Reset Elastix Password to default

If you are using Elastix distro and you have forgotten your password you can reset it to the default palosanto password by executing the following command

sqlite3 /var/www/db/acl.db "update acl_user set md5_password='7a5210c173ea40c03205a5de7dcd4cb0' where id=1"

Note: the command is one line

No Perfect SIP phone

I've tested a number of SIP phones over a 2 year period and I still can't pick one single phone that solves most of the business requirements. I will list the top manufacturers with the main strength and weakness.

1. Grandstream - The company I love to hate. SIP's answer to DLink. Lots of features - good for testing. Terrible for customers. The stuff breaks or starts failing really quickly. Over time everything fails.

2. Linksys/Cisco SPA900/500 series. Good features all around. All the features work as advertised and quality is OK. The speaker phone could be much better and lack of BLF buttons keeps them from being deployed in many offices.

3. Polycom - Excellent speaker phone and call quality. Lack of message button and BLF buttons on 300 series phones keeps these phones from many solutions.

4. Snom - Good features, good quality, but real ugly. This is the volvo of phones. Practical but nothing to get excited about. Haven't looked into the 800 series phones but a huge improvement in the looks department

5. Aastra - Good mixture of features. Speaker phone good, call quality good, BLF buttons good. Variety of models from the traditional looking "office phones" to a more current look. The only real complaint about Aastra phones is the new models handset feels like it was designed for a little girl. The phones also have these funny little peg legs that can't position the phone vertical enough.

The one feature missing from all these phones with the exception of the Polycom 650/670 is end user call recording to usb keys.

This feature is asked over and over when talking to customers. Lawyers and Mortgage firms seem to want this feature more than anyone. If this feature was available I could probably sell more SIP based solutions as it provides a real reason and a business need to replace that aging key system.

FreePBX Phonebook & Directory Confusion

In FreePBX there is a Directory and a Phonebook. Similar functions but a different way of implementing. The biggest confusion comes from the way FreePBX has named everything.

Directory - Is accessed from the IVR by Selecting "Enable Directory" checkbox

When a caller is in the IVR they can press # to access the names directory.

The directory is built from the information that was entered during the creation of the extension and voicemail. For basic look up of extension names this is all you need to enable.

The option to choose searching by first,last, or first and last name is configured from the General Settings under Company Directory

Phonebook or Asterisk Phonebook or Phonebook Directory is a completely different option. The confusion comes from the naming.

There are two install modules - Phonebook and Phonebook Directory

Phonebook appears in the Tools menu as "Asterisk Phonebook" once installed

This is where you can create your own phonebook entries

Phonebook Directory is found in Destination options of FreePBX once installed. In the IVR you can select Phonebook Directory to search for names that were entered in the "Asterisk Phonebook" under the FreePBX tools menu.

This can be used to dial other names besides extension names.

*Note: The Asterisk Phonebook can also be used as a source for Caller ID Lookup Sources Module

Rogers does block VOIP!

In what took 6 days to resolve turned out to be Rogers Internet blocking certain VOIP traffic to certain IP ranges.

This was not a complete blockage of VOIP which makes it really difficult to detect. Rogers was blocking certain voip traffic from a specific carrier. Voip service worked just fine from another carrier.

This was not a home setup that was running peer to peer software constantly. This was a business account that uses voip, encrypted traffic for a financial application, and encrypted traffic of online backup data.

This makes me believe that Rogers does employ software to automatically detect peer to peer traffic (even on business service) and throttle/shape/block suspicious activity. In this case I believe the software was fooled and started blocking legit voip service.

Thank you Roger! I love you as much as Bell now!

Asterisk As A Voicemail Server for Legacy PBX Systems

Not coming from a Telephony background I didn't really understand how legacy PBX systems provided voicemail. It turns out voicemail is considered a separate solution. Many PBX companies left voicemail to third party companies.

To me this seems strange but hardware was much different in the early days and mass storage wasn't as cheap as it is today. In any case there many legacy PBX systems that are still running today.

One way Asterisk can benefit from this using Asterisk as replacement voicemail server. This is good because it gets rid of some of the problems of initially trying to sell Asterisk as a complete replacement to their existing system.

Here are some reasons why Asterisk will make a good voicemail server.

1. No expensive wiring or handsets to consider (client is happy)
2. Extends life of current system (client is really happy)
3. It will add additional features than the voicemail system its replacing, vm to email for example
4. Increase their voicemail storage
5. Client will be more open to move to a full Asterisk system once they are comfortable with it being a VM server first

For some systems it may not be possible to actually integrate Asterisk. But there are systems with SMDI and sip trunking support.

