 |
| Modules | |
| Search | |
| Login | |
|
Don't have an account yet? You can create one. As a registered user you have some advantages like theme manager, comments configuration and post comments with your name. | |
| |
Welcome to T.C.S Knowledge Base
This is a knowledge base website built to provide you the information of our products and services.
Our products relate to Ip telephone terminals and VoIp gateways. You can seek our the further information about our products at website:www.telecomchinasourcing.com.
We'll try our best to offer you the the most qualified products!
|
|
 | A kids school that use trixbox and shang332 in Canada case study |
 A kids school that use trixbox and shang332 in Canada case study This Kids school has 3 school in 3 cities. They want to connect them over the internet to make free internal call. And they want apply more phone number to provide better service.First location - about 30 extensions, second location - about 20 extensions and third location - about 7 extensions. What devices they used They installed a trixbox server in the head office, use a shang332 28fxs4fxo for their 30 extensions. Use a Jin302 phone for the reception. They applied some DID number from didww.com, and connect the old pstn number to the shang332’s fxo port. Connect all those device into the internet by an ADSL with static public IP. In the site 2, there are an shang332 12 fxs4fxo for extension and pstn , and an Jin302 phone for reception. In the site 3, because there are only 7 extension, so they all use ip phone Jin301 and Jin302. What function they get By those DID from didww.com, they have many phone number at low cost about $5 for a number per month. The trixbox support many sip lines capability. When their customers call to those DID, they will be accessed to the trixbox’s welcome message. After that they could dial extension number to reach yips’ office in 3 places. If the caller select 0 for the reception, the reception’s Jin302 ip telephone would ring. After the reception pick up the phone, he/she can transfer to the other extension. For those employees, they can make voip call and also pstn calls. This gateway provides pstn failover. Normally, they can call by trixbox, when the sip service provider is down, they could choose different router to make pstn call. Even when the power is lose, there are still 4 lines are live for pstn calls. |
|
|
Posted by luis on Thursday, April 24 @ 02:32:03 EDT (820 reads)
(comments? | Score: 0)
|
|
|
| SHANG332 For Termination |
SHANG332 For Termination
1. overview
The SHANG332 is because of having the FXO, so it can be used to connect VOIP with the PSTN .The SHANG33232 can provide the varieties, such as 8 FXOs,16 FXOs,24 FXOs and 32 FXO...etc. It can be used for the access method of the calling card service so that the calling card server connects into the PSTN.
2. configuration example
the IP telephone company, when they are lack of the digital E1/T1 connection, can use analogue line to provide termination to PSTN. For example the outside line has 8 lines, number is 82901131.
1、Configure IP address and VOIP server information
The gateway can work under 2 kinds of modes: register and the Peer to Peer. For adopt which kind of method, this depends on the request of the service provider. If use registers mode,it need to set"the Enable Registration"and" Enable Proxy" to yse in the diagram, filling the Registrar and Porxy Address up. If the adoption Peer to Peer method it needs to set"the Enable Registration"and" Enable Proxy" to no.
2、FXO Gateway VOIP account setting
At"port configuration" page, set the number of the FXO to 9. The remote voip equipments will route calls that prefix 9 to this gateway. For example termination in India, the country code is 94, the distant equipments can send 9-94-1234567890, 941234567890 is the real number that send to the PSTN. if using registers model, after configuration complete, this page will show the appearance of the port. If it register success, then it show registered.
3. Phone Number Plan and Route Setting
Still according to that example, the FXO gateway receives number 9-94-1234567890, The number on the PSTN should be 94-1234567890.So the prefix 9 can be deleted here.At "the number transformation rule" page, set it to delete a prefix 9.
On the " regeneration number change " page, set every port applies rule 1. So from FXO , all destination number will be deleted a prefix 9 before they are been sent.
In the Routing Direction Table, add all prefix that will go to the PSTN.
This is the end of all settings for the Termination business.
