Installing Asterisk 1.4 on OpenSuse 10.2

I've had my Asterisk PBX offline for a while now for no good reason, so I decided I'd upgrade and put the latest Asterisk on my new machine. I wrote about the last time I installed, that was on an older AMD Athlon 1200MHz Thunderbird. It worked fine but I got a Linksys PAP2T-NA ATA and became very lazy. The ATA just registers with Les.net and I plug a phone in to it, so there's no need for Asterisk for just basic VOIP phone service.

I decided that I'd just use the ATA for a while then I got into some sound quality problems. I switched out many parts of the system trying to determine where the problem was but the best lead that I have so far is that my modem has issues. I'm not sure that the entire problem is with the modem but I do know that my Internet connection overall gets faster if I reboot it and that shouldn't be necessary.

Anyhow, voice quality issues apparently are common with VOIP and more people are using it anyway - the balance of features and cost is still in favour of VOIP. So I figure that if I get used to my system I can work the kinks out of it later. As time goes on and there are more users it should be easier to find help on my specifics too.

Today I want to revisit some of the work I did getting Asterisk 1.2 working and see what I have to do to get Asterisk 1.4 running. It's apparently not that different.

I wrote in great detail about my plans for Asterisk and all the benefits I want beyond basic phone service a while ago. I had something working with basic voicemail, phone conferencing and hairpinning. Now that I'm coming back to it on my new computer I've got a couple differences. I'm using OpenSuse 10.2 on this machine and it's a much heftier computer. I've got a dual core AMD Athlon 64x2 4200+ and an Asus M2NPV-VM motherboard. If call quality suffers while I'm playing WoW then I'll just have to break down and buy another server. A newer version of Asterisk has also been released, they're up to 1.4.8 1.4.9 (they updated while I was writing!). My old system used Asterisk 1.2. I'd heard something about things changing in the way you write extensions.conf with 1.4 but I don't see any differences in the docs I've read so far.

So I just followed my instructions for installing Asterisk on OpenSuse 10.1 from before. I use the source distribution from Digium instead of the package that's available for OpenSuse. The packaged version was still on 1.2 last time I checked.

My instructions for building and installing Asterisk worked almost completely the same as before.

cd
wget <a href="http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.9.tar.gz<br />
tar"
title="http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.9.tar.gz<br />
tar"
><a href="http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.9.tar.gz">http://ftp.digium.com/pub/asterisk/releases/asterisk-1.4.9.tar.gz</a><br />
tar</a> xvzf asterisk-1.4.9.tar.gz
cd asterisk-1.4.9
./configure
make
su
make install
make samples
make progdocs

The make samples line builds the sample configuration files for Asterisk. The make progdocs line builds Doxygen documentation for the Asterisk source code under your build directory in doc/api/html. A lot of people don't need that but I want to get deeper in to Asterisk.

I found a note in the readme that mentions another target make menuselect. If you use make menuselect, you get a menu where you can choose which modules to build. Right and left arrow keys let you look into different options and the enter key turns things on or off. I left all the options as-is but if you're missing the dependencies for some part of Asterisk you'll see XXX to the left of that part. If you do make menuselect, I think you'd do it after ./configure and before make.

Of course all the steps in the build worked for me. If you try it & have problems leave me a comment, I don't know if I can help but I'll give it a shot.

Once I installed Asterisk 1.4 I ran asterisk -vvvc to get a console. Since that seemed to come up without issue, I quit the console (stop gracefully) and copied a couple of my old conf files over. I just looked through /etc/asterisk/ to see what I'd changed and grabbed those. I also had to get my custom sounds from /var/lib/asterisk/.

After that I went to my control panel on Les.net and set my DID to connect to the peer Asterisk registers with and I'm able to call my new installation. It wasn't hard but these things do take time and exercise some rusty skills for me. Oh and of course I forgot about ztdummy. I'll probably have to go back and set that up again too.

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hi
i'm new to asterisk.
i installed OpenSuse10.2 and asterisk 1.4.13 but my dialPlan is not able to call AGI scripts [.PHP] from ii.
for that i'm using AGI tag and set all required permissins and all that.
but it gives message on Asterisk that..
PHP script completed by returning 0

is there any other setting needed for that configuration of calling PHP scripts from asterisk...?
pls help me out in this.

regards,
rahul

hi again,
also i'm need some help regarding to calling VXML file from asterisk ..

i know we have ready made function of VXML() for that but it gives me error with that no Application configured for that.

so pls help me out in this also.

thanx,
rahul

Hi there,

just tried to follow your instructions for installing Asterisk but ran into a problem and wondered if you can help.. My email i have left.. I get the following error... termcap support not found

Can you help? Its on openSUSE 10.3 .. ( i have pasted the bit where it errors below)

hecking for QDate in -lqt... no
checking for rc_read_config in -lradiusclient-ng... no checking for speex_encode in -lspeex... no checking for sqlite_exec in -lsqlite... no checking for ssl2_connect in -lssl... no checking for tds_version in -ltds... no checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tone_zone_find in -ltonezone... no checking for vorbis_info_init in -lvorbis... no checking for vpb_open in -lvpb... no checking for compress in -lz... yes checking zlib.h usability... yes checking zlib.h presence... yes checking for zlib.h... yes checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no
configure: error: *** termcap support not found
linux-developer:~/asterisk-1.4.9 #

@rahul
I've got no idea. I've only looked at AGI in passing so far.

@Ian
It looks like you need some ncurses package. There's a mention of it on voip-info for CentOS. The equivalent in OpenSuse 10.3 would be to start up Yast and search for ncurses then try it again.

Hi Im triying to installa asterisk on opensuse 10.2, when i execute ./configure everything work fine i guess.

but when i run make or make install i get a message like some error that say that y must run ./configure again.

If it's telling you to run ./configure again then that step probably wasn't successful. Look at the output from ./configure in more detail. It looks pretty boring but it should be clear in reading it if there's a dependency missing or some check failed.