Shang FXO/FXS Gateway analog telephone adapter, IAD, From 1 port to 32 port

1 Overview

The VoIP Residential Gateway is an external standalone device, which can provide a cost effective communication through the Internet. The Voip residential gateways are available in one and two channel models. Both models can connect directly to phones, fax machines, pbx, and internet without extra equipment and setup.
 

Support SIP or H.323

Cost-Effective Edge Device

Shang FXO/FXS Gateway support 4/8/16/24/32 channels of VoIP with 10/100 Base T interfaces in a small box. It's the most cost-effective edge device for the residential users.

Two 10/100 Ethernet Connection

2 10/100 Base T interfaces are supported in Shang FXO/FXS Gateway. In the typical application, one 10/100 Base T port is used as the WAN to connect to the broadband access device like ADSL modem or cable modem, and the other one is used as LAN to connect to local data network.

Simple Configuration and Setup

Shang FXO/FXS Gateway provide the web interface and CLI for configuration. The GUI in the web interface is designed for easy configuration and setup. STUN function is supported in order to provide NAT traversal function.

Remotely Manageable and Upgradeable

Users can manage Shang FXO/FXS Gateway remotely either via telnet or the web interface. New software can be upgraded from the remote site via TFTP protocol. This will reduce operators' maintenance cost.

PSTN Backup Line

One PSTN backup line is supported in Shang FXO/FXS Gateway. If the power is failed or the network connection is broken, the voice line will automatically switch to the PSTN backup line. In normal case, users can also dial the prefix number to force switch to PSTN backup line.

• work with Asterisk, SER and GNUGK.
• Call waiting, call transfer, call hold, caller ID, call forward. Various call feature enhanced.
• Provide CDMA or GSM access.
• Free Callshop software. No need to use ITSP's Brand.

2 Specifications

Voice Features

Ø Number of ports: 1 or 2 ports available

Ø FXS ports: RJ-11

Ø IP telephony protocol: MGCP, SIP and H.323

Ø Voice compression: G.711, G.723, G.729a

Ø Bandwidth management: Voice Activity Detection (VAD), Comfort Noise Generation (CNG)

Ø Echo cancellation: G.168

Ø Signaling: Loop start and polarity reversal

Ø Ringing Equivalent Number: 4

Ø Power outage backup: PSTN loop through backup for power outage

Ø Fax/modem support: Auto-switch to G.711 for Fax transmission

Ø Protocol: RTP, RTCP

Ø Loop current: 20 mA

Ø DTMF detection

Ø Call progress tone generation

Data Features

Ø Number of ports: Two Ethernet ports

Ø Interface: 10/100 Base T auto-negotiation

Ø DHCP

Ø PPPoE

Ø NAT

Ø STUN (Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NAT))

Management

Ø Installation: Plug and play with default setting

Ø Configuration: DHCP option to set up system parameter

Ø IP address: Static, DHCP and PPPoE

Ø Software download: TFTP enable for remote firmware upgrade

Ø TFTP firmware upgrade

Ø Network diagnostic: local and remote diagnostic LED

Ø Power, WAN, ENET, VOIP, LINE

Standard Compliance

Ø EMI: FCC part 15 class B

Ø Safety: UL 1950, EN60950

3 How to make your voip network more stable?

For voip base on the ip network, without a Qos assurence. How to make a voip service more like a telecom carrier grade service? There are many ways to do it.

• PSTN life line. When voip gateway is out of power, all phone that connected to that gateway are directed to the pstn life line. It seem to be better, but when the network is broken, or voip service is out of serivce, the gateway can't do any thing.
• Use Route selector between voip gateway and phone. Every route selector unit have 3 port, one for phone, one for voip gateway and one for PSTN line. When accident happend, you can change to the PSTN route manually, or easily dial phone with a prefix number such as 9. It is 1:1 PSTN backup and it is also a PSTN and VOIP integrated solution. Both VOIP and PSTN can work like normal way. Route selector work to distinguish them by phone number prefix.
• MultiPath technology voip gateway . Some foumous gateway like Quintum and Shang FXO/FXS Gateway use this technology to ensure the enterprise level voip service quality. Base on the Route selector, it can detect voip network quality automatically. When voip network is out of service, it change to PSTN automatically.

Shang FXO/FXS Gateway fxo gateway provides more feature to work with asterisk.

1.play asterisk ivr with no interuption.

when the Shang FXO/FXS Gateway received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

2. pbx voip/pstn inteleged route.

when you make pbx connect to voip/asterisk, how to make voip more stable. Shang FXO/FXS Gateway could detect the voip quanlity, when voip line failed, it change to pstn line automaticly.

3. pstn caller number transfer.

When pstn call in, the Shang FXO/FXS Gateway start voip call to the asterisk using pstn caller number instead of gateway number.

4. multy region pstn singal support.

By using MindSpeed technology, it integrated many region's pstn singal.

Why we choose a voip fxo gateway while not a asterisk card.

• voice process is a realtime task. PC operate system is not a realtime one. Voip gateway use its own dsp to do voice process while asterisk card use PC CPU to do this. Just like the DVD decode, on heavy task, voip gateway hardware will do better. So we sugguest you to use voip gateway on more than 4 phone line system.

• Asterisk PC+ voip gateway model, it is easy to expand to over 100 user. In this scale, you can not plug so many card into one PC.

• There are many analog voip gateway producer, the price is cheap, especially for Shang FXO/FXS Gateway fxs.

How Shang FXO/FXS Gateway work with Asterisk feature.

For FXS voip gateway, interoperate with Asterisk is easy. There are two requirements: one is sip interoperability. The other is DTMF transfer model.

If a voip gateway can make call with asterisk, that could to say sip interoperability is ok. If the auto attendant service is ok, that is mean DTMF transfer is ok.. The other things will be no problem.

Howerver there are some limits in Asterisk, especially on transcoding. If we use g.711, all is ok. but we are normal use g.723 or g.729. when we want to use conference service. Asterisk need to change codec to G.711.

The other things you must be careful. When a call setup successfully, there need call original voip gateway, and also the call termination voip gateway. The interoperation of such two type gateway is also important.

For FXO voip gateway, there is a lot of works. Such as the iver play feature, it need to special configed for asterisk.

How Shang FXO/FXS Gateway backup voip with the PSTN line

Shang FXO/FXS Gateway 4s4o owned 4 fxo and 4 fxs ports. When you work in backup mode, 4 fxs ports connect to pbx trunk line, 4 fxo ports connect to CO line. User start a voip call from fxs port, When a voip call failed, Shang FXO/FXS Gateway will automatically dial on the pstn line through fxo port. The end user use the gateway like it was on voip line. So the voip call quanlity is protected with pstn backup line.

The voip change to pstn condition is: when voip service is unavailable or the power is loose.

It is very difference than lifeline. Lifeline is a n to 1 protection. This is 1:1 protection. Lifeline work only when the power is loose, it has no use when the network is down.

Document download

Software and Firmware download