4. Notes 1. In order not to appear the circuit hangup, you need to configure local tone parameters for the disconnect suppervision. For more details please refer to the user manual. 2. For answer supervision and correct billing , please make sure your analogue line provides polarity reversal signal. |
|
|
Posted by luis on Tuesday, April 08 @ 00:14:23 EDT (493 reads)
(comments? | Score: 0)
|
|
|
| Make use of the SHANG332 as the remote telephone extension |
Make use of the SHANG332 as the remote telephone extension
Overview
Make use of the VOIP technique, one of the biggest advantages is that changed the physics distance to voice communication, having no concept of long distance, and it is also easy for deployment. In some enterprises, there are a lot of branch organizations, having a lot of employeeses working outside. In this kind of case, deploy the SHANG332 will solve the enterprise internal communication, save the cost of long distance call very much. Use the telephone from outside like they are in the local office. In contactl center, this form can forward the agents to any place if there are any internet connections.
Remote telephone extension Before reform, each branch organization contact by llong-distance telephone call. After the reformation, they can talk through the IP network. It is free and easy. They call to Other’s extensions by directly input the extension number. This kind of application is particularly useful in the long distance call center. Transform method Increase two customer lines under the PBX of the headquarters, number are 803 and 804, directly connect to a FXO gateway; place another equipments in the branch organization, install the number to 803; if the scale is bigger, you could consider to use the softswitch or IPPBX. There are a few keys: ü What to use is the FXO gateway in the PBX side, it need to transparent deliver the signal and voice. After get the call sign from the distant VOIP equipments, it can automatically build up with the PBX connection. After it receives the PBX call sign, it can automatically call the VOIP equipments of the distant place. ü This FXO gateway must set to the right signal with PBX. Especially for the busy tone detection. It is useful for not hanged up on phone lines. Let’s take SHANG332 as an example, explaining to set the device: 1、 the PBX side FXO gateway setup. a) The equipments’ IP address and SIP information In the “quick setup” page, fill the network parameter and VOIP parameter. Suppose we use the method of Peer to Peer. Notice the Enable Registration and the Enable Proxy constitution items are to No. b) Configure for accept the distant VOIP call In the page of “port configuration”, set the FXO gateway’s 1st and 2th port phone number of voip to 9803 and 9804, this 2 ports will link the PBX line number 803 and 804. c) Call to distant voip gateway setting In the “FXO Advanced Configuration” page set working mode to “auto mode”. When gateway is in auto mode, call from PBX to FXO gateway, this fxo gateway will call the FWD number automatically, so that it will connect the distant voip device. Click the Display Auto FWD NBR button to set FWD Number. Set 1st and 2th port FXO number is 803 and 804. It is correspond to PBX extension 803and804. In the “Route direction table”, set all number that start with 8 forward to voip. Set Peer to Peer Calling Route Table, set all number that start with 8 forward to 192.168.1.21. 2、 Remote FXS gateway setting The equipments’ IP address and SIP information  Set phone number At the Advance=> SPS set each ports’ hotline number. This hotline number is the correspond FXO Gateway port’s voip number. When distant caller pick up the phone, the distant FXS Gateway will connect to FXO Gateway automatically. It is need point 9xxx number to voip too. After the reform project complete, when call from pbx to remote extension, calls reach PBX’ 803, FXO gateway would start call to 803(the FWD number) numbers; at the branch organization, when caller pick up the handset, (hotline number is 9803, 803 is the extensionses of headquarters), the distant voip gateway will directly connect 803 FXO gateway in the headquarters. after hearring the dial tone, then dial the destination number as you at local extension.
