Koozali.org: home of the SME Server
Legacy Forums => General Discussion (Legacy) => Topic started by: duncan on February 05, 2004, 09:47:59 AM
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Anyone interested in a very brief howto?
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What's this?
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Very much so, I had recently downloaded the source of Asterisk and compiled it on an e-smith 6 dev box in preparation to working out how to get it going. Once I had it configured my next task was to build a control panel for it - as this is my first e-smith project it was going to take some time :-)
If you have a how-to and some RPMs built I would love to see them.
Mark Leman
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What's this?
It is an open source PBX system which can switch and route calls between real phones & lines (with the correct hardware) and VoIP phones and lines. Many other featurs you would expect from a modern PBX and more! An amazing system but not very well documented - hence I have spent several weeks reading the docs and mailing list before coming to the conclusion I'm don't know much yet :-)
See http://www.asterisk.org/ to get started
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Thanks, it's very interesting.
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It is an open source PBX system which can switch and route calls between real phones & lines (with the correct hardware) and VoIP phones and lines. Many other featurs you would expect from a modern PBX and more! An amazing system but not very well documened - hence I have spent several weeks reading the docs and mailing list before coming to the conclusion I'm don't know much yet :-)
See http://www.asterisk.org/ to get started
And I am very much in the same boat.
If I did a howto it would simply be how to install it from source. Configuring it is not something I would want to tackle (Too many scenarios).
Did you find this site?
http://www.voip-info.org/wiki-Asterisk
Regards Duncan
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And I am very much in the same boat.
If I did a howto it would simply be how to install it from source. Configuring it is not something I would want to tackle (Too many scenarios).
Did you find this site?
http://www.voip-info.org/wiki-Asterisk
Regards Duncan
I had found www.voip-info.org but not noticed the /wiki-Asterisk (drowing in info :-).
A howto on installing from source is a good start, and then we could progress to making some RPMs which would be more useful to the average user. Once we can install the application from an RPM and configure via files we could start on a control panel, which makes sense in an e-smith environment. I was aiming write a control panel to automatically generate a setup file providing internal voip phone for each e-smith user and control a limited number of analogue and ISDN cards for external access.
Quite a big thing to bite off as a first e-smith project and the learning curve is proving steep but I have to aim somewhere ;-)
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I'm also intressted in asterisk.
/Mats
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After spending the past couple of days experimenting - I have to say you have picked a beauty for your first effort.
I like the idea of – create a user account – get a sip account (perhaps with voicemail).
ISDN already supported is handy although I did have to patch and rebuild the hisax driver for my netjet cards to allow voice transparency.
I am willing to help with something like this – however my experience is in the communications industry so it would probably be from that side of the IT fence.
Rolling RPMs holds little interest to me.
I have some Digium cards coming this week – so I decided to do a full install with the Zaptel stuff from CVS. The first thing it did was assault my modules.conf. Need to sort that out. I also have a Grandstream sip phone coming which I am pretty keen to try out. Get one of my guys on a wireless link connected in to the PABX.
I will piece a howto together over the next week.
Regards Duncan
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And man, I have got remember to log in. This Guest chap posts everywhere.
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You can enable a one year cockie in your contribs profile, then you don't need to remember to login.
The cockie is disabled as default.
/Mats
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the cookie dun work for me, i still gotta hit the login button EVERy time i come here, might be cos i ahve a gfew xoops sites myself?
IS the aterix stuff a fully featured voicemail system?
Could it do all your voice/fax stuff? i wouldnt mind something like that at home for our small business...several inbound lines, several voicemail boxes and a few fax numbers?
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Could it do all your voice/fax stuff? i wouldnt mind something like that at home for our small business...several inbound lines, several voicemail boxes and a few fax numbers?
Yes.
/Mats
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I would LOVE to see a howto.
Specifically i wish to use an SME gateway with 3 remote VPN sites (remotes have a hardware VPN router).
I would love to have Asterisk on the VPN gateway machine, with maybe 2 or 4 inbound analogue lines with a small number range, with voip handsets at each of the remote sites.
Is it possible to use this over the VPN? I want the remote sites COMPLETELY secured to only use VPN traffic to and from the SME gateway, they have no web access etc, the VPN will be over adsl - is 256kb adsl to each remote site sufficient for this?
