Conference set up with Conferences module on FreePBX hangs up after a few seconds

[Doug]

I have attempted to add a conference option using the Conferences module in FreePBX. However, when a call is placed to the conference, asterisk hangs up after a few seconds.

I have found a few references indicating the problem is because the meetme application (which is what the conference module in FreePBX is configuring) requires the Zaptel timer - http://www.voip-info.org/wiki/view/Asterisk+cmd+Meetme. Other references indicate that the Zaptel timer can be simulated using ztdummy http://www.voip-info.org/wiki/view/Asterisk+timer+ztdummy

1) Can somebody confirm if the Conference module/meetme application is hanging up almost immediately because there is a timer missing? Am I missing something else?
2) What is the correct way to provide a timer on the PIKA Warp? Is it to use ztdummy, or is there some other special hardware on board for doing this?
3) If ztdummy can solve the problem, I suggest it may be a good idea to include it in the future releases of the FreePBX build for the WARP.

[skar]

This issue is not caused by any missing timer. It is caused by missing audio files on the WARP. The conference extension tries to play a file but it is not there so hangs up.

To fix this problem, log onto your WARP and do the following:

#> cd /var/lib/asterisk/sounds
#> ln -sf conf-getpin.ulaw enter-conf-pin-number.ulaw
#> ln -sf conf-getconfno.ulaw enter-conf-call-number.ulaw
#> chown -R asterisk:asterisk /var/lib/asterisk/sounds/

This should resolve the problem.

NOTE: This has already been fixed in the trunk (http://svn.pikatech.com/pads/addons/trunk/freepbx-mysql/) of svn code base but is not part of the latest 2.0.8.3 (http://pikawarp.org/?p=281) load that was released. It is in the preview load found at the bottom of this page (http://pikawarp.org/?p=353) though.


skar (http://twitter.com/skar_pikawarp)

[Doug]

Yes. That was it. Works perfectly now. Thanks

[mikeSGman]

As for me that didn't work; alas I did verify the missing links b/c of your guide thanks! (If it helps no to note, I had to go to Digium's site and download the asterisk sounds library. I extracted it, b/c after reading this page (thank you) I found that I was missing conf-getpin.ulaw. Of course, I extracted it to a directory of my choice, and copied it over to /var/lib/asterisk/sounds and did the appropriate links and chown's needed.)

So, in essence here's my problem: Conference worked great a week ago. I did no updates, and made no changes. (seriously) Last night I decided to try a conference again with a few people, and no go. Announces as needed, and accepts incorrect pins asking for the correct pin. Upon successful pin entry it disconnects, and hangs up.

Any ideas? Very strange. This happened a few weeks back, did a fresh install, and it went away. But now its back... Kinda like that scary old ex-wife.

[skar]

Well if you did no updates, it is hard to say. Perhaps you can post part of your asterisk log with high detail showing the error occurring.

[Deepak Kenchamba]

1. Login as root
2. Execute the following commands



cd /usr/src/
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.4.0+2.4.0.tar.gz
tar -zxvf dahdi-linux-complete-2.4.0+2.4.0.tar.gz
cd dahdi-linux-complete-2.4.0+2.4.0
make all


Note: This might lead to an error stating you don't have the kernel source installed.

3. Download & setup the kernel sources
3a. Find out your kernel version


uname -r

mine was '2.6.18-194.17.4.el5'
3b. Download kernel rpm & extract it with the following commands. Depending on your kernel version & process make your url might change.


cd /
wget ftp://ftp.muug.mb.ca/mirror/centos/5.5/updates/i386/RPMS/kernel-devel-2.6.18-194.17.4.el5.i686.rpm
rpm2cpio kernel-devel-2.6.18-194.17.4.el5.i686.rpm | cpio -idmv

3c Test if download & setup worked


cd /lib/modules/`uname -r`/source

This should change directory to the kernel sources (using the symbolic link).
If this works you are all set to get the conference call to work for you.

4. Complete the dahdi (Digium/Asterisk Hardware Device Interface) setup that was halted in step 2 (above) - with the following commands.


cd /usr/src/dahdi-linux-complete-2.4.0+2.4.0
make all
make install
make config
chkconfig dahdi on
service dahdi start


5. Now try you conference call - it will most likely work.

I found the following link useful to setup asterisk 1.6
http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation