While reading the phone’s manual I found out that you could in fact change the phone’s logo as opposed to what i thought before. I think I had firmware 6.39.x.x and on the manual that came with the CD with instrucctions on how to customize the logo.

Unfortunately the instructions talked about a program that did not come on the CD and a menu which did not exist on my phone’s configuration page. I decided to update the phone’s firmware and browse the net for that program.

As of January the 12th 2010 the latest firmware on yealink’s page is 6.41.0.10. The update added new languages and the menu I was missing, although it wasn’t called Phone -> Function -> Logo as the manual says but Phone -> Features -> Upload Logo *.tar.gz (The pixel < 132*64)

The name is pretty self explanatory but having tried a previous version of the firmware I had a different set of instructions. In those instructions there was no mention of a tar.gz file and I think the resolution was different too…

I also found the required program to convert the bmp file into a format understood by the phone but it didn’t have a .bat file that I needed to create the .tar.gz so I had to create it myself, following the correct folder structure which I didn’t really know about.

Anyway, here’s what you have to do:

1. Create your logo. If you’re working on a T26 like me the resolution should not exceed 132 x 64 pixels and it should only be in black and white. If you’re working with the T28 then the resolution goes up to 320 x 160 pixels and you can add two levels of gray (so 4 colors in total).
2. Download the conversion program. Don’t forget to run it in Vista compatibility mode if you’re using Windows 7.
3. Execute PictureExDemo.exe, add the bmp file and click on Convert:The program will create a.dob file  in the adv folder.
4. You can now edit the file advertising.index located in the adv folder. It contains the name of the logo file and the time in milliseconds it will be displayed on the screen.
5. Now comes the tricky part, the manual says you should execute a .bat file. But I didn’t have that file so you have to manually create the .tar.gz file with the correct folder hierarchy. First download 7zip or use tar if you can move your files to a linux PC. Then compress the entire adv folder which contains the advertising.index and .dob file (include the folder in the compression).
6. Browse to the phone’s IP, select Phone and then Features. At the end of the page you will be able to browse your PC and select the .tar.gz file:So browse for your file and click upload.
7. The file will be uploaded and if everything worked as supposed to you will see your new logo on your phone after 10 seconds or so of inactivity.

It’s a good feature not present in most IP phones, unfortunately the image is displayed as a screen saver and is not visible unless there is not activity on the phone for 10 seconds, in which case you also lose the time and date normally displayed on the screen for the duration configured on the advertising.index file.

It’s still a pretty nice feature though.

UPDATE: It seems the .tar.gz file route has been removed by firmware 6.42.0.30  on my T26P, now you can  upload directly the .dob picture. But no rotating banners I guess.

We normally only work with Polycom and Aastra phones because of the quality, versatility and the brand itself. But we have to understand that not all clients are ready to pay double the price of their phones when other resellers are offering them very cheap alternatives that allegedly do the same thing. We worked with Grandstream a while ago but decided we didn’t want to compromise support and stability in such a way for price.

So we decided to look at Yealink phones because we’ve heard some good things about them and there’s a good price difference that might allow us to compete in different markets.

Yealink has 4 main levels of SIP Phones, high level phones like the T28, medium level (T26) and entry level (T20, T22…), plus video phones which I don’t know much about. There are also very basic level phones (T9, T12…) but I’m now going to discuss those here.

I’m working right now with a T26P (the P is for PoE) and I’m waiting for a T20 next week so I can check it out too.

Yeastar T26P

There are some pretty obvious similarities with Cisco Phones, the menu also looks a lot like the one in Polycom phones. Keyboard is responsive (thought now as much as Polycom’s) but I the keys are a little jumpy.

Configuring the phone is as easy as configuring a Grandstream, it can all be made via web, even ring tones, BLF, background light… But it can also be centrally configured via TFTP or FTP. Another good thing is that all those phones support wideband codec G722, even entry level phones like the T20. They also have two Ethernet ports for switching, the second port can even work as a router which not that many phones do lately.

Audio quality on my first test was pretty good and the 13 configurable keys are very useful for some BLF or speed dials (or using the 3 accounts that can be configured on the phone). Transfering calls and conferencing is easily done via the menu screen and the phone has most of the normal and most useful keys (hold, mute, redial…).

On the other hand, the phone does not support EHS (Electronic Hook Switch) as most of the Polycoms do and it doesn’t seem I can change the background picture on the screen, maybe on a future firmware update.

I’ll continue to work on the phone but right now I’m pretty excited about it, I’ll begin to work on auto provisioning and XML applications next…

The Linksys WRP400 is a pretty good router, it doesn’t stray too far from the average WRT54G but it allows you to map external/internal ports (which is pretty nice if you need to have SSH access but don’t want to open port 22 or use some kind of VPN) AND it has 2 FXS ports.