Does anyone have any expereince in running the VPN network an thre PBX network together? could you post your experriences? or links to furher info?
I have posted some questions on the topic here
http://forums.contribs.org/index.php?topic=20662.msg81486#msg81486
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ASTERISK HOWTO
Go here (http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/)
Feel free to add or flame as you see fit.
Regards Duncan
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Thnanks VERY much for that, will take the time to start learning to install and configure this, and then to learn how to drive it.
SIP is a web/software based client?
Which is the board that connects your PBX to the POTS system?
EDIT: followed your links and ifnd my own answers :)
Are there handsets that can be plugged directly into a hardware VPN device that is VPNd to same machine as the PBX (SME, VPN/PBX machine) and be used as a normal PBX extension?
Is your setup in australia? Love to here from anyone trying this in Australia?
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sip = session initiated protocol
You can have software sip clients ie the xten client or something like this http://www.grandstream.com/y-product.htm which would be cool for what you want to do.
If your in Oz then you can get the gear from these guys.
Australian Technology Partnerships (Sydney) Martin Warner Voice: +61 (0)2 9438 4222
E-mail: sales at atp.org.au
WWW: www.atp.org.au
The pricing is really quite good.
And if your in Oz - it just cracked 43 where i am. Sooo hot.
Regards Duncan
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Thanks will be contacting them tomorrow. I want hansets, not software clients.
http://forums.contribs.org/index.php?topic=20662.msg81520#msg81520
Do you think asterisk will work in the situation i have described here, over the VPN?
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Late last year I played about with both Asterisk and SER (SIP Express Router) http://www.iptel.org/ser/
on SME V6.0.
Asterisk is very nice with lots of functionality but a pig to configure. I was able to set up a SIP gateway between my server in NZ and another Asterisk server in New York. We could call inter office extensions, I was using Xten soft phone and he was using Grandstream Budgetone phones. He also had Digum FXO and FXS cards installed and I was able to make local calls in New York. Quality was excellent.
In the end I decided that having a PABX sitting on the same box that is also a file server, web server, mail server is just asking for trouble and would be better on its own box.
I also set up SER on the SME server. This is purely a SIP Gateway. SER features presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available. It was easier to configure but like Asterisk had problems getting to work properly behind a firewall.
I must get back onto this project one day.
Jon
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Well its still a pig to configure :)
I looked a SER but wasnt to sure of the hardware side of things.
Pete, I am sure that running a VPN wont be a problem (Might even help), However it might be advisable to consider hardware stuff for running the vpns and having a linux box at the primary site. Probably end up being more cost effective that way.
My reason for doing this is primarily for research - mainly for backbone stuff. Most of the systems we sell are coming out with voip built in - so i havnt really thought about this as a solution for customers.
Also - the Digium cards are not austel approved - so you need to consider that. Atp are working with Digium to get a larger card approved later in the year.
ISDN cards can be got from www.traverse.com.au
Regards Duncan
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ISDN costs are prohibitive in Oz, i will run 2 POTS lines into my house, into the cards, into the PBX machine. The PBX will run only Asterisk, maybe an intranet.... I will turn off most/all the firewll stuff and run a hardware firewall/router/VPN server (hopefully easier to drive/maintain), cheaper hardware to buy. The above, unless i can figure a better/cheaper way, maybe one of these online providers?
Behind the firewall/VPN will also exist my SQL server that runs the accounting and POS systems, my home PC for managing these systems and regular PC stuff.
At each remote site i hope to have 1 x 4port VPN/firewall (netgear type device) that will handle the ADSL/vpn stuff. Connected to this i hope to have a network printer, POS pc, and one telephone handset.
Which cards do i need to implement the PBX side of this?
I was going to run VPN from the SME server, but as some one else has pointed out, its going to easier/secure to have the ADSL vpn/routers do this for me, plus its a matter of power off/on to reset, far easier to retail asisstants...
If anyone has better suggestions i would love to hear them. But i figure the above has a smallish hardware outlay that will potentially pay for itself quickly, i already have to have the VPNs for the POS stuff, adding the PBX and some handsets isnt a big deal really - but I dont think i can buy only 25 or 50 numbers from telstra, me thnks it must be in blocks of 100?
Do telstra allow you plug this equipment into the telephone network? I dont want any big fines...
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Your expostcvsroot line is missing the caps.
it should be
export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
:)
and your asterisk -vvvc command should be ./asterisk -vvvc i know its picky, but so many us noobs copy and p[aste the commands :)
Insdie the init script it says that asterisk is found at /usr/sbin/asterisk and your conf files are in /etc/asterisk but none of this is true :( seems as though asterisk only start from with the CVS dir?
and ,my make clean ; make install finishes with LOADS of errors, that seem to be assocaited with ssl??? anyway the PBX launches afterwards so its as you described re the errors?
res_crypto.c:228: dereferencing pointer to incomplete type
res_crypto.c:233: dereferencing pointer to incomplete type
res_crypto.c:233: warning: implicit declaration of function PEM_read_RSA_PUBKEY'
res_crypto.c:235: dereferencing pointer to incomplete type
res_crypto.c:235: warning: implicit declaration of function PEM_read_RSAPrivateKey'
res_crypto.c:237: dereferencing pointer to incomplete type
res_crypto.c:239: dereferencing pointer to incomplete type
res_crypto.c:241: dereferencing pointer to incomplete type
res_crypto.c:241: dereferencing pointer to incomplete type
res_crypto.c:243: dereferencing pointer to incomplete type
res_crypto.c:244: dereferencing pointer to incomplete type
res_crypto.c:245: dereferencing pointer to incomplete type
res_crypto.c:246: dereferencing pointer to incomplete type
res_crypto.c:246: dereferencing pointer to incomplete type
res_crypto.c:248: warning: implicit declaration of function ERR_print_errors_fp'
res_crypto.c:252: dereferencing pointer to incomplete type
res_crypto.c:253: dereferencing pointer to incomplete type
res_crypto.c:260: dereferencing pointer to incomplete type
res_crypto.c:268: dereferencing pointer to incomplete type
res_crypto.c: In function ast_sign':
res_crypto.c:391: dereferencing pointer to incomplete type
res_crypto.c:397: warning: implicit declaration of function SHA1'
res_crypto.c:400: warning: implicit declaration of function RSA_sign'
res_crypto.c:400: NID_sha1' undeclared (first use in this function)
res_crypto.c:400: (Each undeclared identifier is reported only once
res_crypto.c:400: for each function it appears in.)
res_crypto.c:400: dereferencing pointer to incomplete type
res_crypto.c:403: dereferencing pointer to incomplete type
res_crypto.c: In function ast_check_signature':
res_crypto.c:424: dereferencing pointer to incomplete type
res_crypto.c:442: warning: implicit declaration of function RSA_verify'
res_crypto.c:442: NID_sha1' undeclared (first use in this function)
res_crypto.c:442: dereferencing pointer to incomplete type
res_crypto.c: In function crypto_load':
res_crypto.c:462: dereferencing pointer to incomplete type
res_crypto.c:463: dereferencing pointer to incomplete type
res_crypto.c:482: dereferencing pointer to incomplete type
res_crypto.c:483: dereferencing pointer to incomplete type
res_crypto.c:484: dereferencing pointer to incomplete type
res_crypto.c:484: dereferencing pointer to incomplete type
res_crypto.c:487: dereferencing pointer to incomplete type
res_crypto.c:490: dereferencing pointer to incomplete type
res_crypto.c:491: warning: implicit declaration of function RSA_free'
res_crypto.c:491: dereferencing pointer to incomplete type
res_crypto.c:492: warning: implicit declaration of function free'
res_crypto.c:454: warning: nkey' might be used uninitialized in this function
res_crypto.c: In function show_keys':
res_crypto.c:516: dereferencing pointer to incomplete type
res_crypto.c:517: dereferencing pointer to incomplete type
res_crypto.c:518: dereferencing pointer to incomplete type
res_crypto.c:519: dereferencing pointer to incomplete type
res_crypto.c:521: dereferencing pointer to incomplete type
res_crypto.c: In function init_keys':
res_crypto.c:537: dereferencing pointer to incomplete type
res_crypto.c:538: dereferencing pointer to incomplete type
res_crypto.c:542: dereferencing pointer to incomplete type
res_crypto.c: In function crypto_init':
res_crypto.c:596: warning: implicit declaration of function SSL_library_init'
res_crypto.c:597: warning: implicit declaration of function ERR_load_crypto_strings'
res_crypto.c: In function init_keys':
res_crypto.c:531: warning: kn' might be used uninitialized in this function
make[1]: *** [res_crypto.o] Error 1
make[1]: Leaving directory /usr/src/asterisk/res'
make: *** [subdirs] Error 1
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ISDN for voice is not that expensive - same for long distance and cheaper for local up to 5 minutes (Timed local call).
