 1: 202.96.134.133 DNS Server 2: 202.96.128.166 DNS Server 3: 8.8.8.8 DNS Server 4: Reserved Access IP Enable: ON Reserved Address: 192.168.99.1 Reserved Netmask: 255.255.255.0

Figure 5-1-1 LAN Settings Interface

Table 5-1-1 Definition of Network Settings

Opons	Denion
Interface	The name of network interface.
Type	The method to get IP.Factory: Geng IP address by Slot Number (System → informaon to check slot number).Stac: manually set up your gateway IP.DHCP: automacally get IP from your local LAN.
MAC	Physical address of your network interface.
Address	The IP address of your gateway.
Netmask	The subnet mask of your gateway.
Default Gateway	Default getaway IP address.
Reserved Access IP	A reserved IP address to access in case your gateway IP is not available. Remember to set a similar network segment with the following address of your local PC.
Enable	A switch to enable the reserved IP address or not.ON(enabled), OFF(disabled)
Reserved Address	The reserved IP address for this gateway.
Reserved Netmask	The subnet mask of the reserved IP address.

Basically this info is from your local network service provider, and you can ll in four DNS servers.

DNS Servers
DNS Server 1:	221.179.38.7
DNS Server 2:
DNS Server 3:
DNS Server 4:

Figure 5-1-2 DNS Interface

Table 5-1-2 Definition of DNS Settings

Opons	Denion
DNS Servers	A list of DNS IP address. Basically this info is from your local network service provider.

OpenVPN Settings

You can upload the OpenVPN client conguraon, if success, you can see a VPN virtual network card on SYSTEM status page. About the congure format you can refer to the Noce and Sample conguraon.

OpenVPN Settings OpenVPN ON Upload Configuration New configuration file: 选择文件 未选择文件 File Upload Notice: 1. The format of the upload file should be like this xxxx.tar.gz; 2. The postfix of configuration files should be .conf; 3. The upload file can not include any directory; 4. If still confused please download the sample configuration and refer to it; Sample Configuration Sample Configuration Download Samples

Figure 5-1-3 OpenVPN Interface

VPN Settings VPNType: PPTP VPN ▼ PPTP VPN Settings Server: 0.0.0.0 Account: test Password: **** Domain: Use MPPE: Save

Figure 5-1-4 PPTP VPN Interface

DDNS Settings

You can enable or disable DDNS (dynamic domain name server).

DDNS Settings DDNS: ON Type: inadyn Username: admin Password: admin Your domain: www.internet.site.com

Figure 5-1-5 DDNS Interface

Table 5-1-3 Definition of DDNS Settings

Opons	Denion
DDNS	Enable/Disable DDNS(dynamic domain name server)
Type	Set the type of DDNS server.
Username	Your DDNS account's login name.
Password	Your DDNS account's password.
Your domain	The domain to which your web server will belong.

Toolkit

It is used to check network connectivity. Support Ping command on web GUI.

Interface: LAN google.com Ping google.com Traceroute

Output

ping -l 172.16.179.1 -c 4 google.com PING google.com (173.194.72.101) from 172.16.179.1: 56 data bytes 64 bytes from 173.194.72.101: icmp_seq=1 ttl=46 time=596.6 ms 64 bytes from 173.194.72.101: icmp_seq=3 ttl=46 time=600.5 ms --- google.com ping statistics --- 4 packets transmitted, 2 packets received, 50% packet loss round-trip min/avg/max = 596.6/598.5/600.5 ms Result Successfully ping [ google.com ] .

Figure 5-1-6 Network Connectivity Checking

Advanced

Asterisk API

When you make "Enable" switch to "on", this page is available.

General Enabled: ON Port: 5038 Manager Manager Name: admin Manager secret: admin Deny: 0.0.0/0.0.0 Permit: 172.16.123.123/255.255.0.0&192.168.1.0/2 Rights System: read: ✓ write: ✓ Call: read: ✓ write: ✓ Log: read: ✓ write: ✓ Verbose: read: ✓ write: ✓

Figure 5-2-1 API Interface

Table 5-2-1 Definition of Asterisk API

Opons	Denion
Port	Network port number
Manager Name	Name of the manager without space
Manager secret	Password for the manager. Characters: Allowed characters “-_+.<>&0-9a-zA-Z”. Length:4-32 characters.
Deny	If you want to deny many hosts or networks, use char & as separator.Example: 0.0.0.0/0.0.0.0 or 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0
Permit	If you want to permit many hosts or network, use char & as separator.Example: 0.0.0.0/0.0.0.0 or 192.168.1.0/255.255.255.0&10.0.0.0/255.0.0.0
System	General informaon about the system and ability to run system management commands,such as Shutdown, Restart, and Reload.
Call	Informaon about channels and ability to set informaon in a running channel.
Log	Logging informaon. Read-only. (Dened but not yet used.)
Verbose	Verbose informaon. Read-only. (Dened but not yet used.)
Command	Permission to run CLI commands. Write-only.
Agent	Informaon about queues and agents and ability to add queue members to a queue.
User	Permission to send and receive UserEvent.
Cong	Ability to read and write conguraon les.
DTMF	Receive DTMF events. Read-only.
Reporng	Ability to get informaon about the system.
CDR	Output of cdr, manager, if loaded. Read-only.
Dialplan	Receive NewExten and Varset events. Read-only.
Originate	Permission to originate new calls. Write-only.
All	Select all or deselect all.