|
|
|
Posted by luis on Tuesday, April 08 @ 00:08:49 EDT (726 reads)
(comments? | Score: 1)
|
|
|
| VoIP Telephone Survivability Design |
VoIP Telephone Survivability Design Enterprises are constantly deploying IP telephony. Once an enterprise makes the decision to incorporate a VoIP network deployment, a critical issue to consider is the survivability and high availability of external and internal company telephony communication system. This issue becomes more crucial due to the IP network’s heavy task and the compatible to legacy PSTN network. When power was lost or internet was disconnected, every IT managers will face a big challenge for the telephone system’s survivability. Siptang VoIP/PSTN intelligent routing technology answers those challenges. In regular network situations, this technology will keep work as a sip terminal. It keep detect the network connection, when the sip server is out of work, it will take up the backup mission. When the internet is broken or the sip server is down, it can detect them out. Even in the worst case, the power is lost, the PSTN intelligent routing tech will forward all out boundary calls to PSTN lines. PSTN lines will not be affected by the terminal’s power issues. How to make the system to be compatible to the legacy PSTN network? The PSTN intelligent routing technology will take over the old phone number routing policy and redistribute those calls under whole network system’s requirement. It is very important to let the end users don’t feel any change to the number dial style. The following diagram shows when VoIP is broken, the telephone system is still alive for all calls are forwarded to the PSTN line. SHANG308 is the gateway which has intelligent routing technology. It connects the PSTN and VoIP network and can route the traffic automatically. SHANG308 is an ideal for corporate business as it facilitates a gradually and risk-free transition to IP telephony. Enterprises can also benefit from a variety of valuable applications such as PBX extension, remote office connectivity, long distance consolidation and call centers. |
|
|
Posted by luis on Tuesday, April 08 @ 00:01:13 EDT (904 reads)
(comments? | Score: 0)
|
|
|
| Call Shop Solution |
Call Shop Solution IntroductionVoIP CallShops solutions enable people to make long distance and international phone calls at considerably lower prices as compared to the PSTN. It is a lucrative business because of the low initial business investments and the stable increase in a potential customer base. SipTang introduces a new line of low-cost, functionality rich solutions which will allow entrepreneurs' quick entry and competitive stay in the rapidly growing VoIP industry. These scalable solutions are easy to set up and operate with only a minimal investment. This document will provide users information on how to use SipTang products to begin CallShop business. For more information and questions, please visit our website at www.siptang.com. l New entrepreneurs who would like to get into the highly promising industry of Voice over IP without hefty investment costs and complicated start-up and maintenance processes. l Existing Internet Café owners looking to expand their customer base by using their current infrastructure to offer additional services. l Traditional PSTN CallShop owners wanting to reduce costs and boost profits by simply 'switching' or upgrading to VoIP-based systems. l Carriers who would like to increase their wholesale traffic and enlist more customers by offering hosted CallShop services. l Prepaid: u Customer visits the CallShop. u Customer prepays the operator for the call. u The operator activates a phone booth for the customer using the CallShop Billing Software. u The customer goes to the phone booth and dials the destination number. u CallShop Billing Software records the call details and the corresponding call charges for future reporting needs. l Postpaid: u Customer visits the CallShop. u Customer chooses a vacant phone booth and dials the destination number. u CallShop Billing Software keeps track of each phone booth's call details and the corresponding call charges for invoicing and future reporting needs. u When the customer completes his/her calls, the operator generates an invoice for the customer's calls. 