Basic rate (BRI) can have 8 numbers/msn (although you need only start with one) per service (two lines) or a hundred number indial range.
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Thats what happens when you edit for spell checks in word.
Ta
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That looks like a problem with the openssl-devel package. Did it install properly?
[root@server root]# rpm -q openssl-devel
openssl-devel-0.9.6b-28
[root@server root]#
asterisk -vvvc will work from anywhere - asterisk didnt finish compiling. I had problems with cutting and pasting this command. Better if you type it in.
Out fo curiosity - are you running 5.6 or 6
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Duncan can this be used on a SME 6.0 ?
/Mats
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Hmmm i could have sworn i checked that - upon further investigation, i had install ssl-dev on the SME6 machine, but that failed anyway (prolly cos i test heavily and its pretty screwed), so i fired up 5.6 and no ssl-dev, so installed and trying again :( - i got SMEs running in windows and its late :) Seems so far to playing far more nicely.
Thanks - sorry to bother you with trivial stuff....
Matsk - he said he aint tested it yet, but all the dev packages appear to be the same, i dont see why it wouldnt, i will install 6 from new tomorrow and try...
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Matsk,
As I posted earlier I have had Asterisk running on SME V6.0 no problem. You just need the dev rpm's to compile it
Jon
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Do any of you use the GUI type tools? If so which.
Also how the hell do you set the username for remote access? i look in manager.conf and there is a password, i assume for a default username? i have tried all i can think of, how do i speficiy one?
EDIT: in case anyone else tries ot connect remotely, its in manager.conf, the lines are commented out :( I will give up tonight an sleep, before i hurt myself :)
Thanks for your help Duncan.
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Matsk,
As I posted earlier I have had Asterisk running on SME V6.0 no problem. You just need the dev rpm's to compile it
Jon
Thx, I guess I will download the dev rpm's real soon!
I have an RH 9.0 Asterisk server that I can free to an SME development platform.
So now to SME-asterisk integration.
/Mats
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I dont use any GUI tools.
You should be able to get a basic sip setup running by adjusting sip.conf and extensions.conf only (Thats all i have touched along with modem.conf for my ISDN)
One thing i have noticed is that a reload sometimes is not enough. It is better to do a stop now and then restart the process.
Regards Duncan
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Duncan, do you run yours in Australia? connected to the telco network?
Which hardware is permitted? i cant find any that seems to be approved :(
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Maybe this is of any help:
http://www.voicetronix.com.au/
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Hi pete - Im in WA. I am running mine as an extension (ISDN) of our PABX using a netjet card from traverse.
Maybe this is of any help:
http://www.voicetronix.com.au/
This is about the only legit gear in Australia - apart from the Dialogic stuff.
Pretty expensive.
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ISDN isnt really an option.
Does anyone know what is the smallest number range available from telstra? do they do blocks of 25 or 50?
We have 8 x 100 ext number ranges, we used to have some in via 4 POTS, (as a fallback measure), this type of setup, with say 3 lines would work ok for in and outbound calls for the PABX?
Can you get telephone handsets that have RJ45 5terminations, that will plug into one of those netgear routers on the other end of a VPN connection, give it an IP, configure ext/IP in pabx and you now have a full indial telelphone extension at remote location?
I want FXO for handling the connection of PBX to telco network via POTS?
Aside from the detail of PBX config, the above is essentially it?
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Smallest number range - if you are talking indial is 100 - Primary or Basic rate. Analogue indial is now almost unheard of.
The Grandstream phones are IP based with rj45 connections.
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So did anyone get astersik working on a different machine? I don't seem to get a registration back from my viop gateway. I can see the traffic for registration going out (watching with ethereal) but noting get backs to the asterisk box from the esmith box.