Once you set like the above gure, the host 172.16.123.123/255.255.0.0 is allowed to access the gateway API. Please refer to the following gure to access the gateway API by puy. 172.16.123.123 is the gateway's IP, and 5038 is its API port.

172.16.123.123 - PuTTY root@Openvox-Wireless-Gateway:~# telnet 172.16.123.123 5038 Asterisk Call Manager/1.1 action: login username: admin secret: admin Response: Success Message: Authentication accepted Event: FullyBooted Privilege: system,all Status: Fully Booted

Figure 5-2-2 Putty Access

Asterisk CLI

In this page, you are allowed to run Asterisk commands.

Asterisk CLI Command: ? Execute

Output:

! Execute a shell command agi dump html Dumps a list of AGI commands in HTML format agi exec Add AGI command to a channel in Async AGI agi set debug [on|off] Enable/Disable AGI debugging agi show commands [topic] List AGI commands or specific help aoc set debug enable cli debugging of AOC messages cc cancel Kill a CC transaction cc report status Reports CC stats cdr show status Display the CDR status cel show status Display the CEL status channel request hangup Request a hangup on a given channel

Figure 5-2-3 Asterisk Command Interface
Table 5-2-2 Definition of Asterisk API

Opons	Denion
Command	Type your Asterisk CLI commands here to check or debug your gateway.

If you type "help" or "?" and execute it, the page will show you the executable commands.

Asterisk File Editor

On this page, you are allowed to edit and create conguraon les. Click the le to edit.

File Name	File Size
aaa conf	11474
agents conf	2136
alarmreceiver conf	2227
asterisk conf	247
cdr conf	572
cdr custom conf	388
cdr manager conf	59
chan extra conf	283
codecs conf	1655
dns mqr conf	190

Figure 5-2-4 Configuration Files List

Click "New Conguraon File" to create a new conguraon le. Aer eding or creang, please reload Asterisk.

Logs

On the “Log Sengs” page, you should set the related logs on to scan the responding logs page. For example, set “System Logs” on like the following, then you can turn to “System” page for system logs, otherwise, system logs is unavailable. And the same with other log pages.

System Logs System Logs: ON Auto clean: ON maxsize : 1MB

Figure 5-3-1 System Logs Control

[2016/03/30 08:56:05] Restart logfile_monitor (keeper). [2016/03/30 08:58:05] Restart logfile_monitor (keeper). [2016/03/30 09:00:05] Restart logfile_monitor (keeper). [2016/03/30 09:02:06] Restart logfile_monitor (keeper). [2016/03/30 09:04:06] Restart logfile_monitor (keeper). [2016/03/30 09:06:06] Restart logfile_monitor (keeper). [2016/03/30 09:08:06] Restart logfile_monitor (keeper). [2016/03/30 09:10:06] Restart logfile_monitor (keeper). [2016/03/30 09:12:07] Restart logfile_monitor (keeper). [2016/03/30 09:14:07] Restart logfile_monitor (keeper). [2016/03/30 09:16:07] Restart logfile_monitor (keeper). [2016/03/30 09:18:07] Restart logfile_monitor (keeper). [2016/03/30 09:20:07] Restart logfile_monitor (keeper). [2016/03/30 09:22:07] Restart logfile_monitor (keeper). [2016/03/30 09:24:08] Restart logfile_monitor (keeper). [2016/03/30 09:26:08] Restart logfile_monitor (keeper). [2016/03/30 09:28:08] Restart logfile_monitor (keeper). [2016/03/30 09:30:08] Restart logfile_monitor (keeper). [2016/03/30 09:32:08] Restart logfile_monitor (keeper). [2016/03/30 09:34:09] Restart logfile_monitor (keeper). [2016/03/30 09:36:09] Restart logfile_monitor (keeper). [2016/03/30 09:38:09] Restart logfile_monitor (keeper). [2016/03/30 09:39:04] System Update [2016/03/30 09:39:15] Power off [1970/01/01 07:01:24] Power on [1970/01/01 07:01:47] Restore configuration files [1970/01/01 07:01:54] Power off [1970/01/01 00:00:13] Auto restore configuration files [1970/01/01 07:01:19] Power on Refresh Rate: Off Refresh Clean Up

Figure 5-3-2 System Logs Output

Noce: The same to Asterisk Logs and SIP Logs.

Table 5-3-1 Definition of LOG

Opons	Denion
System Logs	Whether enable or disable system log.
Auto clean(System Logs)	switch on :when the size of log le reaches the max size,the system will cut a half of the le. New logs will be retained.switch o :logs will remain, and the le size will increase gradually.default on, max size=1MB.
Verbose	Asterisk console verbose message switch.
Noce	Asterisk console noce message switch.
Warning	Asterisk console warning message switch.
Debug	Asterisk console debug message switch.
Error	Asterisk console error message switch.
DTMF	Asterisk console DTMF info switch.
Auto clean:(asterisk logs)	switch on:when the size of log le reaches the max size,the system will cut a half of the le. New logs will be retained.switch o:logs will remain, and the le size will increase gradually.default on, max size=100KB.
SIP Logs:	Whether enable or disable SIP log.
Auto clean:(SIP logs)	switch on:when the size of log le reaches the max size,the system will cut a half of the le. New logs will be retained.switch o:logs will remain, and the le size will increase gradually.default on, default size=100KB.

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