1. Stand-alone CallShop: You are the Owner of a Stand-alone CallShop u A customer places a call from your CallShop by dialing a destination number. u The CallShop Billing Software running on the operator's PC at your CallShop starts to record the CDR (Call Detail Record) for the call. u Your Gateway sends the call over the Internet to the carrier that was configured in the software for the calls destination. u The carrier routes the call to the destination. u When the call terminiates, the operator's CallShop Billing Software retains the complete CDR for that call. It can be used to invoice the customer and any business analysis reporting needs. What will you need: u Analog VoIP Gateway(i.e, Xia302, Xia304, Xia308) u A PC with Windows XP/2000 u Internet Access  2. Hosted CallShop: You are the Owner of a Hosted CallShop u A customer places a call from your CallShop by dialing a destination number. u Your Gateway sends the call over the Internet to the carrier hosting your CallShop service. u This carrier routes the call to the destination and records the CDR for the call in their system. u When the call terminates, your CallShop's operator logs onto the carrier's website to retrieve the CDR and invoicing information for that call. u You may also log into the carrier's website at any time to access other features such as to define/modify service rates and generate reports. What will you need: u Analog VoIP Gateway(i.e, Xia302, Xia304, Xia308) u A PC with Windows XP/2000 u Internet Access  3. Advanced CallShop: You are the owner of a carrier hosting CallShop services. u You have multiple clients hosting CallShop services from you. Each of them has their own Gateway configured to yours. u A call is placed at one of your client's CallShops. u The call is routed to your Gatekeeper/Gateway and your SoftSwitch Billing System starts to record the call CDR. u Your Gateway then routes the call to the destination. u When the call terminates, the CallShop client logs into your system through a website to retrieve the CDR and invoice information for that call. u Youl bill your CallShop clients using invoices generated by your SoftSwitch Billing System. u You may also use the software for other functions such as to define/modify rate and generate reports. What will you need: u Analog VoIP Gateway(i.e, Xia302, Xia304, Xia308) u SoftSwitch System based on Linux server u Internet Access u Your Own Carrier
|
|
|
Posted by luis on Monday, April 07 @ 23:57:01 EDT (1000 reads)
(comments? | Score: 5)
|
|
|
| The SHANG332 used as the application of the Calling Card |
1. overview
The SHANG332 is because of having the FXO, so it can be used to connect VOIP with the PSTN .The SHANG332 can provide the varieties, such as 8 FXOs,16 FXOs,24 FXOs and 32 FXO...etc. It can be used for the termination service so that the voip calls connects into the PSTN.
2. configuration example
When the Calling Card carries completed the Calling Card server, they can make use of the FXO gateway at small scaled, with low cost, vivid, for general use etc. broadly develop the access number, but don't need the specialized circuit and PSTN Operators’s support. For example deploy each city with a FXO gateway, each business agent can have the independence FXO gateway and access numbers.
a) configure the IP address and the Server parameter of Calling Card.
At"quickly install" page, install the IP with the soft commutation address.
The gateway can work under 2 kinds of modes: register and the Peer to Peer. For adopt which kind of method, this depends on the request of the service provider. If use registers mode,it need to set"the Enable Registration"and" Enable Proxy" to yse in the diagram, filling the Registrar and Porxy Address up. If the adoption Peer to Peer method it needs to set"the Enable Registration"and" Enable Proxy" to no.
b) FXO Gateway VOIP account setting
At"port configuration" page, set the number of the FXO to 123.123 is the SIP account of the Calling Card system. if using registers model, after configuration complete, this page will show the appearance of the port. If it register success, then it show registered.
c) The system of Calling Card connection setting
The Service Provider of the Calling Card will provide a VOIP access umber, for example 5555. the VOIP Gateway can connect into to the system of Calling Card by this number. You will hear the Welcome message. It will ask you to input account and password.
Need to establish the Auto Mode in the FXO Advanced Confiuration page first. At this option, when calls arrive the FXO Gateway from the PSTN, it will automatically call a FWD Number.
