We got a new FoneBridge 2 and wanted to use with a Debian Etch 4 R2 installation.

The pre-built fonulator from redfone was not working, so we had to build it from source (thanks for your help Mark Warren = Redfone).

Here the steps to make the fonulator work with your Debian system:

First you need to install Debian Etch dependencies:

apt-get install libnet1-dev libpcap-dev libreadline5-dev libargtable2-dev

Downlod fonulator-sources-build-11.tar.gz:

wget http://www.intuitinnovations.com/asterisk/download/fonulator-sources-build-11.tar.gz

You can also download the file from the Red-Fone website.

Untar the file:

tar -xzvf fonulator-sources-build-11.tar.gz

Build LIBFB:

cd libfb

./configure
make
make install

Build Fonulator:

cd..

cd fonulator

./configure
make
make install

The fonulator tool will be located in /usr/local/bin directory.

Cheers
Daniel

Hi Guys

You can automatically Configure Grandstream-Phone via TFT-Server.

Download Manual: Auto-Config-Manual

Download Configurator Tool: Config-Tool

Download Template: Template

With best regards
Daniel

The TC400B is a bundle of the half-length, low-profile PCI-2.2 compliant TC400P base card and the TC400M voice processing module. The TC400B is designed to handle, in dedicated DSP resources, the complex codec translations for highly compressed audio as would otherwise be processed by Asterisk in software.

Asterisk, in software and with Digium G.729a licensing, is capable of transforming the G.729a codec into other codecs for the purposes of call origination or termination, bridging disparate calls, or VoIP to TDM connectivity. These transformations in software are very expensive, in terms of MIPS, and require a substantial amount of CPU time to accomplish. The TC400B not only relieves the CPU of this duty, freeing it up to handle other tasks or to complete additional call processing; but also provides Asterisk with the capability of bridging G.723.1 compressed audio into other formats, a capability not previously possible

The TC400B decompresses G.729a (8.0kbit) or G.723.1 (5.3kbit) into u-law or a-law; or, compresses u-law or a-law into G.729a (8.0kbit) or G.723.1 (5.3kbit). The TC400B is rated to handle up to 96 bi-directional G.729a transformations or 92 bi-directional G.723.1 transformations. The TC400B does not require additional licensing fees for the use of these codecs nor does it require the registration process associated with Digium's software-based G.729a codec licensing.

I will try to get one of this card to test it's performance.
This card should allow us to run more concurrent calls SIP to PSTN calls. This especially important in larger projects with many hundred users.

The card is prices at USD 1600.00

Digium launched a 8-port analog telephony interface card.

The launch of the TDM800P card (based on VoiceBus technology) and a Carrier Grade, Software-based Echo Canceller.

Based on Digium’s patent-pending VoiceBus technology, the TDM800P is a half-length, full-height 32-bit 33MHz PCI 2.2-compliant, 8 port modular analog telephony interface card.

The VoiceBus technology, as first introduced on Digium’s TDM2400P, allows the TDM800P to use an industry standard bus-mastering PCI interface, as found in millions of PCs worldwide, to maximize system compatibility and eliminate system conflicts.

Alongside this card, they’ve also introduced their first software-based echo canceller, the High Performance Echo Canceller (HPEC). The HPEC is carrier-grade and provides host-based Toll-Quality echo cancellation on a global scale. The HPEC solution uses an algorithm that is compliant with the industry standard for echo cancellers, G.168.

The cost will be arround: USD 755.00

Cheers
Daniel

Hi guys.

Marco told me that there is a new firmware for the Grandstream GXP-2000 phones. It's still a beta version but Bruce G. MacAloney (Vice President Grandstream Sales) is advising us to upgrade our phones with this new firmware.

The following issues / bugs are fixed in the Build 1.1.2.25 1/9/2007
:

• Fixed the “hissing” noise coming from the other parties handset
• Fixed VLAN not working
• Fixed display phonebook entry name as caller ID not working
correctly
• Fixed We always use the firmware server in the HTTP host header
• Fixed iLBC bad audio quality
• Fixed GXP-2000 incorrectly performed consultative transfer when you
switch line during a blind transfer
• Fixed GXP-2000 results in one way audio when a second incoming call
is not answered while the call is on hold
• Fixed we will not register any account if STUN is down or
misconfigured
• Fixed if network is down-then-up STUN IP checking gets fired
multiple series causing many STUN queries
• Restructured STUN/Registration to simplify account registration
management
• Fixed echo in 3WC problem reported in 1.1.2.23
• Fixed GXP-2000 under 3WC, second call info not displayed correctly
• Added support for BT-200 onhook-threshold.
• Added customizable delayed call forward wait time. Provision
parameter P139/P470/570/670, default is 20 seconds (as is
previously), allowed value 1-120; invalid values ignored.
• Added support for BT-200 delete called/caller entries via MUTE/DEL
key
• Changed NTP will retry 3 times if it receives no response from NTP
server; after that it will retry it after 1 minute. This also fixed
the NTP problem reported on the wiki site.
• Fixed we do not encode "#" in outgoing INVITE To URI

Download Firmware: Release_GXP2000-BT200_1.1.2.25

Download complete release notes: Release-Notes

So please if you have GXP-2000 phones with echo and noise problem pls. update the firmaware and monitor if the situation improves.

At this point i would not recommend to deploy the Beta for all GXP-2000. Just try it on selected "problem" phones.

I will ask Grandstream when the final will be released.

With best regards
Daniel