Our Blog

Today we’re handing over the blog to Scott O’Dell, a Technical engineer here at Broadtela. He works on the front lines assisting our customers and is a wealth of information. Take it away, Scott!

Faxes can fail for several reasons, but this post will touch upon the more common ones.

Since the faxes are sent across the public internet, if they were to traverse a major backbone that was having issues, then this could cause the faxes to fail. Anywhere from latency to packet loss can add to these failures. Familiarizing yourself with the errors can give you an idea as to what the issue is and where to act.

The HDLC Carrier Did Not Stop in a Timely Manner

This means that our fax server initiated a fax transmission with the far end, but there was a timing error that could not be resolved. This happens when the far end sends their Digital Identification Signal indicating their fax capabilities, such as data rate. Our fax server would then respond with our Digital Command Signal indicating our transfer rate. Usually faxes will start at 14,400 which is the Maximum bits per second speed. If that is too fast for the far end, then the appropriate response would be a Failure to Train. Our fax server would then negotiate a slower speed, but if the Term End does not respond then the end user would receive an error that the HDLC carrier did not stop in a timely manner.

Timed Out Waiting for Initial Communication

This error indicates that our fax server is waiting on the far end to send its initial Digital Identification Signal to continue the T.38 negotiation. The fax cannot proceed any further until the far end sends its signal first. If our fax server does not receive the signal, we will make attempts to talk to it by sending a CNG tone. Or, we would simply put a fax calling tone wherein our system would be playing fax tones trying to communicate with the far end. If we do not receive anything, our fax server would then tear the call down, and the end user would receive the error in their inbox.

Invalid Response after Sending a Page

This error indicates the far end did not respond with the proper message to continue the fax flow and the fax was terminated. An example would be our fax server sending an End of Procedure message indicating there are no more pages to be sent, and the Term End also responding with an End Of Procedure in lieu of a Message Confirmation, indicating the last page was received. Below is a screen shot from a Wireshark packet capture illustrating this behavior. As you can see the arrow pointing to the right is our fax server sending the EOP, and the arrow pointing to the left is the Term End responding with the same thing; thus, causing the fax error.

Disconnected after Permitted Retries

This error occurs when our fax server is trying to negotiate speeds with the Term End. When we start out at 14,400 bits per second, if that is too fast the Term End should respond with a Failure to Train. If they do not, and they keep sending us their receiving rate, the fax will fail with that error.

Far End Cannot Receive at the Size of Image

This would indicate the attachment is greater than the standard 8.5 x 11 document size. You can check this in Adobe Reader by clicking File->Properties->Page Size. Oftentimes scanned documents will exceed these dimensions which can cause failures with the Email-to-Fax service. To overcome this potential problem, you can use a third-party program such as Cute PDF to re-save the file before attaching it. Cute PDF has a default size of 8.5 x 11 so the only thing you are doing when using it is shrinking the document you have to fit the standard paper sized used in most fax machines.

Fax failures are not limited to these errors. Troubleshooting fax failures can be difficult due to how faxes are set up. All faxes start as voice calls until fax tones are detected. Once these are detected the negotiation process begins and if any packets were to break down during this process can cause the faxes to fail. If this were to happen, you should check the number you are faxing to ensure the far end is indeed playing fax tones. You can confirm this by placing a test call to the destination. If the far end is playing fax tones make note of how long into the call the term end starts to play these tones. If the term end is taking quite some time, then that can be the cause of the failures. This is because the sending side is waiting to hear fax tones, but if it is taking the far side too long, then the sending side may disconnect the call. If that is the case then the Term party would need to investigate their equipment.

Customers have seen an increase up to 98% success rate when using our Vfax service as we recently turned up a new vendor. Using our Vfax service allows our support team to pull the necessary packet captures in the event of a failure to quickly identify where the issue is originating from. We can then advise the customer the necessary steps to resolve the issue.

Hopefully this post gave you more insight to the Email to Fax service. If you ever need assistance with anything at all, please give me or one of my fellow Technicals a call!

There are two types of call transfer that your system if capable of:

  1. Blind transfer:  This is where the call is directy transferred to another user without any introduction.  Following the transfer, the other phone will ring showing the callerid of the original caller.  If there is no answer, the call would go to the voicemail of the numebr the call was being transferred to.
  2. Attended transfer:  In this scenario, the user transferring the call would have the opportunity to speak to the transfer recipient before the call is transferred.  The recipient on the transferred number sees the callerid extension  number of the user transferring the call rather than the original callerid.  If the recipient is not available or does not want to take the call, the user can return to the original call.

The system is also capable of making conference calls joining three calls.

Blind Call Transfer

YLXfer1

In the first call, press the TRANSFER key

YLBXfer2

Enter the number to transfer to and then press the TRANSFER soft key

The calls will be transferred and the phone will return to the ready screen.

Attended Call Transfer

YLXfer1

In the first call, press the TRANSFER key

YLXfer2

Dial the number that you want to transfer the call to and wait for the call to be picked up.  You may talk with the recipient.

YLBXfer2

To complete the transfer of the call, press the TRANSFER key

The call will be transferred and the phone will return to the ready screen.

In the situation where the recipient of the transferred call is not available, or does not want to take the call, the user should just hang up and return to the original call by pressing the first green flashing line key.

