VoIP- FAQs on troubleshooting
I have heard dialing *99 on Vonage helps faxing, is this correct?
Our small office uses VoIP lines and at some times our call quality is bad. We have been unsuccessful in correcting it; what else can be suggested.
Troubleshooting and Correcting Problems----------------
This issue is most likely the cause of either double NAT or a non-friendly NAT router preventing the return audio packets from getting all the way back your phone. See our VoIP Tutorials to correct many if these issues.
The first thing is make sure your RJ-11 phone cable connection to your fax machine is without splitters or other devices in the path. Then set the BAUD rate (transmission speed) of the fax machine to 9600 and turn off ECM (error correction mode). We have listed how to make these settings on some fax machines such as HP fax machines, Brother IntelliFax, Sharp fax machines, Panasonic fax machines, and Canon fax machines.
ECM sends information and allows the fax machine which is transmitting to communicate with the receiving fax so that it knows know if bits of data have corrupted. Loss of information can happen on noisy telephone lines, as well as VoIP transmissions. When receiving information that some bits of information have been lost or corrupted the fax machine will retransmit the corrupted or lost bits, which ensures a high quality document.
Yes, although your document may not be as clear as what was sent, if it were to totally fail your fax machine would tell you through an error code. In some business cases the received document may be unacceptable, depending on the receivers needs. (For example a multipage real estate contract which would need to be clear and easily readable throughout.)
It is correct. On some ATAs dialing *99 forces the transmission to use a G711 codec (non-compressed), which is needed for successful faxing. It also disables call waiting, another setting for faxing. Many newer ATAs adjust these settings automatically when detecting a fax sound, but you still should lower the fax machines BAUD rate to 9600 and disable ECM.
There are ways to correct this depending on several factors, including your available bandwidth, (both the upload and download). We suggest that you take a few steps in troubleshooting your VoIP connection change your codec to G729 and possible enable QoS on your router if available.
If after trying the suggestions that have been made for checking your LAN network, bandwidth and connection reliability, enabling QoS on your router and you still experience break-up then you might consider a better more robust router that can handle SIP and QoS well. Two choices would be an Edgemarc or a router with DD-wrt or Tomato installed.
Yes. If you have access to your ATA device, then there are a few settings that will help you have success faxing over VoIP. Try these settings including adjusting the Jitter Buffer size.
If you have tried our suggestions on how to fax over VoIP and your faxes still fail, then you might consider an efaxing solution which is a service that you subscribe to that handles your faxes via email and offers other features, as well.
This sounds like a bandwidth issue. Check both your upload and download bandwidth. Turn off any computers that may be running an application that is using too much of the available bandwidth.
The best way to correct this is to either use a compressed codec like G729 or configure your router for QoS for VoIP.
Turn down your phone volume. Acoustic feedback from the speaker to the microphone of a telephone handset can cause echo. Keep your ATA, as well as your cordless phone's base station away from other electro magnetic devices. Unfortunately, most ATAs are poorly shielded. We have included in our VoIP Tutorials some information about echo and what causes it.
In some cases, especially during periods where echo is present when none existed before, it can be due to network (WAN) connections and/or your carrier. In these cases the echo will most likely be short lived (days). In some cases, where you hear echo more often when you call specific people, (most likely local) this can be because the local phone carrier is not using proper echo cancellation techniques and the result will magnify echo with VoIP calls. Baring the fore mentioned cases some echo can be caused by faulty or corroded wiring. Check your phone jacks and make sure the wire and connections are clean. In older homes where the wiring may be old, replacing the wiring may be the only solution.
It is not normal to hear buzzing. This can be caused by interference with cordless handsets or faulty grounding in the phone circuitry. Try another corded phone to see if the buzz stops. Connect your phone directly to the IAD to eliminate any possible faulty wiring problems in the home circuitry.
There is a good chance that hum is being caused by an incorrect power supply (if you are not using POE). Check with the manufacturer for the correct power supply and check that grounding is correct.
First, follow VoIP Mechanic's VoIP Fax Settings suggestions and make sure nothing has changed like your codec setting, fax machine settings or network setup. If after eliminating all possible causes it may be because your provider has changed carriers. They do this to lower their costs and the lower cost new carrier may not have as good of a network as the previous one. They probably will not tell you when and if they make these types of changes as they look at their business relationships as confidential.
This can be a result of a carrier routing particular calls over a route where high latency occurs. Your provider should be able to look at the problem calls and see if there may be a network link that is causing high delay times. Wholesale VoIP routes are typically rated by how good they are and are priced accordingly. Grey routes are lower cost, but lack in quality.
Some phones have a setting to automatically add a 1 to a long distance number. If this happens in some cases it will knock off the last digit of the displayed number.
