Showing posts with label cisco. Show all posts
Showing posts with label cisco. Show all posts

Friday, November 6, 2009

Side By Side Video - Cisco SPA504G & Linksys SPA942

The new Cisco SPA504G compared with the old Linksys SPA942

Roughly the same size - here is what is different

Handset is bigger and feels heavier on the SPA504G
Dial buttons are closer together on SPA504G
Nav button is smaller on SPA504G
Speaker holes are in a "round" pattern compared to a square pattern
Message Waiting Indicator is smaller on SPA504G

Wednesday, November 4, 2009

Cisco SPA504G Paging & Intercom

One of the reasons I wanted to get a Cisco 504G was to test the FreePBX/Asterisk Paging & Intercom module. With every new phone release its always important to test the feature set before deploying to clients.

The paging and intercom work the same way as the SPA942 models. Just need to enable "Auto Answer Page" and remove any vertical service codes that might conflict with FreePBX paging/intercom codes.

There is a new multicast feature that enables paging directly from the phone - need to look into this more.

The New Cisco Cares About Asterisk!

It wasn't too long ago that there was no "official" documentation that would suggest that Cisco products worked with Open Source Asterisk.

Now with Small Business Pro there is a sense that Cisco has put some effort in documenting and being more open to their products working with Asterisk.

There is a great deal of documentation that is really useful - in fact more than any other vendor. Documents can be found at https://www.myciscocommunity.com

You can find the following Asterisk documentation:

Asterisk: Configuring Cisco SPA5xx phones with the Web-UI
Asterisk: Zero-Touch Configuring Cisco SPA5xx phones
Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment
Interoperability information for Asterisk(R)-based Phone Systems
Asterisk: Configuring the Cisco SPA500S Attendant Console

You don't see this level of documentation from any of the following vendors that benefit from Asterisk.

1. Polycom - very little official documentation - even though many of their products work well with Asterisk - if it wasn't for the great Asterisk community many of their products wouldn't even be used with Asterisk systems

2. Grandstream - Just enough to get things working - very little documentation - but at least they acknowledge Asterisk - guessing that most of the IP phones that Grandstream sells go directly for Asterisk deployments. Grandstream is the equivalent of Dlink in the voip world. They sell lots of shitty product at low prices and available everywhere.

3. Aastra - have done a great deal with their xml and auto provisioning toolkit for systems like trixbox and elastix. No enough documenting if you want to configure things manually.

4. Snom - the stuff works - well documented on their wiki for Asterisk - just wish the basic phones weren't so ugly and expensive. The new 800 series look great though!

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.

Cisco SPA942 & SPA962 Multiple Calls with FreePBX

By default extensions created with FreePBX do not support "Call Waiting". This option must be enabled in order to receive "multiple" calls on phones that support more than one line. Nothing needs to be done on the SPA942 or SPA962 - "Call Waiting" must be set to "yes" per extension in FreePBX.

Tuesday, October 20, 2009

FreePBX & SPA942/962 Call Parking

1. Answer the call - talk to the person - say something like "Please hold on for a sec"
2. Press the Transfer (spa962) or Xfer (spa942) soft button
3. Enter 700 (default parking lot)
4. Listen to what number parking lot number the system has put the call in - it will be 701
5. Press Transfer/Xfer to complete the transfer to the parking lot
6. Locate the person who the call is for and tell them to dial 701 from any phone
7. If the call in the parking lot is not answered within the default time it will ring back to the extension that it was initially parked from

Monday, October 19, 2009

Cisco SPA8800 Video

This is just a small video of the SPA8800 configured with one analog phone and Asterisk. Notice the status lights on the Gateway. The silver box provides a number of ways to mount it - except no brackets - so no rackmount.

Its a very nice looking gateway with black plastic in the front. Testing with an Aastra 9116P - analog phone - although no benefit using "P" version as the gateway does not provide power using the 3,4 pair.