Conference Calling

YLConf1

Initiate first call. Press the conference key

YLConf2

Dial second call.  Once connected, press CONFERENCE key .

YLConf3Man

During the conference call, the MANAGE soft key allows the user to manage the other parties

YLConf4Man

Using the up and down arrow keys, one of the other parties can be selected and their line mutes or removed

Important VoIP 911 Information

Enhanced VoIP Communications provides all customers with 911 access to a national emergency call center.  In the case of an emergency you should dial 911 on your telephone to access 911 services.  When you reach a 911 agent, the agent will ask you which emergency service you need: fire rescue, emergency medical assistance or law enforcement.The agent will then ask you for the address where you need the emergency service.  You should be prepared to confirm your address and call-back phone number with the operator.  Please do not hang up unless told directly to do so and if disconnected, you should dial 911 again.  You should print your address and phone number and keep it close to each of your phones or work areas so that visitors and guests can quickly determine where emergency services should be dispatched to. Please see below for an important note regarding keeping your physical location(s) up-to-date in our database.


Keeping your Physical Address up-to-date

Enhanced VoIP Communications provides a web based interface for updating your physical location(s). Customers must review this information and keep it current at all times. In case you are not able to speak during a 911 call, the emergency personnel will dispatch response vehicles to your last registered address. You need to update your VoIP 911 information if you move your device to a different physical location. If different extensions have different physical locations you will need to setup multiple emergency locations and assign each extension to a different location.


Backup recommendations

It is important to note that in the event of a power failure or loss of Internet connectivity at your office you will be unable to dial 911.  You should inform all employees and visitors of this important limitation of 911 service.  The use of a small inexpensive battery backup device available at most electronics retailers will prevent short term power failures from affecting your telephone service.  It is a good idea to have a backup plan for dialing 911 in case of an extended power failure or loss in Internet connectivity.  This might include keeping a cell phone available nearby or attaching a phone to your regular fax phone line.


Important Note Regarding Test Calls

Please do not use our service to make 911 test calls from your telephone.  If you wish to perform a 911 test you must arrange this with our customer service group.  Unscheduled test calls to 911 services will automatically be charged a $95.00 service fee.  Only dial 911 in an emergency.

0

Router Compatibility Guide

Introduction

The following is a list of known routers and modems which we have determined to be compatible, incompatible or compatible with changes to the configuration of the device. We have also provided a list of devices which we are aware of but have not yet tested ourselves.We highly recommend checking all network equipment on site against this list before deploying our Hosted PBX service in your office, or, if you are already using our service, before making changes to your existing network.If your router is not listed, please contact an Easy Office Phone representative to discuss. Should you have additional information on a device and would like to add your findings to our list, please contact us.Whenever possible you should choose a router that has QoS (Quality of Service) if you plan to share the same Internet connection with non-VoIP services (such as desktop computers). We also recommend that you always make sure your modem is bridged and updated to the newest firmware available before troubleshooting unless otherwise specified by this guide. This list is modified when we become aware of new information, and we recommend you check back periodically for any updates.


Quick Links


Compatible Devices

Apple Airport

Current Status: Compatible
Recommendation: Suitable for 1-3 phones
Comments: There are no known issues with this router although it is best used for smaller offices (1-3 phones). This device is not believed to offer QoS which may be a drawback in certain deployments.Back to Top


Cisco Routers (general)

Current Status: Compatible; may require changes
Recommendation: Confirm model with Easy Office Phone
Comments: The Cisco routers we have encountered appear to be quality devices that are compatible with our service, but may require some changes.


Cisco 837

The router’s NAT service for SIP should be disabled, with the command:
no ip nat service sip udp port 5060Back to Top


Cisco RV180W

Current Status: Recommended
Comments: This is a business-grade multifunction router, including Gigabit Ethernet, VPN and Wireless-N capabilities. We deploy this unit on a regular basis and are able to recommend it for a range of small to medium business needs.This unit offers four Ethernet ports; consider adding an unmanaged switch if you require more ports.Back to Top


Cisco Linksys RV082, RV016, RV042, RV042G

Current Status: Compatible if correct firmware version is used. May require settings change
Recommendation: Recommended
Comments:This family of routers will work well with our Hosted PBX services so long as the firmware version on the unit is v4.2.1.02. Other versions may cause issues. The MTU settings under ‘Setup’ -> ‘Network’ should be set to 1500 (Cable) or 1492 (DSL).Advanced tip: If this router has issues with phones not ringing, the issue may be UDP sessions being closed early. Our staff have resolved this by upgrading to 4.2.1.02 firmware, then going to the hidden page at https://(router LAN IP)/f_general_hidden.htm and then changing the UDP Timeout from 30 seconds to 300 seconds.The RV series offer an excellent QoS feature which is an advantage over most competing routers.If you have a large number of devices to connect to your router, the 4-port RV042 may not offer enough ports for you. In this case, choose the RV082 or RV016 for their larger number of ports, or alternatively purchase an unmanaged switch and connect it to your RV042 to increase its port capacity.