Default IP Phone & Router Passwords

One thing that I really need to do these days is create a master list of default passwords for the common routers and IP phones that I use. I just decided to connect my Polycom 330 phone after not using it for a while and I forgot the password. Well it's not admin admin or admin blank password. The user name for the web config is Polycom and the password is 456 - no way I'm going to keep remembering this.

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

Asterisk g722 with Cisco 504G

One of the new cool features of the Cisco 504G is support for wideband (hd) g722 codec. Note that g722 is only available with the 500 series spa phones. You spa 900 series owners are out of luck. Asterisk 1.4.x does not support g722 without patching, I've used the following patch and instructions with Asterisk 1.4.26.3 and it worked.

The one thing I noticed with the g722 Asterisk sound files is that the volume is much louder - I had to take down the volume half way on the spa504g. Everything sounded clearer but none of my gateways support g722. I know technically pots line would not provide hd quality audio but if the gateway supports g722 my Asterisk box would not need to transcode from 711 to g722. Everytime there is transcoding you lose quality, create latency, and slow down your system. These are all bad things and should be avoided.

Now I just need to configure another phone with g722 and do some test. Perhaps play some music and see if I can hear the entire range of sounds.

Aastralink 160 Pro Initial Setup

My initial thoughts about the Aastralink 160 Pro phone system is that it has a good number of features at a reasonable cost (around $850 from many online voip stores.) This is an embedded Asterisk system. Aastra does not hide this fact - although you would not know it if just looking at the box or gui interface.

The physical box is very small - no fan so it is very quiet. It is wall and rack mountable (brackets included for standard 19" rack)

6 fxo ports
2 fxs/fax/analog phone ports
1 lan
1 pc audio out (paging)
1 pc audio in (moh source)
additional interface for what could be used for door open switches or overhead paging

Setup

Normally network guys are accustomed to setting up devices by finding the default ip address of the device. We login and assign a static IP address. With this device its a little different:

1. You must have atleast one Aastra phone
2. Connect the Aastralink to your network and let it boot up completely (steady constant flashing green LED means it has booted OK)
3. Connect a supported Aastra IP phone - the phone also must be reset to factory default settings
4. Let the phone boot - it will upgrade/downgrade to the same firmware level that the Aastralink supports
5. Keep an eye on the phone - you will need to select an extension #, password, name, and email address. Note: You cannot skip this part - you cannot login to the Aastralink unless you setup that first phone which is automatically the "admin" phone
6. Once the admin phone is ready - get the IP of the phone (yes the phone)
7. Open a browser and enter the ip address of the phone - it will redirect you to the IP address of the Astralink
8. Use your extension # and password to login to the system as administrator - that's right! There is no "admin" username - the first extension is the "admin" account - enter the password you created for the extension

Once you are logged in you can configure additional phones and settings.

I will detail more info in follow up posting. This is not a review of the Aastralink 160 Pro, I will provide a review once I am done testing.

FreePBX default password

The default login and password for the FreePBX gui is admin admin

Note that if you are using a distro like trixbox, asterisknow, elastix, PIAF (PBX in a Flash) it is probably different.

Look in the /etc/amportal.conf

This is the default FreePBX config file - the passwords should be readable here.

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!

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.

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.

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

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.

In Production Audiocodes MP-118 FXO Gateway Video

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

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.

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!

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

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.

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

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.

FreePBX CallGroups, PickUpGroups, and Directed Call PickUp

The easiest feature is to use is Directed Call Pickup. If you hear a phone ringing and you know the the extension simply from any other phone dial **ext (where ext is the actual extension number)

The feature code can be changed to something other than ** or disabled completely. This feature code applies to all extensions. You cannot disable for some and enable for others. This has nothing to do with CallGroups and PickUpGroups. Configuration is separate.

More complicated is CallGroups and PickUpGroups. There are some general Rules:

1. Group Call Pickup feature code is *8 - can be changed and disabled/enabled in Feature Code Page. Refered to as Asterisk General Call Pickup
2. An extension can only belong to "one" CallGroup
3. An extension can belong to multiple PickUpGroups
4. CallGroups can range from 0 to 63
5. PickupGroups can be specified by placing the callgroup # they can pickup or multiple call group number separated by comma, eg, 1,2 - can pickup calls for groups 1 and 2 from their extension
6. CallGroups & PickupGroups are Controlled from the Extension Settings Page

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.