All of ALLWIN's product lines can regsiter up to 4 SIP Proxy or GKs simultaneously.



As below diagram, the area code number of 020 will make redundancy according to the orders of 2>3>1 by 3 SIP server.

All of ALLWIN's product lines can support H.323 and SIP protocol simultaneously.





As below application diagram, users can choose the different protocol according to the different call out number making communication via peer to peer, SIP Proxy or H.323 Gatekeeper.

As the CODEC is processed by the specific DSP for all of ALLWIN's product lines, therefore the build-in 32 bit network processor has enough ability to process the internet router functions besides the VoIP functions, these powerful functions include:

WAN connection
Besides support DHCP client. Static IP, PPPoE for accessing network, but also support PPPTP, L2TP, MAC Cloning and so on.



NAT
Besides support basic NAT and DHCP Server, but also support Virtual Server, Special Application mapping, Port mapping, ALG, DMZ and Static Routing table and so on.



Firewall
Provide many kinds of Firewall's configurations to prevent from the Internet malicious attack, and may aim at different Port to provide Client filtering, URL Filtering, MAC control and so on for filtering control.



UPNP
Support the UPNP protocol and can display the use of Port mapping.



DDNS
Build-in Dynamic Domain Name Service supporting various services, such as dyndns.org. and so on that can be applied to no fixed Static IP environment.

Generally, the construction of software from bottom tier to top tier is RTOS (Real Time Operation System)>SIP protocol>application>WEB UI for other VoIP devices' suppliers and their every tier of software is compiled by C language with API.

Therefore, the system would shut down if any tier of software is on trouble by this kind of traditional design; and any modification need to be re-compiled, then link for use. If not consider the inside process from bottom to top tiers of API's for the modification of any tier of software that it will cause the system shut down as well. Hence, the traditional design by C language and API oriented will be difficult to meet the application of customized VoIP devices.

As ALLWIN understands the design way and its limitation, ALLWIN's R&D team spent around USD0.35 million and two years to design the newest construction of SW and HW after getting the R&D project's sponsor of "Ministry of Economic Affairs, R.O.C.".

The project of OneInXML uses specific XML parser to process the high lever's application of top tier of SW, such as CODEC procedure and Web UI.
Therefore, ALLWIN can meet and provide OEM/ODM service due to the separate process of application's tier and protocol's tier to assure the stability and easy to extend the application of devices.

ALLWIN designs all of product lines by many years of rich design experiences, no matter on the design of hardware, communication protocol, or Web UI; and not only to accept the "Me too" products of chip vendor's turn key.
Also, ALLWIN provides the standard Web UI on IP phones, various ports of Gateways, E1 PRI trunk; therefore it's easier for clients to use all of ALLWIN's product lines after users are familiar with one of any products' Web UI.

Nowadays, there are many VoIP devices, specially for low-end products of 1 to 2 ports gateways, which use special-purpose SoC (System On Chip) applied by CPU with DSP on the same chip or not use DSP to execute voice compression, echo cancellation and so on by main CPU's processing ability only.

The soft CODEC is used instead of DSP that this kind of product's price is lower as to reduce the use of components, but the performance is limited as the cost down of hardware, and as a result that the most end products are produced by Turn Key's program provided by chips' suppliers that just can provide simple function and become very difficult to produce customized products or debug program.

ALLWIN's entire OneInGate VoIP series used specific powerful performance of 32-bit network processor and voice signal processes of DSP according to different deployments' devices no matter for 1 port of TA, IP phone, E1 trunk gateway.

Therefore the performance, the sound quality and the stability are much more competitive with the other low cost of VoIPs' devices.

Port Status display the current and last time of VoIP call status & result, and can be renewed the status by the "Reload" function, but all the record will be deleted if reboot the device.

Port : Display which port is on connection status, e.g. 1 or 2.

Display name: Display the remote party name of this VoIP call.

Status: Current status of this port.
Idle: Standby for making a phone call.
Signal: Waiting for DTMF press or VoIP protocol connecting.
In: There is a phone call made from phone port and call out to Network by VoIP.
Out: There is a phone call made from Network VoIP and pick up by phone set.

Connected IP: The remote party's IP of this VoIP call.

Caller ID: Caller ID received from telephone line port.

Start time: Date & Time of this VoIP begins to make a call on this port, the time is calculated from the phone rings.

End time: Date & Time of making last VoIP call on this port.

Talking Sec: Total talking seconds of last VoIP call on this port.

Dialed number:
One the VoIP call out (line status display"In"). This will display the real dial out number for VoIP call.
On the VoIP call in (line status display"out"). This will display the number will dial out to phone line.

Release by: This will display the terminated reason of this call.

(134)onHangup: Represents the local party is for calling out, and as P2P or the call can be transferred via system only; the call cannot be gone through for speaking and hung up by local or remote party.

(212)H323Release: Represents the local party is for calling in (receiver), the call cannot be gone through for speaking and hung up by local or remote party.

(107)inDigitLess: Represents the call number is not enough.

(142)VoipRelease: Represents the local party is for calling out, the call can be gone through for speaking and hung up by the receiver.

(236)onHangup: Represents the local party is for calling in (receiver), the call can be gone through for speaking and hung up by the local party.

(106)noAreaCode: Represents no any area code to correspond with the call.

(132)H323Release: Represents the local party is for calling out, and the call can be transferred by the system; but the system responses the call out number is wrong and hang it up.

(105)inDigitOnHangup: Represents the call is hung up before completing the dial number.

(123)AdmissionFailure: Represents the local party is for calling out, and the call can be transferred by the system; but the system responses not to find the corresponding call out number and hang the call up.

(103)inDigitOnHangup: Represents the call is no any response on remote party.

(146)onHangup: Represents the local party is for calling out, and as P2P or the call can be transferred via system only; the call cannot be gone through for speaking and hung up by local party.

(232)VoipRelease: Represents the local party is for calling in (receiver), the call can be gone through for speaking and hung up by remote party.

(222)VoipRelease: Represents the local party is for calling in (receiver), the call can not be gone through for speaking and hung up by remote party.

(102)inDigitOnMaxTime: Represents the local party is for calling out and not to complete the dialing number before the standard Max Time.

(121)VoipFaliure: Represents the route of VoIP Call out can not be called out. (If the configured route is ineffective or wrong).

When this gateway has been used for outbound call, it can be enabled to check the remote caller gateway's IP to decide to accept or refuse the call. If define the IP range here and enable the "Auth" option on the / VoIP Setup / Routing Setup / VoIP Call in /, only the IP in range will be allowed to call out by this gateway.

Define each item on Forwarding below.

Name: Specify a profile name. Please use UNDERLINE to instead of SPACE due to the HTTP protocol limitation.

Always: Always redirect forward to this IP(or URL) / Phone number, all incoming call will be forwarded to the assigned IP here.

On Busy: Redirect forward to this IP(or URL)/Phone number when busy, all incoming VoIP call will be forwarded to the assigned IP here when this line is busy.

No Answer:Redirect forward to this IP(or URL)/Phone number when no answer over the time No Answer Sec, an incoming VoIP call will be forwarded to the assigned IP here when ring time over the defined No Answer Sec.
Below is the Configuration method for users want to forward the call to the IP of 172.16.39.2 for calling to 66390001 directly, please also revise the call number to 63986399.

Configuration on VoIP Call in below:


Forwarding:

The prefix number will be added on the users' press number, e.g. if users press the number is "567" and the prefix field is set to "123", then the actual call out number will be "123567"; if the prefix field is set to "@123", then the actual call out number will be "567123".