app_swift v1.6.1 released for Asterisk 1.6

found on list today..

 

Thought I'd let everyone know I've released app_swift v1.6.1 which is
entirely based off of Will Orton's work he's placed in the public
domain.

Works great with Asterisk v1.6.0-beta7.1.

In any case, can be downloaded from my site at:

http://www.darrensessions.com

Go easy on me, this is my first release of anything.

Thanks,

- Darren

 

also this follow up

 

I've written an AGI wrapper for it as well, in case you don't want to re-compile
to support.

http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper

Updated Ringtone for Polycom IP500 tutorial

wayyy back in the day we wrote an article about ringtones for polycom ip500/ip600 phones, which apparently is still quite popular on the internet.

We received this from one of our readers today, as an addition to the previous article. Thanks Chris!

I found your blog post on Polycom compatible ringtones recently. It is one of the top results when searching for such information.

I would like to share with you and your readers a classic old ringtone I sampled from one of our company's old AT&T MLS-12 phones. I would have post a comment, but I could not find how to.

The file has been tested and is Polycom compatible, having been built according to the specifications you provided.

Thanks, Chris Olson

You can download the updated ringtone by following this link to the downloadable ringtone

Visual Dialplan v1.3 is released

Now with integrated SSH client, include capability, custom component and more

If you'd like to download APSTel Visual Dialplan Professional or Standard edition to give it a try yourself, please use these links.

Free APSTel Visual Dialplan Download from RatemyDialplan.com

• Windows Visual Dialplan - Std and Pro
• APSTel Visual Dialplan Professional - Windows
• APSTel Visual Dialplan Standard - Windows

• Linux Visual Dialplan - Std and Pro
• APSTel Visual Dialplan Professional - Linux
• APSTel Visual Dialplan Standard - Linux

With the latest Visual Dialplan you can deploy Asterisk dialplan with a single press of a button using integrated SSH client directly from your Windows desktop to remote Asterisk box, or include your existing Asterisk dialplan code created in a traditional way into the Visual Dialplan.

You can even continue developing part of the code in a traditional way, or reuse already developed dialplan code using Custom Component.

• Visual Dialplan v1.3 feature highlights:
• One button deployment with integrated SSH client
• Custom component that facilitates reuse of existing dialplan code
• Dialplan include functionality
• Context or Macro include functionality
• Personalize Visual Dialplan behavior using preferences

and more...

We invite Visual Dialplan v1.x users to take advantage of free upgrade to Visual Dialplan v1.3 and to download the latest Visual Dialplan.

Integrics releases Enswitch 2.9

Integrics are pleased to announce the release of Enswitch 2.9, a
complete integrated solution for commercial telephony services such as:

* Full featured hosted PBX.
* ITSP (Internet Telephony Service Provider).
* VoIP for WiFi providers.
* Toll-free and number translation services.
* Calling cards.

Enswitch is in production today with carriers worldwide on systems from
hundreds of users on single machines to over one hundred thousand users
on redundant/failover clusters. A list of new features can be found at:

http://integrics.com/products/enswitch/guides/latest/en/presales/new/

More links, including a full list of features and a working demo of the
web interface, are at:

http://integrics.com/products/enswitch/

Zaptel 1.2.25 and 1.4.10 released

The Asterisk.org development team has announced the release of Zaptel
versions 1.2.25 and 1.4.10. These releases contain many bug fixes as
well as performance enhancements.

A couple of the more major changes include: modifications to the
wctdm24xxp and wcte12xp drivers to increase interrupt latency
resilience, numerous bug fixes and updates to the xpp drivers, as well
as some Makefile updates. For further details and a more complete list
see the respective Changelog files.

Both releases are available as a tarball as well as a patch against the
previous release. They are available for download from downloads.digium.com.

Thank you for your support!

OpenVox First Announces Built-in Global CID Support for Open Source Analog Cards!

Hi All,

Just FYI.

OpenVox has now released new device driver for A800P/A1200P that has
built-in global CID support including the British Telecom, Brazil,
Taiwan CID format etc. This new improvement enlarges the applicability
of asterisk which enables the users especially from South America,
Asia-Pacific and North Europe to use asterisk in a more convenient
way. Meanwhile, this improvement also makes OpenVox the First open
source hardware device manufacturer in the industry that provides the
Global CID support. Users do not need to add any patch to zaptel or
asterisk and can directly have it with these two cards.

About CID:

Caller ID (caller identification, CID, or more properly calling number
identification) is a telephone service, available on POTS lines, that
transmits a caller's number to the called party's telephone equipment
during the ringing signal, or when the call is being set up but before
the call is answered. Where available, caller ID can also provide a
name associated with the calling telephone number. The information
made available to the called party may be made visible on a
telephone's own display or on a separate attached device.

Caller ID is often helpful for tracing down prank calls and other
unwanted intrusions. However, it can also impede communication by
enabling users to become evasive. The concept behind caller ID is the
value of informed consent; however, it also poses problems for
personal privacy.

Echo and Volume Tweaks for Asterisk using RXGAIN and TXGAIN

One of our friends contributed this today. A quick howto on how to configure TXGAIN and RXGAIN on Asterisk 1.4. Now, he's got a analogue trunks from Bell Canada, so YMMV and you will almost certainly have to change all of the values mentioned here to be customized to your environment. From what we've read, some phone companies may even know this information when you call one of their tech's and ask for it, bypassing the need for this work all together. However, echo and voip have long been battling siblings for many people, so we hope this little howto will help in that battle.

Here's the Howto:

1. Dialed the miliwatt test number from every zap channel

2. Adjusted RXGAIN on each channel until ztmonitor gave me a value as close to 14844 as possible. This number is the "magic number" according to many sites on the Internet. We won't link to them here, use Google. We ran this command to actually monitor each channel while testing:

# ztmonitor <zap line> -vv

3. Now we need to test and verify TXGAIN settings. In our case, we are using FreePBX, so we made modifications to extensions_custom.conf. In your environment things may be different. Just add the following to your extensions configuration file in Asterisk.

# nano /etc/asterisk/extensions_custom.conf

[from-milliwatt]
exten => s,1,Answer()
exten => s,2,Milliwatt()
exten => s,3,Hangup()

4. Then we set all Zap channels to go directly to that context - "from-milliwatt" in our case. By editing the default zapata configuration file in /etc/asterisk

# nano /etc/asterisk/zapata.conf
<your zap channels>
context=from-milliwatt

5. Then a singular Zap channel was chosen to do all testing/dialing from.

6. Then we dialed into each of the other Zap Channels.

7. While we were on the line from the Zap Out --> Zap, we did the following.

a. We ran ZTMonitor on the channel we were dialing FROM
b. We adjusted TXGAIN on the channel we were dialing into via zapata.conf. Note that Asterisk will accept changes in realtime, and simply requires a "reload" or "module reload" instead of a full start/stop to test these values in zapata.conf.

8. This continued for each channel, trying to attain the same value as before in ZTMonitor - 14844.

This is a lengthy, and boring task to complete, but, if you are having echo problems it's definately one to try out for your Asterisk Phone System.

« Previous Page — Next Page »