Showing posts with label voip. Show all posts
Showing posts with label voip. Show all posts
Tuesday, August 2, 2016
Wednesday, July 27, 2016
Tuesday, February 25, 2014
Rogers One Service
Error
Please check your internet connection, and/or verify that your firewall security software is set to allow the web phone to establish a connection. If you are still experiencing difficulty, please contact technical support at 1888-Rogers1
error EB503
If you are getting the following error on your rogers one service when you try to login it's a DNS issue.
After wasting my time with rogers support i just decided to change the DNS to googles DNS servers (8.8.8.8) and see what happens. Like magic the service works. This issue might be on some of rogers DNS servers which have not been updated recently to reflect changes in www.rogersonenumber.ca portal.
Initially thought it was rogers just blocking my ip because I was able to establish a vpn tunnel and get the service to work. This probably worked because once I established the vpn tunnel I was using the remote dns settings.
Thank you rogers!
Please check your internet connection, and/or verify that your firewall security software is set to allow the web phone to establish a connection. If you are still experiencing difficulty, please contact technical support at 1888-Rogers1
error EB503
If you are getting the following error on your rogers one service when you try to login it's a DNS issue.
After wasting my time with rogers support i just decided to change the DNS to googles DNS servers (8.8.8.8) and see what happens. Like magic the service works. This issue might be on some of rogers DNS servers which have not been updated recently to reflect changes in www.rogersonenumber.ca portal.
Initially thought it was rogers just blocking my ip because I was able to establish a vpn tunnel and get the service to work. This probably worked because once I established the vpn tunnel I was using the remote dns settings.
Thank you rogers!
Thursday, January 28, 2010
Rogers Business Line Service
Some of you may be wondering how this service works. Roger unlike AllStream and Primus does not simply resell Bell analog/digital lines as their own.
What they do sell is "digital" business lines. There is no mention of voip or internet in their marketing.
But really what is it. Yes its digital but once the line is terminated on the business location it is an analog interface, just like your existing Bell lines.
Rogers places a modem but unlike their internet modem this has RJ11 ports to plug your analog phones or analog PBX into. The modem itself has 8 RJ11 ports and internal battery that acts like UPS for the unit. I know there is a battery in there because I unplugged the unit to do some cable management and to my surprise it remained powered on.
But is this really digital or just voip marketed with the Rogers touch. It definitely runs under the same coaxial cabling they use for internet. And with my luck I personally discovered this the day after installation when my area had intenet issue. Both the Rogers Business Internet and Rogers Business Lines failed. It was only for about 45 minutes but it didn't look good.
The service quality seems really good (when working) so what ever method they are using encode/decode from the customers site works really well. To me this looks like their "Home" phone but for business.
I will take a little video next time i am at the customers site.
Wednesday, October 21, 2009
Echo with Cisco SPA942 SPA962
First - there are many causes for echo and it's usually what causes the most confusion on how to solve the problem.
One of the simplest causes of echo with Linksys/Cisco devices is that users increase the handset volume to the max with the SPA942 and SPA962. This causes more of a feedback issues but users like to call it echo.
This is a little bit hard to describe but its almost like hearing someone faintly spiting in the handset - if you can imagine that. You can tell your users to lower the volume but you will get a complaint that its too low and they can't hear the person - which is a good point.
If your environment is strictly VOIP you don't have many options for increasing the volume before it hits the handset. If you are using a gateway such as the Audiocodes MP 114/118 you can increase the volume from the PSTN side of things.
Please use at least version 5.6 of the firmware - previous versions actually increased the volume before doing echo cancellation - which in turn created more echo issues.
One of the simplest causes of echo with Linksys/Cisco devices is that users increase the handset volume to the max with the SPA942 and SPA962. This causes more of a feedback issues but users like to call it echo.
This is a little bit hard to describe but its almost like hearing someone faintly spiting in the handset - if you can imagine that. You can tell your users to lower the volume but you will get a complaint that its too low and they can't hear the person - which is a good point.
If your environment is strictly VOIP you don't have many options for increasing the volume before it hits the handset. If you are using a gateway such as the Audiocodes MP 114/118 you can increase the volume from the PSTN side of things.
Please use at least version 5.6 of the firmware - previous versions actually increased the volume before doing echo cancellation - which in turn created more echo issues.
Friday, October 9, 2009
Asterisk SIP Channel Driver & DNS - it's broken!
One of the most frustrating things about Asterisk is that it can't handle simple problems. These are problems that you would think have been solved by now. I'm talking about DNS issues and the SIP channel driver.
You would think that if your internet connection went down that you would only lose your VOIP trunks or remote extensions. Well think again! When you lose the internet or better yet DNS access your Asterisk SIP Channel driver comes to a halt. That means that your local SIP extensions will also not work. In fact a lot of things just seem to halt until your internet or DNS is functioning again.
Why is this happening? Certain things should time out! My local SIP extensions and local SIP gateway should still work. I expect the internet to go down - I expect DNS resolution failures. I don't expect that after all this time the SIP channel driver is still broken.
I've looked in the Asterisk mailing list and this has been pointed out many times before - nobody wants to fix this.
I have developed ways around this that don't require code changes but this is something that should be fixed at the root cause.
You would think that if your internet connection went down that you would only lose your VOIP trunks or remote extensions. Well think again! When you lose the internet or better yet DNS access your Asterisk SIP Channel driver comes to a halt. That means that your local SIP extensions will also not work. In fact a lot of things just seem to halt until your internet or DNS is functioning again.
Why is this happening? Certain things should time out! My local SIP extensions and local SIP gateway should still work. I expect the internet to go down - I expect DNS resolution failures. I don't expect that after all this time the SIP channel driver is still broken.