Anyone got any suggestions as to how I can manage this?
Another Aussie.
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Could you give a little more detail on your setup. Are you running Asterisk on the e-smith box - and if so are you using IAX.
Regards Duncan
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For those who don't want to install the devtools - and instead want a RPM install then you may be interested in my experiment today. I tried Asterisk on SME 6.0.1 using RPMs rather than a CVS snapshot. You need to install the asterisk with
rpm -ivh --nodeps asterisk-040904-rh73.LSE.1.i386.rpm
and add these lines to modules.conf
noload => cdr_odbc.so
noload => cdr_pgsql.so
noload => chan_zap.so
The RPMs are from
http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
ftp://ftp.linuxsys.com/pub/LSE/packages/7.3/asterisk/
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hi
Will Asterisk work with the voip services providers?
Paul
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Yes it will.
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When I try to install libpri and asterisk I get the following error:
Makefile:73: .depend: No such file or directory.
I just made a clean installation of SME 6.0.1, and I have the following development tools installed :
binutils-2.11.93.0.2-11.i386.rpm
bison-1.35-1.i386.rpm
cpp-2.96-113.i386.rpm
gcc-2.96-113.i386.rpm
glibc-devel-2.2.5-44.i386.rpm
glibc-kernheaders-2.4-7.16.i386.rpm
kernel-source-2.4.20-18.7.i386.rpm
libogg-1.0rc3-1.i386.rpm
libvorbis-1.0rc3-1.i386.rpm
mpg123-0.59q-1.i386.rpm
ncurses-devel-5.2-26.i386.rpm
openssl-devel-0.9.6b-35.7.i386.rpm
readline-4.2a-4.i386.rpm
readline-devel-4.2a-4.i386.rpm
sox-12.17.3-4.i386.rpm
I did the checkout and tried to compile everything in the following order:
zaptel
libpri
asterisk
Could you please tell me what am I missing??
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I did everything in the howto. down to this last part where it says i should be able to connect with:
"asterisk –r"
I get this message:
"Unable to connect to remote asterisk"
I made sure that i ran these properly too:
cp /usr/src/asterisk/contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk
ln -s /etc/rc.d/init.d/e-smith-service /etc/rc.d/rc7.d/S93asterisk
/sbin/e-smith/db configuration setprop asterisk status enabled
/sbin/e-smith/signal-event console-save
/sbin/reboot
any ideas whats causing this? evidently its Asterisk is not starting upon boot... but im' not sure what to do about it...
cheers.
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organetic - did you do "make and make dep" in your /usr/src/linux-2.4.20-18.7 directory.
cydonia run asterisk –vvvvc and see where it flakes out. If you are running a AMD, C3 or 386 machine the makefile needs editing.
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cydonia run asterisk –vvvvc and see where it flakes out. If you are running a AMD, C3 or 386 machine the makefile needs editing.
seems to be fine. does a lot of stuff and settles at;
== Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
*CLI>
Any ideas?
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One thing i've noticed...
asterisk -r doesn't work from anywhere, even /usr/src/asterisk . but, if you go to /usr/src/asterisk , then type:
./asterisk
hit enter, then type asterisk -r. it connects like this:
Asterisk CVS-HEAD-05/19/04-20:01:14, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <markster@digium.com>
=========================================================================
Connected to Asterisk CVS-HEAD-05/19/04-20:01:14 currently running on centralserver (pid = 2923)
centralserver*CLI>
I have no idea how to get this to work all the time. Perhaps you could create a link from /usr/src/asterisk to /usr/bin/asterisk ??
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Ok,
Sounds like it didnt install properly. Do a 'make clean' 'make install' in the asterisk directory and see where it flakes out. It should end and say something about running 'make samples'.
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Ok,
Sounds like it didnt install properly. Do a 'make clean' 'make install' in the asterisk directory and see where it flakes out. It should end and say something about running 'make samples'.
Hmmm. i think that fixed it:D. I had done it before. but i did "make clean ; make install". Weird. Now, afterwards, when i did 'asterisk -vvvc' I got:
Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect.