Cisco/Linksys WRVS4400N

Current Status: Compatible
Recommendation: Recommended
Comments: This is an excellent router with all the features of the RV-series routers (above) that also offers Wireless-N connectivity and Gigabit Ethernet speeds. The router has four ports; if more are required we recommend connecting an unmanaged switch to the WRVS4400N.Back to Top


Cisco/Linksys E1200

Current Status: Compatible
Recommendation: Recommended
Comments: Suitable for small office of 2-5 phonesBack to Top


Cisco/Linksys E2500

Current Status: Compatible
Recommendation: Recommended
Comments: Suitable for small office of 2-5 phones and offers stronger Wi-Fi than the 1200 with Wireless-N.Back to Top


Cisco/Linksys E3500

Current Status: Compatible
Recommendation: Recommended
Comments: Suitable for small office of 2-5 phones, and offers very strong Wi-Fi.Back to Top


Cisco/Linksys WRT160N

Current Status: Compatible
Recommendation: Suitable for 1-2 Phones
Comments: There are no known issues with this model router at present. The router does not offer QoS, but this may not be a major issue in offices with only one or two employees. We would not recommend this router for larger-scale deployments.Back to Top


Cisco/Linksys WRT54G2

Current Status: Compatible
Recommendation: Recommended for 1-2 phones
Comments:
This router is compatible and recommended for 1-2 sip phones. Appears to have problems with 3 or more SIP phones caused by NAT issues. Use DDWRT firmware instead in this case if hardware revision allows for this.Back to Top


Siemens Speedstream

Current Status: Compatible
Recommendation: Suitable for 1-3 Phones
Comments: There are no known issues with this router. This router has no QoS (Quality of Service) which is a drawback if you are using it in an environment where there are computers or other non-VoIP equipment. We would not recommend this device for larger-scale office deployments.Back to Top


Sonic Wall

Current Status: Compatible
Recommendation: Firmware v5.8.1.12 or lower, Disable SIP ALG
Comments: Firmware v5.8.1.13 and higher contain a bug that causes issues with incoming calls. We recommend customers set enable ‘Consistent NAT’ (check the box) and disable ‘SIP Transformations’ (uncheck the box). Ensure that the MTU is set correctly for your ISP.Select the Firewall Settings -> Edit LAN>WAN Rule -> AdvancedTCP Connection Inactivity Timeout (minutes) to 60
UDP Connection Inactivity Timeout (seconds) to 3600Select the Firewall -> Access Rules and add a new rule with the following:

    Action: Allow

Service: Any
Source: WAN, Address Range 67.223.99.1 to 67.223.99.254
Destination: LAN, Address Range * to *
Comment: Easy Office Phone D1
If there is an option for ‘Allow Fragmented Packets’ then please enable this option.
TCP Connection Inactivity Timeout (minutes) to 60
UDP Connection Inactivity Timeout (seconds) to 3600Select the Firewall -> Access Rules and add another new rule with the following:

    Action: Allow

Service: Any
Source: WAN, Address Range 64.15.155.1 to 64.15.155.254
Destination: LAN, Address Range * to *
Comment: Easy Office Phone D2
If there is an option for ‘Allow Fragmented Packets’ then please enable this option.
TCP Connection Inactivity Timeout (minutes) to 60
UDP Connection Inactivity Timeout (seconds) to 3600Back to Top


Devices believed compatible after changes

2Wire Routers and Modems

Current Status: Possibly compatible after Changes
Recommendation: Disable SIP ALG or Bridge to Supported Router
Comments: To use this router you need to login to the web interface for the router and disable the ‘SIP ALG’ setting on it. If this option is not available you will need to bridge the router to a compatible router so that the 2wire router acts as a simple modem rather than a router.If bridging is not possible either, you must replace this device as it will prevent you from using our service.


D-Link DIR-615

Current Status: Compatible after Changes, Hardware Revision B Only
Recommendation: Disable SIP ALG
Comments:
This applies to Hardware Revision B only. Other hardware revisions are not compatible with VoIP as they do not have the disable SIP ALG option.This router has known issues with VoIP services if SIP ALG is not disabled. We recommend disabling the SIP ALG setting on the router available in firmware releases 2.2.4 and higher. This option is available within the firewall settings under the ‘Application Level Gateway (ALG) Configuration’. The SIP option should be unchecked.Back to Top


D-Link DIR-625

Current Status: Possibly Compatible after Changes
Recommendation: Disable SIP ALG
Comments:
We recommend disabling SIP ALG through the DIR-625’s web interface. After the change, this router appears to be compatible and able to serve up to three or four phones. Customers who have tried to connect a larger number of phones to this router have found that the additional phones fail to register with our server. We would not recommend this router for a larger-scale deployment as it does not appear to perform reliably.Back to Top


D-Link EBR-2310

Current Status: Compatible after Changes, Hardware Revision B Only
Recommendation: Disable SIP ALG
Comments:
This applies to Hardware Revision B only. Other hardware revisions are not compatible with VoIP as they do not have the disable SIP ALG option.This router has known issues with VoIP services if SIP ALG is not disabled. We recommend disabling the SIP ALG setting on the router available in firmware releases 2.2.4 and higher. This option is available within the firewall settings under the ‘Application Level Gateway (ALG) Configuration. The SIP option should be unchecked.Back to Top


Fortigate/Fortinet

Current Status: Compatible after Changes
Recommendation: Disable SIP ALG
Comments:
In the CLI of the fortigate type the following:config system settings
set sip-helper disable
set sip-nat-trace disableReboot the deviceIn the CLI type the following:Config system session-helper
show
(now look for SIP, mostly it will be “12”)
delete 12

Don’t use any protection profiles on the firewall of the sip rules.Back to Top


Juniper – Netscreen

Current Status: Compatible after Changes
Recommendation: Disable SIP ALG
Comments: You must disable the SIP ALG on the Netscreen router. Navigate to the ‘Configuration’ page and choose ‘Advanced’. Then choose ALG and un-check the SIP check box.

