Asterisk and Budgetone 102

Hello All,

I have the following configuration, 3 sites all connected by vpns to each other. In the main office I have an Asterisk 2 box with a budgetone 102, this works perfectly here, voice quality is excellent. In the two remote offices i have a budgetone 102 installed and connected to the asterisk box in the main office. Everything is working fine but the call quality if very bad to the remote offices. The voice is choopy, scratchy and has echo. Does anyone know how I can fix this.

Trevor lewis

You haven’t given any information about the VPNs. A VPN on its own is meaningless - what sort of internet connection are they running over?

What codecs are you using? If you’re using g711, then that would explain it. You want to use a low-bandwidth codec - e.g. ILBC. The BudgeTone supports the ILBC codec, which gives decent quality with low bandwidth usage.

The connections are as follows.

All ADSL Broadband

Main = 1Mb/128
Remote 1 = 2Mb/256
Remote 2 = 1Mb/128

The vpns are all using dlink dfl-700 devices.

The codecs are in the default order that the budgetone selects them, I have not changed this.


[quote=“trevor lewis”]Main = 1Mb/128
Remote 1 = 2Mb/256
Remote 2 = 1Mb/128
the 128kbps uplink from Main is not very big. What else is using that connection? Have you got any QoS queuing happening?

That’s not necessarily what determines what codec you’re using. Have you got canreinvite set to yes or no in sip.conf? What codecs are configured in sip.conf?

But, really, most importantly, what codecs are the phones actually using during a call?

Canreinvite is set to no. No Qos queuing. Have diallow = allow = in AMP for each of the extensions.


I don’t know AMP, so that doesn’t mean anything to me, i’m afraid.

Anyway. To try and solve your problem, make sure you’re using a low bandwidth codec. If you do that and you’ve still got the problem, post here again with the rest of the information i asked for.

OK the latest update.

All calls that either originate or terminate at the main office ( the one where the asterisk box is located ) are now perfect, sound quality is excellent. However calls between the 2 brach offices are abysmal, very bad echo, choppy sound etc. Anyone got any ideas on this.

The codecs I have configured are ulaw, alaw and ilbc in that order.

How do I find out what codec is in use during a call?


you can watch it setup on the asterisk console. You should configure the codecs in the order (both on the phone and on *) of ilbc, ulaw, alaw. You want to prefer ilbc due to MUCH lower bandwidth usage.

The problem is the bandwidth at the main office, only 128k upload. Each ulaw/alaw call uses 64kbit in each direction, so if you have two of those, you peak out. You should try to increase that if you can, and consider getting some kind of QoS. Also make sure all the phones’ sip.conf entries have canreinvite=yes (or dont have =no) unless this breaks something, which it shouldn’t. That will make the phones in the offices talk directly to each other, which will save a bunch of bandwidth on your side.

An alternative to trying to follow the messages when the call is set up is to use the following command:

sip show channels

That will tell you what codecs are in use for each channel.

If you’re trying to use Asterisk in a business environment, you really do need to know a lot more about how it works. You need to do some reading - starting with this book: … +Telephony

You can download it for free, but please buy a copy (as you’re using it for commercial purposes) to support the author and the publisher and make sure that new editions come out in the future.

Then you need to read your way through as much of this wiki as possible:

and use it to refer to when you are trying to solve problems.

When you’ve done that, you will be starting to learn about Asterisk.

You are currently groping in the dark - and that’s not a good look in a business environment!

Thanks to all for the pointers, its now working.