Asterisk 1.4.5 SpanDSP Integration Howto

Here is a quick and dirty HowTo on getting SpanDSP to work with the latest editions of Asterisk 1.4. If you have any issues let me know in the comments and we'll get them sorted out.

First we'll have to download the asterisk source from their site.

# cd /usr/src
# wget ftp://ftp1.digium.com/pub/asterisk/asterisk-1.4.5.tar.gz
# tar xzvf asterisk-1.4.5.tar.gz

Now we'll have to grab the latest SpanDSP from Soft-Switch. We'll also have to compile this and make sure no other SpanDSP versions are available on our server. The paths may change if you are using /usr/local so edit the following commands appropriately.

# cd /usr/lib
# rm -rf *spandsp*
# cd /usr/include
# rm -rf *spandsp*
# cd /usr/src
# wget http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre3.tgz
# tar xzvf spandsp-0.0.4pre3.tgz
# cd spandsp-0.0.4pre3
# ./configure && make && make install
# ldconfig -v

Now that SpanDSP has been compiled and installed, we can continue with the integration to Asterisk 1.4.

First we will need to copy the app_txfax.c and app_rxfax.c into the Asterisk Applications directory. This can be done like this:

# cd /usr/src
# cd asterisk-1.4.5
# cd apps
# wget http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/app_rxfax.c
# wget http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/app_txfax.c
# cd ..

Now we'll have to add the SpanDSP support to Asterisk in order to make use of these applications. I prefer to manually apply the patch file from Soft-Switch because it almost always fails. You can try either way you want.

Open up the patch file in a web browser so you can easily see what lines need to change, the patch file is located here

patch file: http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.4/asterisk.patch

Now we'll need to manually apply these lines. Assuming you know something about patch files it should be pretty easy for you to modify the files, I'll briefly outline what needs to be done below.

First, Make sure you are in the Asterisk source directory.

# cd /usr/src/asterisk-1.4.5

Now, We'll have to modify the menuselect dependencies.

# nano build_tools/menuselect-deps.in

Search for RADIUS=@PBX_RADIUS@ and directly below this line put in the following new line

SPANDSP=@PBX_SPANDSP@

Exit and save the file.

Now you'll have to add a line to configure.ac

# nano configure.ac

Search for AST_EXT_LIB_SETUP([RADIUS], and directly below this line, put this in

AST_EXT_LIB_SETUP([SPANDSP], [spandsp Library], [spandsp])

Now, in the same file (configure.ac) we also have to add another line. Search for AST_EXT_LIB_CHECK([RADIUS], and directly below this line, add the following all on one line:

AST_EXT_LIB_CHECK([SPANDSP], [spandsp], [fax_init], [spandsp.h], [-ltiff])

Exit and save the file.

Now our last file to edit is the make options, which you can do like this

# nano makeopts.in

Search for the line like this RADIUS_LIB=@RADIUS_LIB@, add this directly below it

SPANDSP_INCLUDE=@SPANDSP_INCLUDE@
SPANDSP_LIB=@SPANDSP_LIB@

Exit and save the file.

Now we are almost ready to get the compiling going. But first, we have to copy the plc.h file to the Asterisk source so we won't get errors during compiling later on. You can do this as such:

# cd /usr/src/asterisk-1.4.5
# cd includes/asterisk
# cp plc.h plc.h.backup
# cp /usr/include/spandsp/plc.h .

Now because we've added new options to Asterisk, we'll have to bootstrap it

# cd /usr/src/asterisk-1.4.5
# ./bootstrap.sh

Now we re-run configure like the following, your options might be different

# ./configure --with-netsnmp --with-ssl --with-spandsp

Now we should go into the menuselect program to ensure that app_rxfax and app_txfax will be included when you compile Asterisk.

# make menuselect

Ensure that both applications are selected under the Applications menu. Exit the menuselect utility by pressing "x" to save your changes if required.

Now we can continue with the regularly scheduled compiling of the program.

# make
# make install

You should have a brand new copy of app_txfax.so and app_rxfax.so located in /usr/lib/asterisk/modules

Note that you might have to add /usr/local/lib or /usr/lib to your ld.so.conf in /etc in order to make this work, once you add the library path, re-run ldconfig, then try to start asterisk.

Have fun!