Back to Top


Motorola Surfboard

Current Status: Compatible after changes
Recommendation: Disable SIP ALG and ensure appropriate firmware is running
Comments: The following changes/checks are required:

1) Must be running firmware v3.3 or higher
2) Must have SIP ALG disabled.

Back to Top


Netgear FVG318

Current Status: Compatible after Changes
Recommendation: Update firmware and disable SIP ALG
Comments:
This router is known to cause issues related to storing configuration and failed firmware downloads with Aastra series phones. Aastra phones on this router require frequent reprogramming and firmware replacement. This router appears to work with VoIP services with the SIP ALG setting is disabled. The router does not work properly with VoIP with the SIP ALG setting enabled (which is enabled by default). You should make sure you have the most recent firmware for this router before using it. To disable the SIP ALG, first login to your router and select ‘Rules’ from the left hand side menu. Next, check the box that says ‘Disable SIP ALG’.

Back to Top


Netopia

Current Status: Possibly compatible after changes
Recommendation: Disable SIP ALG or replace device
Comments:
Netopia has SIP Passthrough “on” by default and will not work with VoIP services properly unless SIP ALG is disabled. (Or SIP ALG in newer models) In order to work with voice-over-IP, you must turn SIP Passthrough (SIP ALG) off. Note that not all Netopia routers may allow this change; if yours does not you must replace the device with a compatible router in order to use our service.

The SIP ALG setting is not in the web gui. You have to telnet into therouter, then type. Each line is a new command.

configure
set ip sip-passthrough off
save
exit
Restart

On the Netopia 4622xlt
Telnet into the device
Use the arrow keys to highlight ‘quick menu’ and press “ctrl-n”
That brings you to the console.

type:

ip nat alg sip enable no
save
exit
restart

Back to Top


SMC 3100

Current Status: Possibly compatible after changes
Recommendation: Bridge device to supported router
Comments: This router is not compatible with VoIP services unless it is bridged. We recommend bridging the router to alternative router so that it acts as a modem only.

Back to Top


Tomato Firmware Routers

Current Status: Possibly compatible after changes
Recommendation: Adjust router settings
Comments: We have encountered a scenario in which a router running the Tomato firmware would only let one phone register at a time. We were able to resolve this problem by entering the router settings, under ‘Advanced’ and then ‘Conntrack/Netfilter’, where we enabled the SIP option for ‘Tracking/NAT Helpers.’ The router on which this was encountered was an Asus RT N16 but this issue and resolution may apply for other routers running Tomato.

Back to Top


Zxyel

Current Status: Possibly compatible after changes
Recommendation: Disable SIP ALG or bridge device to router
Comments:
This router is not compatible with VoIP services without modifications. We recommend either bridging the device to another router or disabling the SIP ALG. To disable the SIP ALG you must first telnet into the router/modem. Choose (24) System Maintenance and (8) Command Interpreter Mode. From the command line type:

ip nat service sip active 0

You will receive the message “SIP ALG Disable”

Back to Top


Incompatible Devices

Actiontec GT704WG

Current Status: Not Compatible
Recommendation: Replace device
Comments:
This router is not SIP Compatible. Drops calls after 20 seconds.

Back to Top


Asus RT-N66U

Current Status: Not Compatible
Recommendation: Replace device
Comments:
This router is not SIP Compatible. It has no SIP ALG or SIP Fixup options and does not work with polycom, aastra or cisco phones.

Back to Top


Belkin

Current Status: Not Compatible
Recommendation: Replace device
Comments:
This router is not compatible with VoIP services. We cannot support this router in any way and it must be replaced before using our service.

Back to Top


D-Link DIR-825

Current Status: Not Compatible
Recommendation: Replace Device
Comments:
This router is not compatible with VoIP.

Back to Top


D-Link WBR-1310

Current Status: Not Compatible
Recommendation: Replace Device
Comments: This router is not compatible with VoIP applications as it lacks an option to disable the internal SIP ALG which interferes with IP Telephones.

Back to Top


Linksys WRT54G, WRT54GX, WRT54GL, WRT54GS

Current Status: Not Compatible
Recommendation: Replace Device or Replace Firmware
Comments: This router is not recommended for use with VoIP services using the stock firmware from Linksys. This model router is known to damage and modify SIP packets in such a way that it prevents VoIP packets from being formatted correctly and can cause a multitude of problems.

You may be able to install custom firmware such as DD-WRT () as a replacement. Note that the procedure does require some technical knowledge and ability. The DD-WRT firmware allows you to disable SIP ALG and enable QOS (Quality of Service), but this has not been tested in our labs yet.