I havn't even done this part:
cp /usr/src/asterisk/contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk
ln -s /etc/rc.d/init.d/e-smith-service /etc/rc.d/rc7.d/S93asterisk
/sbin/e-smith/db configuration setprop asterisk status enabled
/sbin/e-smith/signal-event console-save
/sbin/reboot
I guess the links are there from the previous times i tried. Will reboot and see if it works.
Thanks alot Duncan.
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Ok,
After reboot, "asterisk -r" doesn't work. But "asterisk -vvvvc" works from anywhere in my dir structure.
I'm thinking its a problem to do with this part of my install process:
cp /usr/src/asterisk/contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk
ln -s /etc/rc.d/init.d/e-smith-service /etc/rc.d/rc7.d/S93asterisk
/sbin/e-smith/db configuration setprop asterisk status enabled
/sbin/e-smith/signal-event console-save
/sbin/reboot
Is yours installed on a 6.0.1 box? Maybe the structure is slightly different for me(on a 6.0.1 box).
EDIT:
"This is for 5.6 – you may need to adjust things if you are running something else"
Ok, so there is my problem, just gotta work out the correct way to add asterisk to the start up. If i wasn't such a newb with linux i'm sure the solution would be clear to me :P.
Cheers and thanks for all the help. I will have to keep tinkering around. Anybody out there installed Asterisk on a 6.0.1 box that can offer some advise?
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I am now running on 6.0.1. Pretty well followed the howto word for word. After looking at my setup - I reckon the commands should work as is.
Watch it during the boot sequence (near the end) to see what happens.
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Watch it during the boot sequence (near the end) to see what happens.
Thanks for the tip, getting closer:).
It says:
Starting Asterisk: [DISABLED]
SO all i have to do is enable it. How do i do that?
Cheers.
Tristan
btw. have you looked into www.atp.org.au? they lok quite good.
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Even following the guide for 5.2 step by step I'm still having the same problem:
Makefile:73: No such file or directory.
What about a step by step guide on the compilation for the latest asterisk CVS and SME 6.0.1?
Best regards to all contributors,
Organetic
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Well, i havn't figured out how to get asterisk to start on boot yet(havn't really tried:P).
But, i did install X-Lite and make my first call using the asterisk server:D. Called Digium HQ. something like:
-------
Tristan: Hey, sorry i've just installed asterisk i was just testing it.
Digium: no problems. it seems to work. have a good day.
-------
This is great. only voice chat i've ever done before was with msn. cant wait to set this up more, with voicemail etc.
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You could remove the /etc/rc.d/rc7.d/S93asterisk link and simply link to the program ie
ln -s /etc/rc.d/init.d/asterisk /etc/rc.d/rc7.d/S93asterisk
I like doing it the other way as it easy to enable/disable the service.
I have bought digium gear from atp. No problems.
Organetic - sorry, cant think of a reason for your problem.
Regards Duncan
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Could anyone please tell me if any of these dev-tools is wrong or inadequate for the compilation of the latest asterisk CVS on SME 6.0.1-01?
binutils-2.11.93.0.2-11.i386.rpm
bison-1.35-1.i386.rpm
cpp-2.96-113.i386.rpm
gcc-2.96-113.i386.rpm
glibc-devel-2.2.5-44.i386.rpm
glibc-kernheaders-2.4-7.16.i386.rpm
kernel-source-2.4.20-18.7.i386.rpm
libogg-1.0rc3-1.i386.rpm
libvorbis-1.0rc3-1.i386.rpm
mpg123-0.59q-1.i386.rpm
ncurses-devel-5.2-26.i386.rpm
openssl-devel-0.9.6b-35.7.i386.rpm
readline-4.2a-4.i386.rpm
readline-devel-4.2a-4.i386.rpm
sox-12.17.3-4.i386.rpm
Sorry to bother you again.
Best regards,
Organetic
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You could remove the /etc/rc.d/rc7.d/S93asterisk link and simply link to the program ie
ln -s /etc/rc.d/init.d/asterisk /etc/rc.d/rc7.d/S93asterisk
Regards Duncan
Great. that did it Duncan! Dont know why the other way wouldn't work on mine...:S.
Thanks alot. Now i just gotta muck around with SIP/IAX and figure this all out. SIP experience so far has been a bit so so...