The MWI led works out of the box - you do not need to configure anything special. The speaker phone also works equally well. There is no auto answer option so receiving Intercom and Paging will not work with Asterisk. You can however initiate intercom and paging as there are many buttons that can be programmed for the feature codes. No BLF or any kind of line appearance. It is however a good "nortel" looking phone that a lot of clients expect phones to look like.


Saturday, October 17, 2009

Linksys/Cisco SPA962 with SPA932 Sidecar

Here is a little video of the Cisco/Linksys SPA962 with SPA932 sidecar all lid up. Also with a Panasonic 3.5mm headphone. The sidecar left buttons have been programmed for speed dial, BLF, and call pickup. The right side buttons have been programmed for intercom for each extension.

Saturday, October 10, 2009

Linksys/Cisco SPA941 - Don't buy

One of the first SIP phones I ever worked with was the Linksys SPA941. This was a great for learning but with newer firmware releases this phone became worse instead of better. If you find a great deal on it and want to use for learning its still ok. It's not okay to deploy for new customers - the SPA942 is better choice.

The SPA941 is very similar to the SPA942. The only differences are:

1. No Backlit display
2. No PoE - You have to use the AC adapter (it is included)
3. No Dual switch (you cannot plug a computer to share network drop)
4. Terrible speaker phone
5. Low volume

The SPA942 shared the Low Volume and bad speaker phone with some of the releases. With the current firmware (as of this posting) the speaker phone and low volume for the SPA942 have been improved - but it still not perfect. You are not going to get Polycom speaker phone quality out of these devices - but you are also not paying anywhere near the amount for similar features.

For testing the SPA941 is good, for deployment use SPA942.

FreePBX Feature Codes & Cisco/Linksys SPA942 - SPA962

FreePBX provides many features but out of the box some of these will not work without some changes to the default Cisco/Lisksys SPA phones.

1. Your Dial plan should allow you to dial * plus at least 2 digits.
a) Can dial things like *97 or *98 for voicemail access
b) *100 to transfer calls to voicemail or dial directly into someone's voicemail without calling them

eg. (*xx.|xx.)


2. Your dial plan should also allow ** plus atl east 2 digits - this is used for Directed call pickup. Dialing **100 from a different extension will answer a ringing phone at extension 100.

eg. (*xx.|**xx.|xx.)

3. For Paging and Intercom to work you need to set "Auto Answer Page" to yes

4. Delete any vertical service codes that conflict with your assigned FreePBX feature codes

Friday, October 9, 2009

Cisco/Linksys SPA942 & SPA962 Anonymous Caller ID issue

This may be a common problem across all SPA devices from Cisco/Linksys but I have only tested with the SPA942/962 IP phones. For most users this would not even be an issue.

The problem is the way that the SPA deals with anonymous calls. Instead of passing the actual "anonymous" caller id received from the carrier (or gateway) it transforms it to "Anonymous Caller" which may have been a nice touch by one of their engineers to give it a nice looking appearance but it truly messes things up in a scenario where a multi-tenant phone system is used.

In one of deployments we had configured FreePBX to handle three different companies. We didn't do anything special with FreePBX except change the Caller ID prefix for each companies incoming lines. eg. Company A would receive calls with the caller id "A:XYZ Comp <1234567890>"

All this worked great and the receptionist knew how to answer the calls for the different companies. This unfortunately didn't work with anonymous calls. So instead of displaying "A:anonymous" it displayed "Anonymous Caller" - and the receptionist had to no clue for which company the call was for.

To determine that this was truly was a problem with the SPA phones we tested using the following:

1. x-lite - showed the correct caller id for anonymous with the company prefix
2. snom phone - also worked
3. Even a GXP2000 from Grandstream worked - these phones have other issues but the caller id worked

Tried getting this resolved with Cisco - but didn't get far on their public community board.

I am not a Cisco partner so I can't submit a bug report

I did submit as much info to their myciscocommunity.com site as i had time for.

I didn't have time to continue with the additional requests for more info.

At this point I think submitted enough info to reproduce the problem and we had already resolved the problem with an an Asterisk solution - we completely changed all the anonymous calls to "Unknown" before they hit SPA phones.

Just putting this out for anyone who may run into the same issue.

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.