Back to Top


Thompson TG-585

Current Status: Not Compatible
Recommendation: Replace Device
Comments:
This router is not compatible with VoIP services. There is no method to disable the internal SIP ALG on device. Other customers have found that bridging the device to a compatible router did not result in a workaround. We recommend replacing this device with a compatible modem and router.Back to Top


Netgear FVS318

Current Status: Not Compatible
Recommendation: Replace Device
Comments:
This router is not recommended for use with VoIP services. This router causes issues related to storing configuration and failed firmware downloads with Aastra series phones. The router appears to work some of the time but lacks the ability to disable the SIP ALG which causes intermittent issues with phone service. If possible we would recommend avoiding this router until Netgear makes it possible to disable the internal SIP ALG setting on the router to be compatible with voice-over-IP.Back to Top


Netgear WGR614

Current Status: Not Compatible
Recommendation: Replace Device
Comments:
This router lacks the ability to disable SIP ALG and is not compatible with VoIP services.Back to Top


Netgear WGT624

Current Status: Not Compatible
Recommendation: Replace Device
Comments:
This router lacks the ability to disable SIP ALG and is not compatible with VoIP services.Back to Top


Netgear WRN2000

Current Status: Not Compatible
Recommendation: Replace Device
Comments:
This router lacks the ability to disable SIP ALG and is not compatible with VoIP services.Back to Top


SMC routers – general

Current Status: Not Compatible
Recommendation: Replace Device
Comments:
Most SMC routers do not appear to be compatible with VoIP services. We would suggest that you either replace this router with a compatible one or attempt to bridge it to another router.Back to Top


Devices not yet Tested

Cisco Pix

Current Status: Untested
Recommendation: Untested
Comments:
We have Untested this router in our labs, but the following information may be helpful.The cisco pix firewall requires that in version 7.0 or later look at the “inspect” fields. (Prior to version 7 it was called “fixups” not “inspect”.) Make sure that the “inspect SIP” field is NO. “Oo inspect sip” is the proper setting.With ‘inspect sip’ on the PIX firewall replaces the internal IP address of the phone with the external IP address. We use both pieces of information to accurately deliver the call. The internal IP address must be inside the SIP packet for proper functioning.Back to Top


D-Link DI-524

Current Status: Untested
Recommendation: Untested
Comments:
This router has not been tested for compatibility with VoIP services. We recommend testing the router before using in a production environment.Back to Top


D-Link DI-604UP

Current Status: Untested
Recommendation: Untested
Comments:
This router has not been tested for compatibility with VoIP services. We recommend testing the router before using in a production environment.Back to Top


D-Link DIR-628

Current Status: Untested
Recommendation: Disable SIP ALG
Comments:
This router has not been tested for compatibility with VoIP services. If the router has an option to disable SIP ALG we would recommend disabling it.Back to Top


D-Link DIR-655

Current Status: Untested
Recommendation: Disable SIP ALG
Comments:
This router has known issues with VoIP services if SIP ALG is not disabled. We recommend disabling the SIP ALG setting on the router. This option is available within the firewall settings under the ‘Application Level Gateway (ALG) Configuration’. The SIP option should be unchecked.Back to Top


D-Link DIR-855

Current Status: Untested
Recommendation: Disable SIP ALG
Comments:
This router has known issues with VoIP services if SIP ALG is not disabled. We recommend disabling the SIP ALG setting on the router. This option is available within the firewall settings under the ‘Application Level Gateway (ALG) Configuration’. The SIP option should be unchecked.Back to Top


D-Link DGL-4100

Current Status: Untested
Recommendation: Disable SIP ALG
Comments:
We suggest trying to disable SIP ALG by going into the router’s web interface and selecting Advanced -> Firewall -> Application Level Gateway and disabling (unchecking) the SIP checkbox.Back to Top


Dynex Routers

Current Status: Untested
Recommendation: Untested
Comments:
These routers have not yet been tested for compatibility with VoIP services. We recommended testing the router before deploying in a production environment.Back to Top


Firebox Routers

Current Status: Untested
Recommendation: Untested
Comments:
These routers have not yet been tested for compatibility with VoIP services. Discussions with engineers at Firebox indicate that the router is not likely compatible with SIP based devices such as IP telephones due to NAT concerns. We recommended testing the router before deploying in a production environment. If the router has SIP ALG or SIP Translations we would recommend trying to disable them.Back to Top


Hitron CGN3 (Rogers Advanced WiFi Modem)

Current Status: Not Compatible
Recommendation: Replace Device or Bridge it to a compatible router.
Comments:
The router feature of this device lacks the ability to disable SIP ALG and is not compatible with VoIP. We recommend that it be replaced with a standalone modem and router that is compatible, or that it be bridged to a router is compatible. We DO NOT recommend placing a phone on the DMZ of this decvice as it would compromise the security of the phone.