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ASTERISK HOWTO
This is for 5.6 – you may need to adjust things if you are running something else
My server is in server only mode and is protected by another firewall
The following software represents a security risk – so remove the compiler stuff when done (This is entirely up to you)
If you want to keep copies of the software just download each item and run rpm –ivh *.rpm from the directory
h.323 is not included in this howto – use sip.
rpm -ivh http://mirror.contribs.org/smeserver/contribs/dmay/mitel/contrib/dev-tools/SME56/cpp-2.96-112.i386.rpm
rpm -ivh ftp://at.rpmfind.net/linux/redhat.com/dist/linux/updates/7.2/en/os/i386/kernel-headers-2.4.9-34.i386.rpm
rpm -ivh http://mirror.contribs.org/smeserver/contribs/dmay/mitel/contrib/dev-tools/SME56/glibc-devel-2.2.5-40.i386.rpm
rpm -ivh http://mirror.contribs.org/smeserver/contribs/dmay/mitel/contrib/dev-tools/SME56/gcc-2.96-112.i386.rpm
rpm -ivh http://mirror.contribs.org/smeserver/contribs/dmay/mitel/contrib/dev-tools/SME56/ncurses-devel-5.2-26.i386.rpm
rpm -ivh ftp://rpmfind.net/linux/redhat/7.3/en/os/i386/RedHat/RPMS/bison-1.35-1.i386.rpm
rpm -ivh http://mirror.contribs.org/smeserver/contribs/dmay/mitel/contrib/dev-tools/SME56/openssl-devel-0.9.6b-28.i386.rpm
GETTING THE SOFTWARE
Mpg123 for music on hold (optional)
cd /usr/src
wget http://www.mpg123.de/mpg123/mpg123-0.59r.tar.gz
tar -zxvf mpg123-0.59r.tar.gz
cd mpg123-0.59r
make linux
make install
Asteriskpbx
cd /usr/src
export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
cvs login
(The password is anoncvs)
cvs checkout zaptel libpri asterisk
(You will most likely need the kernel sources if you want to install zaptel and libpri. This stuff is for digiums cards and is not necessary if you are only going to be running Sip.
cd asterisk
make clean ; make install
(you will see some errors – don’t worry too much about them – they are generally about X – and codecs you dont have etc )
make samples
(Lets test to see if it starts ok)
asterisk –vvvvc
(To stop type in - stop now)
SET IT TO START AT BOOT TIME
cp /usr/src/asterisk/contrib/init.d/rc.redhat.asterisk /etc/rc.d/init.d/asterisk
ln -s /etc/rc.d/init.d/e-smith-service /etc/rc.d/rc7.d/S93asterisk
/sbin/e-smith/db configuration setprop asterisk status enabled
/sbin/e-smith/signal-event console-save
/sbin/reboot
You should be able to connect to the server by typing asterisk –r
Config files are in /etc/asterisk
Most of this stuff was ripped off from http://www.automated.it/guidetoasterisk.htm so credits go to Andy Powell
A good place to start is http://www.voip-info.org/wiki-Asterisk as well as the above link.
A nice free sip client can be got from here
http://www.xten.com/download/download.php?A=D&brand=x-lite&F1=2910514834
Feel free to add or flame as you see fit.
Regards Duncan
I just did a fresh install of SME 5.6 and followed the guide in every detail. The result when I try to compile asterisk v1.0 stable is
"... bad interpreter: Permission denied"
"make: *** [.depend] Error 126
I'm frying my brains trying to install asterisk on SME since last month...
Best regards.
-
If you like - I can build an RPM over the next couple of days. It would only be suitable for Pentium II and up - Not for C3, AMD, 386 etc.
It would also be built on 6.0.1
Regards Duncan
-
If you like - I can build an RPM over the next couple of days. It would only be suitable for Pentium II and up - Not for C3, AMD, 386 etc.
It would also be built on 6.0.1
Regards Duncan
that would be great if you could
-
Its done. I just want to test it for a bit before letting it loose.
-
If you need any beta testers let me know...
btw. asterisk is up and running beautifully! I have termination with atp, a did number, and have configured an IVR.
Thanks for all your help getting it set-up on my SME server.
-
Its done. I just want to test it for a bit before letting it loose.
Duncan it would be nice to add AM (http://www.voip-info.org/wiki-AM (http://www.voip-info.org/wiki-AM)) into a panel and add that to the rpm ?!