This issue can happen after a "bulk port" of numbers by the provider. Instead of porting all numbers with exact specific information, the provider, for ease placed the group of numbers together under their name. The CLEC or entity that now owns the numbers then grabs the name supplied by the port record, or the provider's company name. This name, now the VoIP provider's name (and probably address, as well.) are associated with that number. The CLEC that holds the VoIP providers numbers now passes along the providers name when that number is passed to the next CLEC or phone company.
Check to see if call forwarding has been placed on the line or DND (do not disturb) has been set.
This is most likely a bandwidth issue. Since a single two-way call needs approximately 100kbps on both the upload and download side, a single three-way call would require twice that or 200kbps. (These figures include some bandwidth for the PC and are generally recommended by some VoIP providers.) There are some Codecs that use compression that can reduce the amount of bandwidth required. If theses codecs are being used, then less bandwidth would be required.
First, make sure you have an Internet connection. If you do have a connection to the Internet, but no dial tone, do a complete power cycle, starting with the modem, router and IAD. Power them back in the sequence, modem first, then the router and last the IAD, making sure to give the modem some time to reconnect with the ISP. See our VoIP Tutorials and VoIP troubleshooting sections.
Try to determine the IP address of the IAD by using an IVR command response, if you are using one with this feature. If it is 0.0.0.0 then it is not communicating with DHCP server (NAT router). Try to reseat and check the Ethernet cable from the IAD to the router. If that does not work try connecting the IAD directly to the modem and do a power cycle. In the case of a Linksys ATA/Router check the front panel lights for a solid power light and a solid phone 1 light when the phone is on hook.
If you haven�t changed anything on your end, it could be that your ISP is having issues with their network. You could check for increased latency and packet loss. Give your ISP a call and ask them if anything is going on.
Yes, actually Jitter is a key measure of VoIP quality and can increase during some periods. Jitter is the variation in time between packets sent and packets arriving caused by network issues such as route changes, router congestion, packet loss, and other network or ISP related issues. VoIP is especially sensitive to Jitter, so a MOS measurement application can be a valuable tool to identify these instances.
Connect the IAD into the room�s Ethernet connection. You will need to know the IVR command to get the assigned IP address of the IAD. (For instance, with a Linksys/Sipura 2000, this would be ****, then the command 110#.) Once you get the assigned IP address, call the hotel�s help desk and with that information they should be able to log the device on. There could be other issues with firewalls, but in many cases by following these instructions you will be able to use your VoIP connection.
This can be caused form your VoIP provider doing a bulk port of numbers from one carrier to another and listing these numbers under their own name and address. Then after the port when the carrier dips into their database to send on the caller ID name they pass the name of the provider, even if the provider passes the correct name. Until the carriers database gets changed to reflect your name, the display will show the VoIP providers.
Dial the number as a 10 digit number. If you still do not get through, call your provider and tell them the number and time of the call and they can look at the situation. In some cases, a LERG update would need to be done by carriers to correct this issue.
This can happen if your provider placed numbers together under one port request (called a bulk port) and moved them to another carrier. To do this they pace many numbers under their name and address. Once the port occurs the new carrier now has these numbers under your provider's name and may be passing that name along.
VoIP calls can take a longer period of time for connection than traditional PSTN lines. In most cases, this time should only be a matter of a few seconds more. By pressing the # on the phone you can cause the call to connect right away. In many ATA devices, like the Cisco/Linksys ATAs there is a setting that can be adjusted to decrease the time before the call goes.
Yes, you should open a ticket with your VoIP carrier who can look into the situation. Calling a "bad number" over AT&T can confirm that the number is either good or not. If it fails with AT&T, then the number would be considered to be having a problem at the far end and they (the in country phone company) needs to correct it. If the number completes with AT&T with good quality your provider may be able to work with their international carrier to route the call over a better route for better quality.
Telecom companies are always looking for ways to increase profits and one way is to send their calls over cheaper carriers. These carriers are sometimes referred to as gray routes, but can also be referred to by numbers, such as Tier 1, Tier 2 or Tier 3 carriers. Each one having somewhat less quality than the other.
Having calls routed over the Internet can lead to occasional changes that result in less than ideal quality. Periods of high latency, either from congested routers or problematic routes, can cause packets to be dropped and dead spots in voice transmission. If unusually high latency occurs, the entire call could go dead.
In some cases, yes. Your provider or a carrier that your provider routes some calls over may have a preset disconnect. Typically, this time limitation might be 120 minutes or 180 minutes. It is done as a precaution or safe guard to prevent endless open connections. (see more about timed disconnects)
If a VADVoice Activation Detection algorithm that is being used is not advanced enough and silence suppression is enabled then some of your voice could get cut off or "clipped" during a conversation.
This could happen if your IAD is not connected to the Internet and ready to receive calls and you do not have voice mail activated with your provider. The initiating call cannot be directed to a recipient, either the IAD or the VM server, so a 404 �not found� is returned. If voice mail is active, then even if the IAD is not registered the call should go to voice mail.
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