Hotbrick Routers

Current Status: Untested
Recommendation: Untested
Comments:
These routers have not yet been tested for compatibility with VoIP services. We recommend testing the router before deploying in a production environment. If the router has SIP ALG or SIP Translations we would recommend trying to disable them.Back to Top


Linksys BEFSR41

Current Status: Untested
Recommendation: Untested
Comments:
This router has not been tested for compatibility with VoIP services.Back to Top


Linksys RVS4000

Current Status: Untested
Recommendation: Untested
Comments:
We have not yet tested this router in our labs, but due to its similarity to the RV series routers (which are recommended) we believe it may be compatible.However, the SIP ALG setting on this router should be disabled. To disable the SIP ALG you will need to login to the web interface and select ‘Firewall’ and then ‘Basic Settings. Make sure that the ‘SIP Application Layer Gateway’ is set to ‘Disable’.Back to Top


Linksys WRT110N

Current Status: Untested
Recommendation: Untested
Comments:
This router has not been tested with VoIP services to determine compatibility. We recommend testing it prior to deploying it in a production environment.Back to Top


Linksys WRT310N

Current Status: Untested
Recommendation: Untested
Comments: This router has not been tested to determine compatibility with VoIP services. We recommend testing the router first before deploying it in a production environment.Back to Top


Linksys WRT350N

Current Status: Untested
Recommendation: Disable SIP ALG
Comments:
Firmware release 1.03.7 on Sept 14, 2007 introduced an internal change that disables SIP ALG within the software. SIP ALG interferes with VoIP services and we would suggest trying to upgrade to this firmware version or higher. This router has not been tested for VoIP compatibility.Back to Top


Linksys WRT610N

Current Status: Untested
Recommendation: Disable SIP ALG
Comments:
This router has not been tested for VoIP compatibility. If this router has a SIP ALG setting we would suggest disabling it.Back to Top


Thompson TG-585

Current Status: Untested
Recommendation: Untested
Comments:
The internal SIP service needs to be disabled. You will need to use a telnet (port 23) client to connect to the ip address of this router using putty or another telnet client. The username and password is the same as the one used to access the Web UI (default username Administrator). You will need to first login to the Web UI to and change the password before you can telnet into the device. Once connected to the router you will need to type these commands:connection unbind application=SIP port=5060
saveallBack to Top


Thompson Speedtouch ST-585

Current Status: Untested
Recommendation: Untested
Comments:
The internal SIP service needs to be disabled. You will need to use a telnet (port 23) client to connect to the ip address of this router using putty or another telnet client. The username and password is the same as the one used to access the Web UI (default username Administrator). You will need to first login to the Web UI to and change the password before you can telnet into the device . Once connected to the router you will need to type these commands:connection unbind application=SIP port=5060
saveallBack to Top


Trendnet Routers

Current Status: Untested
Recommendation: Untested
Comments:
This router has not been tested for compatibility with VoIP services. We recommend testing it extensively before using it in a production environment.Back to Top


List of all Devices, by Model

 

Back to Top

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Introduction

Before you deploy voice-over-IP or a Hosted PBX service in your office there are a few considerations you must first address.  Switching from traditional telephone service to voice-over-IP (VoIP) requires sufficient bandwidth, a proper switch and router, and a good battery backup solution to protect you from power failures.
The key voice-over-IP requirements discussed in this article are:

  • Bandwidth – Determining how much bandwidth you will need for voice-over-IP in your office is your first step.
  • The Router – Choosing a low quality or under performing router is a costly mistake which will degrade your call quality.
  • Quality of Service – You must decide whether voice traffic will be separated from regular internet users or if it will share the same network.
  • VoIP Equipment – There are many digital office phones, soft phones, headsets and telephone adapters on the market to choose from.
  • Power Failures – Voice over IP does not work when the power goes out so you should install a battery backup system and possibly a Power-over-Ethernet switch if your budget permits it.

 

 How much bandwidth do I need?

Voice over IP needs a certain amount of bandwidth in order to keep your conversations clear and free of disruptions.  Bandwidth is the amount of information which your internet connection can send and receive in a certain period of time.  Your first step should be to use an online speed test to find out what your maximum upload stream and download stream is.  We suggest you do this test using a fixed connection to the internet rather than using your wifi (wireless) connection to get accurate results.  Try to use numerous tests during different times of the day to get a good average of what you can expect from your internet connection.  Bandwidth is normally measured in kbps or kilobits per second. You will need to have a high speed (broadband) connection to use voice-over-IP.  A typical DSL connection will be rated at 600 kbps for the upload stream and 5000 kbps on the download stream.  You will notice that your upload stream is almost always smaller than your download stream which becomes your limiting factor for using VoIP service. Your next step is to determine how many people in your office are likely going to be using the phone at the same time.  For instance, having ten people on the phone will require ten times as much bandwidth as having one person on the phone.  Below is a chart which will help you calculate how many people can be on the phone at one time:

  • Ask your voice-over-IP service provider what audio codecs they offer as there is a trade off between audio quality and bandwidth usage…

    Full Quality Audio (G711 Codec)– Uses 87 kbps for each concurrent phone call
    Compressed Audio (G729 Codec)– Uses 33 kbps for each concurrent phone call

    So the calculation for a typical DSL connection would be:

    DSL connection:600 kbps upload / 5000 kbps download
    Gives us (Full Quality):600 kbps / 87 kbps = 6 concurrent calls
    Gives us (Compressed Quality):600 kbps / 33 kbps = 18 concurrent calls

Notice we used the upload bandwidth in our calculation as this is the limiting factor for voice-over-IP.  You also don’t want to push your connection to the limit as most cable and DSL connections do not have guarantees in terms of how much bandwidth they will deliver.  If you Internet connection drops in bandwidth at some point during the day you don’t want your call quality to be affected.  Other factors affecting voice-over-IP are the latency of your connection and how much packet loss there is on it.