Then click the Display Auto FWD NBR buttone, set the FWD Number. At"Routing direction" page, add a new line, the number 55xx will be routed to VOIP. Set the 2th time dialing. For input for the Pin code accurately, please confirm the RFC2833 set to Yes d) Peer to Peer Setting If using Peer to Peer method connects to the Calling Card Server, you need to install the Peer to Peer Calling Route Table. The destination IP is the server address of the Calling Card. |
|
|
Posted by luis on Monday, April 07 @ 23:51:28 EDT (895 reads)
(comments? | Score: 0)
|
|
|
| SHANG332 for Company Auto Reception |
SHANG332 for Company Auto Reception
1. General For SHANG332 has the IVR( interactive voice response) and also has the extension line FXS and the trunk line FXO. It can provide basic PBX function for small company. It will be easier to change from old PBX to VOIP’s new feature and cheaper price. Local auto reception For some basic requirement in small office, it is only need this device to take place the old PBX. It has the IVR function. When calls are in, it connected automatically, play welcome message to the caller. the caller could select to call the extension or transfer to the reception person. It is used as same as what we did on the PBX. But because it has voip interface, call could be over VOIP network. 2. Upgrade method: As the up figure showed, there is a 4FXO12FXS device. It’s FXS port use 8001, 8002, 8003, 8004 registered to server 192.216.224.85. Configuration details: The equipments’ IP address and SIP information. In the “quick setup” page, fill the network parameter and VOIP parameter. 2、Calls from pstn to voip by FXO ports setting In the “FXO Advanced Configuration” page set working mode to “normal”. It can play IVR or just a dial tone. For a small company, this IVR is the welcome message. At"port configuration" page, set the phone number 8001、8002、8003、8004. In the Routing Direction Table, add 8xxx to ip. In the Peer to Peer Route Table, add 8xxx to local ip. Those setting will make the call from fxo can reach extensions 8xxx by second time dialing. 3 long distance call setting In the Routing Direction Table, add 0 prefixed number to ip. This will add those calls to softswitch. 4、VOIP call from FXO to PSTN At"port configuration" page, set the FXO phone number 9. In the Number Change Rule Table , add a rule to delete prefix 9. In the regeneration Number Change table, aplly those rule to each port. In the Routing Direction Table, add 9 prefixed number to PSTN. This will add those calls to PSTN. After that, end user can dial 9 + phone number to PSTN Notes: For record welcome message please contact us. |
|
|
Posted by luis on Monday, April 07 @ 23:40:19 EDT (456 reads)
(comments? | Score: 0)
|
|
|
| Basic Call Termination With SipTang |
Basic Call Termination With SipTang Digital Gateway Zhou500 or Analog Gateway Shang332 IntroductionCall termination is a very common type of application where a customer installs a VoIP Gateway at their location and they have this connected to the local PSTN lines. The customer then engages an international provider to send VoIP calls to his unit for termination within his country to take advantage of local rates. The customer charges a rate back to the call providers for this service. This document will provide users information on how to use SipTang Digital Gateway ZHOU500 or Analog Gateway SHANG332 to terminate VoIP calls to PSTN lines. We will provide specific information related to the termination of calls only. For more information and questions, please visit our website at www.siptang.com Application DescriptionIn a typical use, you would have a SipTang Digital Gateway ZHOU500 or Analog Gateway SHANG332 installed in your location. This unit would then have a connection to the Internet (via a router) and then some number of connects to your local PSTN provider. Depending on the amount of traffic that you will receive, you may choose T1(or E1) connections, as shown in example 1, or analog lines, as shown in example 2. Example 1 Example 2 They will receive VoIP calls from providers that they have contracted with to provide traffic for a particular country or area. Typically, these providers will send the calls with a specific number format and SipTang SHANG332 needs to be configured to accept the call and terminate it to the first available analog line or E1 channel and dial out the correct digits. ConsiderationsWhen setting up for this type of application, the following issues should be taken into consideration and in some cases are necessary to know before configuring. Type of connection to PSTN and Number of Lines You must decide what is the best connection to have from the PSTN provider to SipTang SHANG332. Many things must be taken into consideration for this. n Price This is always the biggest consideration. Not so much as the initial installation, but the monthly charges and rates that you get will, many times, determine the type of connection. If you choose a T1 (or E1) connection, the monthly fee will be more expensive that that of several analog lines. Depending