It could also be used for the lines from PSTN which is connected to FXO gateway ( such as Shang308)

(1)      Have 1 phone connected to the Xia304's fxs port. use voltage meter to test the line's voltage. 
(2)      measure the line's voltage.
(3)      make a phone call, the callee doesn't pick up the phone
(4)      measure the caller line's voltage.
(5)      callee pick up the phone.
(6)      measure the caller line's voltage and record the voltage difference.

Result :

(1)      the voltage is between 48V—51V.
(2)      After the caller pick up phone, voltage change to 7V.
(3)      Callee ringing. The Caller will hear the ringback tone. Voltage keep 7v.
(4)      Callee pick up the phone, if there are polarity reversal, Voltage will change to -7v while not keep 7v.
(5)      when the call is going on,  the voltage will keep no change.
(6)      when the callee hang up the phone, the voltage will turn to 48v.

Those voltage parameters will be different according to different country's standard. But for the polarity reversal, it is very easy to find if it work.

• 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.

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.

Shang FXO/FXS Gateway fxo gateway provides more features 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.

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.
1. 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.
The Xia 304 has 4 lines to provide voip service and a lifeline for PSTN. When power is lost,  the lifeline connect the last port telephone with PSTN.

2. 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.

3. 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.

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.

 

There are many countries which are facing the voip block problem.

Generally, there are several means to block the voip business:
1. block the IPs and domain names of servers which have the main Voip stream.
2. shield the main Voip server ports, for  example the 5060 port for SIP protocol, or the 1720 port for H.232 protocol and so on.
3. monitor the Voip stream, if detected, send out the fake IP packages to disturb the Voip communication.
4. a lot of Middle-East countries like Dubai shut up most of the ports , only letting free limited ports such as 80 port.

There are 3 ways to resolve this problem:
1. use VPN. The flow picture is as follows:

     The first thing is you need to clear which port you are connecting.
     You could dial ***# and hear the ip of "wan" port. and set your pc to the same ip range and get it connected.
     Another way is try to ping the device's "Lan" port with the following ip : 192.168.0.1, netmask 255.255.255.0; 192.168.0.2,netmask 255.255.255.0; 10.10.0.1, 255.255.0.0. If you are using the PPP.O.E or DHCP, please stop the Wan and try to use Lan only.

You can change the configuration of device from supporting SIP to supporting H232, and the procedure is quite easy.
   
1、Users can access WEB management through WAN interface. The default IP address for WAN interface is 192.168.0.2.

Firstly, to make sure the WAN interface is connected, try Ping 192.168.0.2 from a PC. If it’sconnected, start IE browser, enter 192.168.0.2 in address link，press Enter to access WEB, Log on window. If you forgot what is IP address, please connected one phone with one of phone ports, and then press ***# on phone to report the address of WAN port.

Users can use LAN interface to access WEB management as well. The procedure is exactly same as using WAN interface without enable NAT on LAN configuration. If NAT is enabled on LAN port configuration, the default IP address for LAN interface is 10.10.0.1 or 192.168.0.1. Generally, IP address of LAN interfaceis 10.10.0.1, but it changed to 192.168.0.1 when the WAN interface is 10.10.0.0 segment On the assumption that LAN interface is 10.10.0.1, access XIA30 as following: At first, choose one IP address from 10.10.0.0 segment for your PC, such as 10.10.0.100. Secondly, try ping 10.10.0.1 to make sure the connection to LAN port is good. If the connection is good, start IE browser, enter 10.10.0.1 in address link，press Enter to access into WEB Log on window.

For example, we access the equipment with WAN port, If IP address is 192.168.0.2, The log on window is as following:

When we do configuration on WEB,user/password must be provided to confirm.System has two default users: super user ‘root’ , password ‘root’ and management user ‘admin’, password’admin’。

When successfully logged in, the WEB management interface is as following: