Archive for the ‘VoIP’ Category.

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Avaya Playtime

March 7, 2009, 12:27 am

OK, so I got to borrow an Avaya 4610SW from work, cos it would be faster to play with firmware via TFTP on my home network than it would be when using the business network. These are some notes…

turn off 802.1q – my network doesn’t do tagging,

Grab the latest firmware from ftp://ftp.avaya.com/incoming/Up1cku9/tsoweb/ip_telephone/071008/ – in this case it was plain old 46xxH323_071008.zip, and extract into TFTP root, or provide symlink

Turn on the phone, and it collects the following from TFTP…

Mar  6 23:01:59 senior in.tftpd[2619]: RRQ from 172.24.32.29 filename 46xxupgrade.scr
Mar  6 23:01:59 senior in.tftpd[2620]: RRQ from 172.24.32.29 filename 46xxsettings.txt
Mar  6 23:02:05 senior in.tftpd[2622]: RRQ from 172.24.32.29 filename b10d01b2_9.bin

That’ll be the bootloader getting downloaded (or Bootapp, as Avaya like to call it), then saved to flash. It then rebooted, updated with BIG SCARY WARNINGS, rebooted again, and then it picked up these…

Mar  6 23:05:02 senior in.tftpd[2666]: RRQ from 172.24.32.29 filename 46xxupgrade.scr
Mar  6 23:05:03 senior in.tftpd[2674]: RRQ from 172.24.32.29 filename 46xxsettings.txt
Mar  6 23:05:06 senior in.tftpd[2707]: RRQ from 172.24.32.29 filename a10d01b2_9.bin

which were saved to flash again, rebooted, Updated, rebooted, and then off it went.

So, upgrades went smoothly, whilst staying on the H323 featureset. Point proven for the boss, and should be easy to deploy the new firmware after some more testing. I now want to play with SIP…

The above file includes SIP in the downloaded zip file, but APPARENTLY… the SIP firmware is chosen based on the SIG parameter in 46xxsettings.txt. So, essentially, to enable SIP, you just have to enter SET SIG 2 into 46xxsettings.txt and let 46xxupgrade.scr work the magic. However, in my experience, this failed miserably. The phone just carried on blindly using the currently loaded firmware. To manually change the phone into SIP mode, you have to hit Mute, then dial SIG (744), and then hit #. It will then prompt you to change the SIG value, at which point when you save it, it will pick up the new firmware from the TFTP server.

After a short delay whilst it loads s10d01b2_2_2.bin, it’s off and running. As long as you’ve provided some basic SIP details in 46xxsettings.txt, it will prompt for a SIP username and password after a short delay. I believe this can also be automated based on MAC address, according to some conjecture and hearsay on the internet…

Category: Geek, VoIP  |  Comment

Asterisk Jabber Call Notifications

January 5, 2009, 8:06 pm

The other evening I busied myself with tweaking Asterisk to do some more geeky things. One such item was where I configured it to send a message to my Jabber account every time there was a call to a particular extension. Great for call logging, integration, and general user friendliness. If I was so inclined, it could message a Yahoo, MSN, ICQ, or AOL account through the Jabber platform to inform me of a new call. How rather flexible.

Category: Asterisk, Computers, Geek, Technology, VoIP  |  Comment

A Reason to Hate Cisco

January 4, 2009, 10:13 pm

Last week I ‘accidentally’ bought a Cisco 7911 phone on Ebay. My bid was low, the device was missing a few bits, but I still thought I wouldn’t get it.

Suffice to say… I’ve just written this piece on how to load the SIP firmware onto a Cisco 7911 phone. The procedures are covered elsewhere, but I thought I could compress it into a format that’s slightly easier to understand. Hope it makes sense. :-)

Whilst I’m at it, if anyone can email me a working SIP based cnf.xml file, it would be greatly appreciated. Getting the firmware onto the phone is easy… configuring it for Asterisk seems to be a completely different ballgame.

Category: Asterisk, Computers, VoIP  |  1 Comment

Location Location Location

September 8, 2008, 9:47 pm

[ad]
It’s time, and Ofcom are beginning to enforce the latest revision to General Condition 4, whereupon VoIP providers must, where technically feasible, provide location information for their VoIP users. In order to aid emergency services, they will transmit details of your fixed line location to the recipient when you make an emergency call.

Obviously this is quite an important thing, and not something you want to get wrong. So take a few moments now to ensure that the details held by your VoIP provider are the correct details.

Category: Internet, VoIP  |  Comment

Chan_Skype

September 9, 2007, 12:07 am

Dead or alive? It’s in a sorry state – it was a bit flaky to start with, and now won’t work at all. They only support Ubuntu Dapper, Skype only provide for Ubuntu Feisty – I see a conflict of support interests. They don’t seem to be interested in replying to any emails enquiring about Feisty support too.

Time to move onto another SIP to Skype solution :-(

Category: Linux, VoIP  |  Comment

SIP URI Support, and Siemens

August 2, 2007, 11:30 am

A while back, I wrote about the way I see VoIP going, and asked upon Siemens to provide support for dialling a SIP URI address instead of just numeric addresses.

Siemens responded with information stating that they will pass the information on to the product manager, who will hopefully respond appropriately. Unfortunately no such response has been forthcoming. Another email has been sent requesting an update to the support case, and I’ll wait with baited breath.

In the meantime, the more people that show a desire for SIP URI dialling, the more likely it is to be incorporated into future releases of the software. If you have the time or the need, email Siemens at support.gigaset@siemens.com to show your support.

Category: Technology, VoIP  |  2 Comments

Dear Siemens

May 23, 2007, 9:46 am

I have a Siemens C460IP phone at home, and so far I’m very happy with it. Very easy to use, lovely integration with Asterisk, and ideal for general usage around the flat. However, one thing bugs me, and that’s the lack of alphanumeric addressing capability in the phone book or dialing screen. Siemens have been rather good with issuing firmware updates to the system (all done through the handset – how neat!), so it shouldn’t be that hard to get the system updated. This is all to do with what I described in my previous posts about SIP addressing and the future of VoIP

So, in an attempt to get this functionality, and in good tradition, I submitted a feature request to them. Here it is for posterity, and I’ll see if we get any mileage out of it…

Hi,

I would like to submit a feature request to your engineering team
for the C460IP phone (which I have and so far am very happy with). It
would be a simple addition to allow alphanumeric entries into the
phonebook or dialling screen. At the moment it only allows numerical
addresses and IP addresses. Allowing alphabetic characters
would allow the user to call SIP addresses anywhere in the world
without having to rely on a number lookup system. It would allow
similar functionality to your Gigaset.net system, but be global
and open, just like the SIP support in the phone.

Please see http://lodge.glasgownet.com/blog/2007/05/12/more-voip/ for more reasoning behind this request.

I look forward to hearing from you.

Kyle

***update***

Siemens have gotten back to me already, rather quickly. Along with the standard blurb about customer service centers, I get this…

We forwarded your feedback and your request to our product manager. Thank you very much for your feedback.

Unfortunately at the moment the phone will not allow alphanumeric entries in to the phone as you pointed out. We have not received any information about this changing in the future with this phone but we will send you an answer when we hear back from the product manager.

Category: Internet, Technology, VoIP  |  1 Comment

More VoIP

May 12, 2007, 2:25 pm

To follow on from my previous post, I’ve gone about setting up SIP addressing like this for a very good reason. Other than the eventual demise of telephone numbers, I believe that larger VoIP networks will increasingly grow in size, whilst marginalizing the smaller networks. This is what happened to free email provides many years ago, and it’ll happen again to VoIP providers. Continue reading ‘More VoIP’ »

Category: Asterisk, Internet, Technology, VoIP  |  2 Comments

VoIP

May 12, 2007, 1:52 pm

Lately I’ve been poking around Asterisk and related VoIP stuff, and generally trying to make it work. I’ve (not) read my fair share of documentation, and it still seems pretty patchy. I’ve signed up for a free 0141 number from Sipgate and it seems mostly good. However, SIP addressing is a big itch that I’ve recently scratched… Continue reading ‘VoIP’ »

Category: Asterisk, Internet, Technology, VoIP  |  2 Comments

More Zap Bugs

April 2, 2007, 8:15 pm

In an astounding piece of ignorance, the Digium bug tracking managers have closed off bugs 8763 and 9081. Both bugs are clearly reproducable, clearly showstoppers, and clearly a big problem, yet they refer the original reporter to Digium support and close off the bug as “No change required”.

Now, if you have a bit of hardware used by millions the world over, and your software randomly hangs on it without a word of warning and then refuses to work until it is unloaded and reloaded, then it really should be investigated a bit more closely. Digium & Open Source… just like clowns but without the fancy dress.

Category: Asterisk, Bugs, Linux, Things that annoy Kyle, VoIP  |  Comment
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