on the amount of traffic that you will receive, you may not be able to make enough money to pay for a T1 or E1 line, or you may need to charge a higher rate to your provider and at some point the provider may decide to switch to a less expensive competitor. n Number of calls to support. You need to determine how many calls you want to support and see what type of PSTN connection and how many will support your requirements. You should choose T1(or E1) connections when the traffic is more than 30 concurrent calls, otherwise, Analog lines would be better choice. n Availability. Not all countries have all types of connections. Some may only have analog connections. You should check with your local PSTN provider about this. IP Bandwidth and Quality You need to determine how much bandwidth you will need. You can do this through some simple math. You first determine how many VoIP calls you want active at the same time on your SipTang SHANG332 and multiply this by the bandwidth required based on your audio compression. If you plan to use G.729, then figure on about 19kb per call in each direction. If you plan to use G.723.1 @ 6.3kb, the bandwidth will be about 13.5kb per call in each direction. For the quality, this will mainly depend on the ISP that you use for your Internet connection. If it is a lower tier ISP, the quality may not be there in that you may experience high packet loss or long delays on your Internet connection. You should discuss this with your ISP. To reduce the effect of bad internet connection, SipTang SHANG332 provide many QoS features, such as Dynamic Jitter Cancellation, Voice Activity Detection, Silence Suppression, Echo Cancellation(up to 128ms), etc. Provider Information and Security At a minimum, you will need to know what the number/digit pattern is that your providers will send to you. Typical patterns are international prefix (like 00 or 011) + CountryCode + number, or Countrycode + number. In many cases, providers may add a special prefix to the front of the number. This is usually a 4 or 5 digit number (could be more or less) that gets added to the front of all numbers and is sent as part of the phone number. For example, if the prefix is 6789, then they may send you the number as 6789+0101234567890. It is very important to know what the digit pattern is that your provider will send to you as SipTang SHANG332 needs to have these patterns configured in it to allow the call to terminate. You may also want or need to know the provider’s IP address for security. SipTang SHANG332 allows you several ways restrict access to it. The easiest way is a simple access list of allowed IP addresses. More complex would be with a radius server for authentication. SipTang ZHOU500 have an integrated billing system which can be used for authentication. Load Balancing and redundancy You may have several units installed at the same location to terminate calls. In these situations, most customers are looking for calls to be distributed evenly over all the units or in the least, if the first unit is filled, for the call to overflow to the next. This feature can only be used from the call origination point, not from the termination. Once the termination SipTang SHANG332 receives a call from IP, it cannot re-route it back to IP to another termination unit. You should discuss this with your provider/originator to have them perform this function at their side. In some cases, you may have several connections from different PSTN provider to avoid the effect of single PSTN lines’ failure. You can use SipTang ZHOU500 to route the traffic to different PSTN lines for load balancing, redundancy and overload protection, in order to assure the stability, robustness and scalability. Operation & Maintenance Ease of operation and maintenance is important to provide excellent service, SipTang ZHOU500 have an integrated billing system which can provide flexible billing functions and diverse statistics, such as calls analysis, average call duration, channel use factor, etc. The billing system is easy to configure and monitor with Web GUI interface. These features are helpful for operation and reasonable service optimization. |
|
|
Posted by luis on Monday, April 07 @ 23:30:56 EDT (769 reads)
(comments? | Score: 0)
|
|
|
| Network Construction: SHANG332 for Small / Medium business VOIP application |
SHANG332 for Small / Medium business VOIP application
1. Gerneral
For SME business, SHANG332 can be used to integrate old telephone network and change them into voip to save the telephone cost. Compare to install some ip telephone, company user need more design to obtain unique outlook, compatible to old system, best cost rate.
According to company’s situation, there are two kind of solution, one is with PBX, the other is no PBX.
2. For the company without PBX
ü All long distance call or even the oversea call will be turned to voip. The cost will be saved very much.
ü Internal call is free, especial for those call between different branch.
ü For those lines such as local call, it will still over the pstn line to get more convienent.