Small Asterisk-Addons 1.4 Chan OOH323 compilation howto

Today I was trying to upgrade to the latest Asterisk-Addons on my Asterisk server. I am using the 1.4 series (as you all should be by now)
and the most current release of Gentoo Linux. Because Gentoo uses somewhat "unstable" software I have differing versions of libtool.m4 and ltmain.sh. In order to get around this I had to do the following

# cd asterisk-addons-1.4.2
# cd asterisk-ooh323c
# nano configure

and then I had to hit control-w, and search for libtool.m4, once I found the section that it checked for this problem of the software being differing versions I commented it all out. Basically the whole section starting with the beginning if statement of the libtool check all the way to the bottom fi statement.

Next you'll need to compile the Asterisk-Addons package by doing the following:

# cd ..
# make

Notice that you will be presented with yet another error that it can't find the library file to copy for you when you try to issue the make install after compiling. This is a simple fix for now, just do this.

# cd asterisk-addons-1.4.2
# cd asterisk-ooh323c
# cd .libs
# cp libchan_h323.1.0.1 libchan_h323.so.1.0.1

Now you can continue with the installation by running this

# cd ..
# make install

That's it, you should now have chanooh323 available for your asterisk installation. If you have any difficulty let me know in the comments and I'll try and help.

[asterisk-biz] idefisk rc4 released

-----Original Message-----

A bit earlier than expected, we released idefisk rc4. (RC3 seemed to be
broken on certain configurations).

Changelog:

Fixed blank STUN option in a new SIP account.
For dialing without using Window dialing rules Idefisk strips spaces,
'+', '(' and ')'.
Fixed bug with STUN usage selector.
Fixed microphone mute bug.
Fixed echo cancelation bug
Fixed an issue with the installer

Thanks a lot to the people who reported those issues.
I wish to apologize to all the people where the upgrade from RC2 to RC3
didn't work. (RC4 should be a lot better!)

If you would find new bugs, please report them on support@asteriskguru.com

Zoa.

[asterisk-biz] Sipcat 1.2 is here

Hi list

Just a quick mail to let everyone know that our latest version of Sipcat
finally has arrived. We've incorporated much of the feedback we've
received over the past few months and have introduced a number of new
features. Here's a short overview of some of the new stuff:

- Multi-language interface in French, Germand and Dutch
- Video telephony support (H.264 and H.263 codecs)
- New IP phone provisioning support for all Polycom models
- Speedier interface, with possibility to activate changes later
- Much, much more.

Of course there's still a free version available for up to 4 users. For
more information, check out our website at http://www.sipcat.com/ or
read our new blog at http://www.sipcat.com/company/blog/

Best regards

Tom Schouteden

[asterisk-biz] idefisk IAX / SIP softphone RC 3 released

Yo yo,

We released release candidate 3 of our softphone, as usually freely
downloadable from http://www.asteriskguru.com/idefisk/
All bug reports are still welcome, as noted below we now support UTF-8,
anybody interested in making a translation into his / her language for
inclusion can contact me at support@asteriskguru.com.

Changes include:

Free version:

Added UTF-8 and Unicode support.
Fixed IAX blind transfer bug.
Fixed some audio quality issues on some crappy audio cards
Fixed some echo canceller issues (replaced the echo canceller with a new
one)
Added default STUN server.(stun.idefisk.com)
Added Disable DTMF sounds option.
Added audio enchancement option.
Fixed check box hint

Biz version:
Added "callto" handling to dial and without "extension=". (Biz)
Added in the Outlook dial window the first available phone to be
selected. (Biz)
Fixed outlook menu bug.(biz)
Added all users functionality.(biz)
Fixed exiting when the API cannot be initialized.(Biz)

There are 2 things we didn't fix yet

1) the windows DEP doesn't seem to like our phone yet, a workaround can
be found below:

If Idefisk generates fatal error or crashes during execution on windows
XP sp2 or windows VISTA please consider to do the following :
Disable Windows DEP feature for all programs and services or add Idefisk
as an exception to the DEP exceptions list.

1. Go to My computer Properties.
2. Select the Advanced Tab
3. Click on the Settings button in the Performance section.
4. Select the Data Execution Prevention tab.
5.1 Make sure that the option 'Turn on DEP for essential Windows
programs and services only' is selected.
OR
5.2 If 'Turn on DEP for all programs and service except for those I
select :' option is selected and don't want to change it.
5.2 Click the add button.
5.3 Select Idefisk executable file from Idefisk installation folder and
click OK.

2) on some audio cards we seem to cause the microphone to be muted at
startup, we are still investigating this issue.
If you experience this or other problems, please send us an email at
support@asteriskguru.com.

Cheers,

Zoa