Thursday, October 29, 2009

Cisco SPA504G Review

Just got my hands on a Cisco SPA50G, this is the revised SPA942 (probably the most popular voip phone)

The SPA942 was a very nice voip phone - it offered extensive features at a very nice price. There were some things that needed improving. The SPA504 fixes most of the problems. The one problem I was hoping it would improve immensely was the speaker phone but I am out of luck here.

Although with recent firmware upgrades for the SPA942 the speaker phone became usable its still was not a "great" sounding speaker phone compared to Polycom phones. Seems that Cisco just can't figure out the speaker phone.

The SPA504G has the following improvements:

1) Support HD/g722 codec
2) Handset - heavier - more polished -sounds better - sounds deeper
3) Buttons - have more of tactile feel - don't feel cheap anymore
4) Speaker phone - louder (no improvement is sound quality)
5) Sidecar - all the SPA phones now support the new and old 32 button sidecar - you do not need a "receptionist" phone anymore - anyone of them can be turned into one - all have the AUX expansion
6) Configuration Interface is more polished - looks modern (same options though)
7) Line/nav buttons and MWI are smaller now
8) Unit is a little heavier - seems more solid

Overall a nice improvement - if you don't need g722 and louder speaker phone your better off getting an SPA942 while they are still available since there is such a huge price difference.

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.


video

Saturday, October 17, 2009

In Production Audiocodes MP-118 FXO Gateway Video

Mounted Audiocodes MP-118 FXO Gateway. Working with an Asterisk server and Cisco SPA942 phones. video

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. video

Sunday, October 11, 2009

Asterisk DTMF

Some things you need to know about Asterisk and DTMF.

Asterisk does have problems with DTMF - more in the 1.2 branch than the current 1.4 branch. 1.2 did not handle variable length DTMF.

No matter what you have done with your particular Asterisk setup there will be some issues that you cannot resolve!

You can however minimize what can go wrong with DTMF

In my experience there are two ways that DTMF can be somewhat reliable with Asterisk.

1. If you are using _only_ internal Analog (FXO) cards (no voip carriers or SIP gateways) than its best that you set everything (pbx and phones) to inband and set your phones to use a non compressed codec (711 ulaw)

2. if you are using other codecs (g729) and voip carriers than the best method is RFC2833. Don't mix different methods across your voip network. Your phones should use RFC2833, your gateways, and your PBX!

Most of the problems comes when people try all kinds of crazy setups.

This is not a 100% solution since DTMF problems can be caused by noisy (analog lines) and voip carriers not correctly passing your DTMF signals to upstream carriers.

You may be sending DTMF to your voip carrier using out of band DTMF and your carrier might be terminating your call via PRI. The DTMF signal will be lost!

Saturday, October 10, 2009

VoIP Caller ID Settings

One of the biggest headaches with VoIP is caller ID. There is some confusion on what you can do with caller id. For the most part its very simple.

For outgoing caller id (the person you are calling gets to see)

1. If you are using a gateway SIP to PSTN - Your PSTN carrier controls this - you cannot overwrite. It doesn't matter what you put in any caller id settings for your phone or gateway.

2. If you are using a VoIP Carrier (SIP trunk) - You may be able to change your caller id from either the SIP PBX or through a control panel interface provided by the carrier. Or in some cases the Carrier will need to change this manually for you.

It all depends how the carrier wants to do this. Some carriers are very strict about this because they don't want you spoofing your caller id which could result illegal activity and leave them liable for damages. eg. you could impersonate a popular charity and start calling for donations

For incoming caller id (what you see)

1. Generally any caller provided by PSTN is sent between first and second ring. If you are not seeing the caller id it may be due to:

a) You do not have Caller ID service - check with your carrier - i spent an hour once trying to get caller id working on a PBX because the client insisted he had caller id
b) Your PSTN TO SIP gateway or your FXO Analog card is answering the call too quickly. Increase the delay before the call is answered
c) You are using the wrong Caller ID detection method for your region - make sure you specify what part of the world you are trying to detect caller id

2. Caller ID provided by a SIP carrier should show up instantly if connected to a phone. if it is going through a gateway you may need to verify the caller id is passed as it is received. Gateways or PBX's allow you to manipulate the caller id before being passed to another gateway or end point.

Linksys VOIP Product Quality

One of my favorite voip brands has turned out be Linksys (now under Cisco). First small PBX I used was the LVS9000. It didn't turn out to be a great PBX but a lot of the associated products turned out very well.

The SPA942 is great general purpose phone. Can be used with many SIP phone systems including Asterisk. Also great for companies offering hosted PBX services.

The SPA962 with sidecar - great receptionist phone. The sidecar buttons can be configured for BLF, speed dial, and call pickup.

The SPA400 turned out be a very reliable 4 port analog gateway - although not advertised to work with Asterisk.

The PAP2T ATA's are great quality when used with a good analog phone - try the Aastra 9116 - MWI indicating light works, call waiting, hook flash, transfer button, and speaker phone.

The SPA8000 & SPA8800 are also great new products under the Small Business Pro line up.

And the legendary SPA3000 and SPA3101. Great combination of FXO,FXS, and router (SPA3102)

The SPA3000 lets you test a variety of PSTN / VOIP setups - truly one of the best hack devices!

The Truth About the Grandstream GXP2000

This phone is really great if you are learning Asterisk and want to test most of the features. It has many advanced features for a really low price. You can easily find the GXP2000 for around $85 at many online stores.

The first time I encountered this phone was at a clients site. The clients setup was done by someone who decided try their hand at VOIP. From my understanding they decided it was not worth the trouble and took a job in the Bahamas (idiots get all the breaks).

The client had numerous issues regarding dropped calls, echo, static/pop sounds, and general inconsistency with dialing certain numbers.

Many of the issues were to do with just sloppy configuring on the initial setup.

The only items that I could not fix were the static/pop sounds - I found it strange since all the phones were GXP2000 but only some of them were causing problems. A firmware update did not fix the problem phones.

I later purchased one of these phones for my own personal use and testing. I updated the firmware and used it for several months. To my surprise everything worked great. No pop or static sounds and even the speaker phone worked well.

My initial thought was that Grandstream fixed everything with a new firmware. At this point I thought this was is not a bad entry level phone for some clients looking to save some money. Even recommended the product when someone would ask.

Over time I figured out what the main problems were with the GXP2000:

1. Different versions of hardware - the more recent versions support an additional green LED instead of just a red LED for the line and speed dials buttons. The newer phones come with 2.5mm headset jack - the older ones come with bigger jack and 2.5mm adapter.
2. The speaker phone quality greatly varies from different batches. I have used versions which sounded great and other that were terrible
3. General quality is bad - over time these phones have a high failure rate. They will develop various problems. One issue is the static/pop sounds and the other is the "buzz" which comes and goes.
4. If the phone is configured to be a remote extension and at some point the phone is brought into the office and simply plugged into the local network it create all kinds of havoc on the call quality of the other phones. You need to do a "sip show peers "in asterisk command console. Look for any phones that have an IP address that points to the router (you need to fix those)

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

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.

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.

video

Thursday, October 8, 2009

AastraLink Pro 160

Just won myself an Aastralink Pro 160 PBX - Lucky Me! I haven't had much time to test it but here are few interesting things about the small box.

1. Completely solid state - no noise at all
2. Supports 6 FXO - usually you get only 4 in something in this price category
3. Only Supports Aastra IP Phones (but they do have a nice selection and prices)
4. Clean web Interface - would like to see more option but it covers the basic for a Small Biz PBX
5. All configuration is done from the web interface - except for the main "admin" phone which needs to be set up first.

More to come in later posts