ü No need to add new telephone, easier to use both voip and pstn.
ü All telephone numbers are kept. The old pstn telephone continues working.
ü User’s dial pattern keeps no change.
Reform method:
the company had our straight lines before reform, the telephone number is 010-88880002 ,010-88880001 ,010-88880003 and 010-88880004.
1. install one set SHANG332 equipments between the telephone and the PSTN outside line, a 4 fxs 4 fxo card modle, the outside line connects on the FXO ports, the telephone connects the FXS in equipments. 1st outside Line connect the 1st FXO port. the telephone connect the 1st FXS port, and so on.
3、Regester 0001, 0002, 0003 and 0004（free for internal call）
Configuration details:
FXS port IP information
configure the IP address and the Server parameter in the quick setup page.
At"port configuration" page, set the number of those pstn line 88880001、88880002、88880003、88880004 ‘s voip number 8001、8002、8003、8004.
At"Routing direction" page, add 2 new lines, the number 8xx and number prefixed 0 will be routed to VOIP. the number 8xx is for internal call, number prefixed 0 is for long distance call.
2、FXO call out
In the “FXO advanced configuration” page, set its “working mode” to Dual mode. Enable “IP routing Fall back to PSTN”. Those number not prefixed 0 will be send from fxo directly.
with this solution, the end user’s dial usage was kept. And if IP network error, call can be routed to PSTN.
3. For the company who has PBX
a) Solution 1
For those company who has PBX, the end user will get the following benefit from SHANG332:
all long distance call or even the oversea call will be turned to voip. The cost will be saved very much.
Meet 911 requests, under the circumstance that the power is lose or the internet network is breakdown, all call still from the original circuit, don't break off the usage of the telephone.
User’s dial pattern keep no change.
No need to change the whole telephone system, but upgrade to VOIP. It is a economic way to improve.
Reform method: This company has an reception number 010－82901131 with 2 lines connected to PBX directly. this PBX has 4 extensions which numbered 8001、8002、8003 and 8004. install one set SHANG332 equipments between the telephone and the PSTN outside line, the outside line connects on the FXO ports, the PBX connects the FXS in equipments. Connect a net line from the WAN of the equipments, link the company IP network. The configuration details is similar to the before.  Reform method: This company has an reception number 010－82901131 with 2 lines connected to PBX directly. this PBX has 4 extensions which numbered 8001、8002、8003 and 8004. Dial pattern: 0 for outboundry call, 8 for internal call. After reform, add a new router. 9 for voip , install one set SHANG332 equipments between the PBX and the extensions telephones. It use dual mode. the telephone connects on the FXS ports, the PBX connects the FXO in equipments. Connect a net line from the WAN of the equipments, link the company IP network. The configuration details of SHANG332 is similar to the before. |
|
|
Posted by luis on Monday, April 07 @ 18:44:32 EDT (760 reads)
(comments? | Network Construction | Score: 0)
|
|
|
| How to reset Jin501 |
If you found that your Jin501 don't work after being configured, you must do something wrong with it, so the best way is restoring factory defaults. You can reset your Jin501 as follows:
1. connect Jin501 with your computer.
2. configure the ip of your computer with 192.168.10.50
3. keep on pressing the “reset” button which is in the base of your Jin501, power Jin501 at the same time.
4. telnet 192.168.10.1 in your computer
5. after entering, input 3.
6. now the factory defaults has been restored.
|
|
|
Posted by alex on Tuesday, November 13 @ 02:29:22 EST (573 reads)
(comments? | Score: 0)
|
|
| |
| Contact Us | |
MSN
sale1 |  | SKYPE
sale2 |  | YAHOO
sale3 |  |  |
| |
| Languages | |
Select Interface Language:
| |
| Who's Online | |
There are currently, 4 guest(s) and 0 member(s) that are online.
You are Anonymous user. You can register for free by clicking here | |
|
|