Choosing a router

A router is the device that connects all your computers and network equipment to your Internet connection.  It is an often overlooked piece of the puzzle that can have a major impact on the success or failure of your voice-over-IP implementation.  There are many routers on the market, some are very cheap (less than $40) and others can cost you thousands of dollars.  There is nothing worse than putting a poor quality or underpowered router in your office which could cause an otherwise good VoIP installation to go bad.Your router needs to be powerful enough to handle the number of phones you will have in your office and should also work flawlessly with voice-over-IP equipment.  A good place to start when deciding on your router is to speak with your voice-over-IP service provider. We also recommend checking to make sure that your router is compatible with voice-over-IP services. Please review our Router Compatibility Guide.The following is a list items which will help you to determine whether your router is right for voice-over-IP:

  1. How many voice-over-IP phones will you be connecting to the router? The more phones you will be connecting, the more powerful the router needs to be. Don’t use a $40 router to run an office with 10 IP Telephones.
  2. Will your voice-over-IP phones have their own dedicated Internet connection? If not, a router with a quality of service (QoS) setting to prioritize voice traffic over regular traffic is an absliute must. Without QoS you will encounter poor quality telephone calls regularily.
  3. What other functions will the router need to perform? You might need your router to handle vpn connections, allow wifi (wireless) connections or perform other tasks.
  4. Make sure you can bridge your router to your modem. Routers that are not bridged can cause problems with voice-over-IP installations.
  5. Never use more than one router or nat gateway on the network at a time as this will cause problems for IP Telephones when they attempt to do NAT.
  6. Check our Router Compatibility Guide to make sure your router is compatibile.

It is always best to get a recommendation from your voice-over-IP service provider as some routers are known to perform very poorly with VoIP phones.

 

Quality of service

Call quality is a function of your network and the public internet. Some delays and network congestion cannot be avoided due to information traveling over the public internet while other types can be avoided. Good network design is critical to a stable and reliable voice-over-IP implementation.Quality of service (QoS) refers to the ability for your router to prioritize voice traffic (VoIP) differently than regular internet traffic on your network or the seperation of voice traffic.  Voice over ip is a real-time protocol which means that if information is lost or delayed it will result in a noticeable drop in call quality or a complete loss of it. Synptoms of network congestion include garbled audio, dropped calls and echo.   When setting up voice-over-IP in your office there are three possible ways handle voice traffic. Some customers report perfectly good results without any quality of service (especially in a small 1-2 person office) and others report worse results with quality of service enabled on their router as some routers do a poor job of implementing this. Generally speaking however the best way to deliver realiable voice-over-IP service is through a dedicated internet connection that is only used by the voice-over-IP equipment rather than sharing the internet with computers. Below are the different methods of doing quality of service:

  1. No QoS – Voice traffic and regular internet traffic in your office are sharing the same internet connection.  No prioritization of voice traffic over regular traffic is being performed and thus there is the high potential that voice quality could be degraded if there is insufficient bandwidth for both voice and regular traffic. Some customers experience very few problems using this method while others report a high frequency of poor quality calls, dropped calls and garbled voices. It all depends on how much network congestion your office has. Most internet connetions are more likely to be upload bound which generally results in people not being able to hear you, because all of your upload bandwidth is being consumed by something on your network.
  2. Router enabled QoS – Voice traffic and regular internet traffic in your office are sharing the same internet connection, but your router is able to distinguish between voice traffic and regular internet traffic and give the voice traffic a higher priority.  The problem with this method is that routers can only prioritize upload bandwidth which means your voice will be clear but the router cannot ensure that download bandwidth will be prioritized. If employees on your network are downloading often this will cause a noticible drop in call quality but this method is better than no quality of service. Some internet providers can prioritize the download bandwidth using TOS or COS methods from their end which will create an end to end quality of service solution. Most customers find that even prioritizing upload bandwidth for voice-over-IP offers a dramatic improvement in call quality because most internet connections are limited by their upload bandwidth and have lots of download bandwidth free.
  3. Separated Traffic – Voice traffic and regular internet traffic are separated onto two different internet connections and networks. This is especially critical for larger offices with 5 or more employess.  Voice traffic is carried on one internet connection and data from computers is carried on the other connection. In this case no prioritization is required by your router because voice traffic has its own dedicated internet connection.  This is the best way to ensure clear voice communications and the method we generally recommend customers whenever possible.

The method you decide on largely depends on how much bandwidth you have, what you are using your internet connection for besides voice-over-IP and the level of call quality desired.  Many offices report perfectly good results without using any QoS, while others find that it makes a major difference in the quality of their calls.

 

Choosing VoIP phones and equipment

Before deploying voice-over-IP in your office you will need to decide how each employee will be connected to your voice-over-IP provider.  There are many choices on the market today.

  • Digital IP Telephones – These types of phones look just like regular multi-line business telephones except that they connect directly to your internet connection using a network cable.
  • Soft Phones – A soft phone is a software program running on your computer that looks and feels just like a real telephone.  This requires you to purchase a USB headset which connects to your desktop or laptop so you can make and receive calls.
  • Wifi Phones – A wifi phone looks and feels very much like a regular cell phone except that it connects to your wireless router in the office.
  • Analog Telephone Adapters (ATA) – An ATA is a small box which connects to your router and allows you to plug in regular analog telephones so they can work with voice-over-IP.  ATAs are generally low cost alternatives to digital office phones and are easy to take with you when you travel.