The FXS addition is nice (although having extensions that register and have two-way audio when your Asterisk service is behind the WRP400 is pretty tricky and obscure) but they can cause a little bit of a problem if you tray to register SIP extensions that are anywhere on the Internet. Unless you change just a couple of things on the router’s configuration.

So, you configure the router’s public IP address, you redirect ports 5060 and 10000-20000 towards your Asterisk’s private IP, you configure sip_nat.conf (or sip_general_custom.conf) with the corresponding externip and required locanets, but when you try to register your remote phone (or softphone) you receive  a message “404 Not Found”. If you put some traffic analyzer in your remote office you’ll see that you are indeed receiving a 404 from your Asterisk’s public IP, from port 5060, to your original REGISTER message…

So, what happened? The thing is, even if you are redirecting UDP port 5060 to your Asterisk’s private IP, your router will keep intersecting SIP messages destined towards it because the “Line 1″ extension listens by default on port 5060 and will say that no one is supposed to register on it, it’s not a SIP registrar… So the messages never reach your Asterisk server and the phone never registers. You just need to log to your router, access the voice configuration menu (with admin privileges) and change “Line 1″ ‘s SIP Port to something different from 5060, 5070 for example (carefull with 5061 as it’s for Line 2)

The thing I see with the WRP400 is that it seems they cramed an SPA2102 inside the WRP54G’s firmware, it’s like there’s no connection between the two, separate users/passwords, configuration, even presentation… Which seems halfhearted but hey… It works. Just try avoiding putting your Asterisk server behind this router otherwise you’ll experience this kind of problems, and I haven’t talked about working with those FXS ports…

Anyway…

I’ve checked the phones and the server, it seems to be hardware related. It’s the first time this thing happens to one of those gateways which might not be very pretty (at least now they don’t use those brown cardboard boxes) but work quite alright, if you can tweak a couple of things here and there.

Anyway, I was thinking about resetting to factory defaults to help with the problem and saw this:

Are you sure??

Well I mean, you could click… but why would you??

I just thought it was funny…

Enter the Disconnect Tone detection in the SPA3102. It allows the SPA to listen to the line waiting for the busy tone that the telco sends you after the other party hanged up the phone.

How could you know if you’re having disconnection problems on Asterisk? Just make a call to your server through the line connected to the SPA (with you cellphone for example), answer it somehow (on another phone, an IVR, a queue…) and then hangup the initial call (hangup on the cellphone). Now start counting how many seconds asterisk keeps the channel active… If it’s more than 10 you probably have a problem…

Now, there are many ways of obtaining the parameters of your disconnect tone, for example:

1. Asking your telco. Results may vary depending on the telco…
2. Using an oscilloscope or some kind of signal processing hardware. Maybe a little expensive…
3. Searching for your tone in this tone database (you might have to register but it’s free). This could really help you understand all the parameters that you have to configure in the SPA (as you can format the tone as “Sipura/linksys”) You might either find the tone you’re looking for or at least you can look for one that is alike. This is not the most accurate way of doing it but you might get lucky…
4. Measuring the tone’s parameters manually. Check out this link. I think this is the most reliable way of doing it, specially if you use sigview (which is free for a limited time) as it will give you a very accurate cadence and frequence values.

There’s a very extensive guide on the link I provided and here are some screenshots of how sigview works, courtesy of my friend Luis:

Selecting a couple of tones for measurement

Measuring in detail the tone's duration

Measuring the tone's fequencies

With this method we went from hanging up after more than 20 seconds to less than 5… Before the changes, in some cases, the channels remained active for hours and hours and Asterisk kept recording those calls so we found recordings that were bigger than 400MB and our client’s hard drive was becoming full daily…

So, the links in this post might help you with an SPA3102 but the sigview measurements will help you with any kind of PSTN gateway like an AudioCodes… But configuring those is a far different beast.

Until next time.

My name is Juanhy, I’ve been working with VoIP and particularly with Asterisk for almost two years now and thought this could be a nice way of telling everybody about the problems and solutions I’ve found along the way.

For those completely lost, Asterisk is a great and extremely powerfull application which acts as a PBX (software PBX). It was created by Marc Spencer 10 years ago and allows communication between phones through many IP protocols including SIP, SCCP, IAX, H323… It can also connect to the outside world (otherwise it wouldn’t be much of a PBX) through PSTN lines, SIP trunks, E1/T1…

Working with Asterisk has been a great experience so far but given the huge amount of platforms, operators, protocols, and types of users, problems are bound to happen. There’s a great Asterisk community out there to help you and I figured this blog could allow me to provide some light on some obscure problems but also keep track of all the things that have happened/keep hapening to me so I won’t forget them all, I’ve been working on this for some time now but I keep learning new things every day!

So, here we go…