I've looked in the Asterisk mailing list and this has been pointed out many times before - nobody wants to fix this.
I have developed ways around this that don't require code changes but this is something that should be fixed at the root cause.
Monday, August 17, 2009
A Bunch of Gateway Reviews
One of hardest things to find in VoIP hardware is a good low cost FXO gateway. The FXS gateways and ATA’s are easy to locate. Finding an equivalent FXO has been a never ending task. I have tested Patton, Audiocodes, Linksys/Cisco, and Grandstream gateways.
The Patton products are just way too complicated – someone needs to go back and put an easy mode on these things. The price is just a bit high for every situation I have encountered in deploying them.
The Audiocodes are not as expensive as the Patton products but they also have their own strengths and weakness. It takes a lot of effort to fully utilize or understand the full routing potential of these products. Once you master the configuration you can do all sorts of cool stuff. I have been able to deploy these products in a number of locations. They can work very well with Asterisk if you can match up the right version numbers (for both the gateway and Asterisk.)
The Linksys/Cisco SPA400 has been a real price gem for me. For locations that need a couple of FXO ports and money is an issue you can’t go wrong this device. Note that I have specifically tested a certain firmware version for my deployments. This product is intended to work with the LVS9000 but can be configured to work with Asterisk. The sound quality is good – not great. Good enough for the low end installations.
The Grandstream FXO gateways are a piece of junk – not worth a single penny because they cannot be deployed anywhere. I have tested with the current firmware and they continue to break what was working or cannot fix the long list of outstanding bugs. How many tries does it take guys to fix the caller-id issues? You Grandstream owners should know that you are missing many calls – hope you are not using these devices for your business.
Finally the best for last – I have purchased a Welltech 3804A because this Taiwanese company said they supported Asterisk. The price was right, the features were good, and the sound quality was actually very good. The main problem with this gateway is that it can’t detect a dial tone. This kind of limits the use for this anywhere with multiple lines and hunt group configurations.
Welltech’s latest gateways do support dial tone – I just wished I knew about this little issue prior. What gets me though is not dial tone aspect but that I was able to brick this product with configuration changes. At some point all the FXO ports went amber, I reset the device to factory defaults and it never came back! So much for confidence in this companies products.
I am currently testing the Cisco SPA8800 which is a 4 port FXS and 4 port FXO. Everything looks good so far but I can’t make a recommendation if this product is ready for any clients yet.
The Patton products are just way too complicated – someone needs to go back and put an easy mode on these things. The price is just a bit high for every situation I have encountered in deploying them.
The Audiocodes are not as expensive as the Patton products but they also have their own strengths and weakness. It takes a lot of effort to fully utilize or understand the full routing potential of these products. Once you master the configuration you can do all sorts of cool stuff. I have been able to deploy these products in a number of locations. They can work very well with Asterisk if you can match up the right version numbers (for both the gateway and Asterisk.)
The Linksys/Cisco SPA400 has been a real price gem for me. For locations that need a couple of FXO ports and money is an issue you can’t go wrong this device. Note that I have specifically tested a certain firmware version for my deployments. This product is intended to work with the LVS9000 but can be configured to work with Asterisk. The sound quality is good – not great. Good enough for the low end installations.
The Grandstream FXO gateways are a piece of junk – not worth a single penny because they cannot be deployed anywhere. I have tested with the current firmware and they continue to break what was working or cannot fix the long list of outstanding bugs. How many tries does it take guys to fix the caller-id issues? You Grandstream owners should know that you are missing many calls – hope you are not using these devices for your business.
Finally the best for last – I have purchased a Welltech 3804A because this Taiwanese company said they supported Asterisk. The price was right, the features were good, and the sound quality was actually very good. The main problem with this gateway is that it can’t detect a dial tone. This kind of limits the use for this anywhere with multiple lines and hunt group configurations.
Welltech’s latest gateways do support dial tone – I just wished I knew about this little issue prior. What gets me though is not dial tone aspect but that I was able to brick this product with configuration changes. At some point all the FXO ports went amber, I reset the device to factory defaults and it never came back! So much for confidence in this companies products.
I am currently testing the Cisco SPA8800 which is a 4 port FXS and 4 port FXO. Everything looks good so far but I can’t make a recommendation if this product is ready for any clients yet.
FreePBX User Speed Dial Functions
1. These are Phone System enabled - they have nothing to do with actual SIP Phone Directories in the Phones themselves. These features work with any type of Phone.
2. The user has to use their Phone to set these speed dial functions.
To Set a speed dial number Dial *75 from your phone. Follow the instructions. Enter speed dial number 1 to 99 is good, followed by #. Enter the number to dial as though you were actually dialing. eg 4161234567
In some cases you may need to dial 9 or 8 to get an outside number, so enter 94161234567
Once the system stores your speed dial number you can use the following format to dial using speed dial:
*0X or *0XX (where X or XX is the speed dial number you had previously configured.
^zero
Speed dial feature codes can be changed in the Feature Code Page.
2. The user has to use their Phone to set these speed dial functions.
To Set a speed dial number Dial *75 from your phone. Follow the instructions. Enter speed dial number 1 to 99 is good, followed by #. Enter the number to dial as though you were actually dialing. eg 4161234567
In some cases you may need to dial 9 or 8 to get an outside number, so enter 94161234567
Once the system stores your speed dial number you can use the following format to dial using speed dial:
*0X or *0XX (where X or XX is the speed dial number you had previously configured.
^zero
Speed dial feature codes can be changed in the Feature Code Page.
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