 

Battery backup and Power-over-Ethernet

With voice-over-IP and most office telephone systems you must consider what happens when the power goes out.  For some offices this can be a regular occurrence and for others it might happen with a very low frequency.  Once of the things you will need to decide is whether or not you will install a battery backup system. Here are a few important terms your should know:

  • Power over Ethernet (PoE) – Is a technology that allows VoIP over ip telephones to be powered using regular network cables rather than power adapters which plug into the wall.  This has the advantage that you can power all the phones in your office from a single source and makes installing a battery backup unit much easier.
  • Uninterruptible Power Supply (UPS) – Is a device that powers your equipment when you lose power at the office.  The system has a built in battery which keeps your network devices operational when the power goes out.

The easiest way to protect your phone system from a power outage is to power all the phones using a Power-over-Ethernet switch that would normally be connected in the back room where your router and cable/DSL modem is located.  This has the advantage that all your phones are drawing power from a single source which you can backup using an uninterruptible power supply (UPS).  All you need to do is plug in your PoE switch, router, and DSL/cable modem into a sufficiently powerful UPS device so that when the power goes out all your phones remain up and running.

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How To Send A Fax Using ViewMyFax.com

1.) Open Internet Explorer and go to http://viewmyfax.com.

2.) Once the home page loads, you will be prompted to provide a user name and password to login to your account. Enter your user name and password into the appropriate fields and click the “Login” button, as shown below:

3.) Once logged in to your account, you will be able to navigate the Electronic Fax Customer Portal. This is where you will send faxes, view faxes, view fax status and organize faxes and contacts. To send a fax, find and press the button at the top of the portal page that says “Send a Fax”, as shown below:

4.) Once you press this button, you will be brought to a page that will ask you to submit a few different pieces of information needed in order to send the fax. The first piece of information needed will be the recipient’s fax number. Enter the number you are faxing to in the provided filed, then press the + button, as shown below,

5.) Pressing the + button will submit the fax number provided and will load the next page. On this page, you have the choice to provide an additional fax number for a different recipient, or you can press the “Continue” button to proceed to the next step of sending your fax. If you are faxing to the number you have already submitted alone, press the “Continue” button, as shown below:

6.) Pressing “Continue” will bring you to the next page where you will have the options to upload the files you want to fax, create a cover page for your fax and edit your files online. To create a cover page, click the button next to the option that says “Cover Page Creator”, as shown below:

7.) After pressing the button, you will be brought to the page where you will submit the information that will be reflected on your cover page. Fill out all applicable fields, then hit the “Submit” button, as shown below:

8.) Once your cover page is submitted, you will need to attach the files you wish to send to the fax recipient. To do this, press the button next to the option that says “Upload Your Own File”, as shown below:

9.) After clicking the button, you will be brought to a page where you will upload the files you wish to send. To do this, click on the “Browse” button next to the empty field on the page, as shown below:

10.) Pressing the “Browse” button will prompt an Explorer window to open. This is where you will search for and select the file you wish to send. Find the file you wish to attach to your fax and select it by either double clicking on the file or highlighting the file and pressing the “Open” button in the Explorer window, as shown below:

11.) Once you click the “Open” button, the blank field next to the “Browse” button in the View My Fax window will be populated with the name of the file you selected. Once this is done, press the “Upload File” button below the file field, as shown below:

12.) Once the file has been uploaded and attached to the fax you are sending, you will be given the option to attach additional files or send the fax. If you need to attach additional files, repeat steps 8-11. If you are ready to send the fax, find the “Send Fax” button to the right of the page and press it, as shown below:

13.) Once the “Send Fax” button has been pressed, the following message will be displayed:

14.) This means that the fax you just submitted is being sent. This usually takes a few minutes to complete. You can check the status of your fax by pressing the “Fax Status” button, as shown below:

15.) The Fax Status page will display a list of faxes you have sent and will indicate whether the fax is processing, successfully sent or failed to send. As long as the fax recipient’s fax number was submitted correctly and no errors occurred while processing, the fax should be sent without issue. Once the fax has been successfully sent, it will be flagged with a green check mark and a Successfully Sent Status, as shown below:

16.) As long as the above message is displayed after a few minutes of the fax processing, you have successfully sent a fax using ViewMyFax.com!

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WebRTC – short for Web Real-Time Communications – is a new standard that makes it possible to have a conversation from your Web browser.

Overview

WebRTC is a communications standard developed by the W3C in close cooperation with the RTCWeb standard developed by the IETF. RTCWeb functions at a lower protocol layer; WebRTC enables the embedding of this functionality in applications and websites.
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The festive season has arrived! Please take note that the Broadtela Office will be operating during reduced hours over the Christmas and New Year period.

For the full listing of Broadtela operating hours over the festive season please refer to the information below.
Customer Service

OPEN:

23rd December (9am – 5pm EST)
24th December (9am – 5pm EST)
27th December – 31st December (9am – 5pm EST)

CLOSED:

25th & 26th December 2014
1st January 2015

Business Team

Business Support and Business Sales will be closed on public holidays including: Christmas Day and New Years Day, as well as on weekends. The Business team will continue to operate during regular hours for the rest of the time.

Emergency support will continue to be available throughout the holiday season.
From 2nd January 2014, the Service Centre will return to standard operating hours of 8.30am to 5.30pm EST Monday to Friday.

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