SIP VoIP - Everything SIP

Grandstream GXP2000

2010-02-17T21:08:00.000-08:00

I have roughly 30 of these phones in various client sites. My worst fears are coming true - these phones are failing at greater rates than ever. Grandstream has taught me some very important lessons. First - don't buy anything Grandstream no matter how tempting that low price is. Second - spend more time testing. Finally - If had spent more time viewing issues in Grandstreams own support forums and viewing their release notes for new firmware I probably would have not chosen to use these phones. The only thing I can do now is warn others of what issues they have to look forward to by using these phones.

The symptoms seem to be consistent with the ones that are failing.

These include:

1. Snap - Crackle - Pop sounds

2. Buzzing sounds

3. Decreasing volume - starts normal - but volume starts to go low to very low

4. Failing buttons - they stick or stop responding

Note: These phones work well in the beginning but over time they start acting up.

Rogers Business Line Service

2010-01-28T15:42:00.000-08:00

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.

Why G729 is the best broadband codec!

2010-01-19T19:43:00.000-08:00

For some reason people don't understand when to use G729 codec. Lets make this really simple, here is a list of scenarios of when to use G729.

1. Limited internet speed - G729 allows you to maximize the amount of users that can simultaneously make calls. If you are using something like ulaw/711 you will be lowering the mount of simultaneous users and creating quality issues.

2. Shared Internet. You are using your internet for both voice and data. G729 allows you to lower the amount of bandwidth required for voice. You voice bits will get out/in quicker because less data to deal with. Your router also needs to be configured for QoS.

3. Your PBX is hosted. Even calls between extension to extension have to make a round trip. Again keep the amount of data that goes out/in small.

4. Connecting to gateways. Do the math - if you have a 10/100 connection from PBX to a gateway and your data between PBX and gateway is saturating the link you are going to have quality issues. Using G729 reduces the amount of data being sent between the link.

5. Using a good quality phone is also important with G729. It can reduce noise/background sounds to further reduce the amount of data being sent. My own personal tests indicate that using a Polycom 33o with G729 end to end creates a better call than using a Grandstream GXP2000 using ulaw/711 end to end on ideal broadband conditions.

PS. Don't buy Grandstream products - just used as example here. Grandstream products have quality issues/high failure rates.

Aastra space bug bites me again!

2010-01-14T22:55:00.000-08:00

Aastra has some nice phones and I needed to demo on of these for a client the other day.

I punched in some settings and copied others using copy and paste - big mistake!

With my copy and paste I had an extra character which was a space.

It looks like Aastra phones don't like the extra space in a field where they expect an ip address.

It took me about 30 minutes but I eventually found the extra space - removed and everything registered nicely. What bothers me is that this is the second time I've managed to make the same mistake. I am betting others are going through this exact same thing - hope it helps someone.

SIP Carriers and ISP's still suck in 2010!

2010-01-13T20:08:00.000-08:00

I've been in the voip/pbx business now for over three years and the SIP carriers and ISP's are providing the same crap service they were three years ago.

I blame both the ISP's and SIP carriers - but mostly the ISP's!

Here is Ontario we have two key players providing internet. Bell and Rogers - they both provide crap service in their own special ways.

1. Rogers cable likes to mess around with traffic shaping screwing with SIP/voip and p2p programs.

2. Bell has such old copper cabling for DSL they don't even have to try. When it rains major parts of the city impacted with degraded DSL services. Bell is not interested in repairing anything unless they have to.

SIP carriers have to rely on ISP's for good service so they can deliver their product. Unfortunately many of them do not invest in the proper equipment and good upper tier carriers to provide that good product.

Combine that with bad ISP service and you have a recipe for lousy SIP service!

I never have trusted SIP carriers and for the few clients that I regretfully setup with SIP crriers I will be removing in 2010!

Until we have excellent internet service and excellent SIP carriers i won't be going setting up anyone with SIP carriers. Good is not good enough for SIP carrier service.

Please note that I believe the future is SIP and VOIP protocols and I will these technologies in phones and gateways. I just won't be using over the internet. voip does not equal internet. it's voice over "internet protocol" - we are using the this in phones and gateways over standard analog/pri lines. We do not need to use the internet - its just the protocol!

Reset Elastix Password to default

2009-12-31T19:08:00.000-08:00

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

2009-12-29T21:39:00.000-08:00

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

2009-12-14T22:58:00.000-08:00

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!

2009-12-01T17:48:00.000-08:00

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

2009-12-01T16:44:00.000-08:00

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

2009-11-24T21:30:00.000-08:00

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

2009-11-06T16:50:00.001-08:00

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

2009-11-05T20:26:00.000-08:00

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

2009-11-05T13:20:00.000-08:00

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

2009-11-04T22:04:00.000-08:00

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

2009-11-04T20:37:00.000-08:00

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!

2009-11-04T18:59:00.000-08:00

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

2009-10-29T18:45:00.000-07:00

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

2009-10-21T19:06:00.000-07:00

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

2009-10-21T16:24:00.000-07:00

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

2009-10-20T18:38:00.000-07:00

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

2009-10-19T12:00:00.000-07:00

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

2009-10-17T20:23:00.000-07:00

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

Linksys/Cisco SPA962 with SPA932 Sidecar

2009-10-17T19:52:00.000-07:00

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

2009-10-11T09:37:00.000-07:00

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!