I have been playing around with the documentation of mason so maybe a panel with asterisk config could be in place soon...
/Mats
-
That looks interesting, cant get it to work though. I just get server errors in the cgi-bin. Anything special that needs to be done to get it to work.
I would like to keep the rpm reasonably clean. At the moment it installs with no dependencies on 6.0.1 and sets the server up disabled. It follows SMEs template system - so its pretty easy to enable/disable. I have only tested it in a vmware machine.
It wouldnt be that hard to build a second rpm with stuff like AM etc built in. That would give the user a choice on how they want run the system.
-
The rpm is available from
http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/
It is not suitable for 386/C3 platforms.
It was built and tested on a fresh vmware machine from 0.9.0.
If you have and existing installation - back up your config files.
The howto should be fine.
Let me know if there are any dramas.
Regards Duncan
-
I did not had the right opportunity to test it yet but anyway thanks for your attention duncan.
greetings and keep up the good work.
I'm getting back to asterisk... :hammer:
-
Nice work Duncan.
I'm gonna buy another computer to muck around with that rpm on...
-
If someone has installed the package - can they let me know how it went. I have no way to test it outside of a Vmware machine at the moment.
Regards Duncan
-
Duncan,
the installations seems to be fine so far, I am running SME Server 6.0.1-01, I was able to connect with a cisco sip phone and get the IVR promts and messages. Still more to be done.
Great job..
Frank
-
The rpm is available from
http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/
It is not suitable for 386/C3 platforms.
Duncan,
Is it possible to create an i386 version (which should then also be able to be used on C3)?
-
I have a Mini-ITX here that I was planning to do a build on. I will sort something out later this week.
Regards Duncan.
-
That looks interesting, cant get it to work though. I just get server errors in the cgi-bin. Anything special that needs to be done to get it to work.
Think PHP needs an update, I'll get back on this.
I would like to keep the rpm reasonably clean. At the moment it installs with no dependencies on 6.0.1 and sets the server up disabled. It follows SMEs template system - so its pretty easy to enable/disable. I have only tested it in a vmware machine.
I can agree on this, but my future thoughts was to integrate the user creation part with a field for SIP phone nr.
It wouldnt be that hard to build a second rpm with stuff like AM etc built in. That would give the user a choice on how they want run the system.
Agree.
/Mats
-
Hi
I have just poped back to see How thinks were getting on. And must say it is looking good!
I will have a go with your rpm over the weekend?
Just a couple of questions
is there a control panel or AM installed?
And I am going to need a softphone to test with any sugestions as what software I should use?
Paul
Keep the great work, you are added a very usefull function to SME!
-
No panel or AM yet and for software try http://www.xten.com/.
/Mats
-
I have been lurking this forum for some time ... have followed your discussion and tried some of the ideas you mention ...
biggest problem I have encountered was getting the zapata and zaptel pieces to compile properly ... suspect that if I had been more patient, would have eventually figured out how to do it ...
as it happens, I chanced across a site that had pre-built RPMs for RH 7.3 for the zapata, zaptel and libpri pieces ... I just installed them on my SME 6.0.1 (fresh install) and they dropped right into place ... my Digium cards are ALIVE !!!
the link to the ftp site is
ftp://ftp.nacs.net/asterisk/rh73/RPMS/
hope this saves someone else the 2 days I killed trying to do it myself ...
regards
-
Hi,
everybody is talking about working external ISDN cards in their * config. I'd like to do so, too.
See http://forums.contribs.org/index.php?topic=23168.0
Could anyone give me some basic education on configuring * to work with an (passive) ISDN card for my SIP clients (that work) to place and recieve calls?
I'd be so thankful... I'm trying for days... just too stupid?
whte_rbt
-
I have updated the Asterisk rpm to smeserver-asterisk-1.0-RC1.i686.rpm.
You can grab it here (http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/)
Back up your config files before installing. Enjoy - let me know of any problems.
Regards Duncan
-
Duncan,
Great job on the Asterisk rpm. Works great. Am having trouble compling the drivers (zaptel, libpri, zapata).
Can you help?
JV
-
I have updated the Asterisk rpm to smeserver-asterisk-1.0-RC1.i686.rpm.
You can grab it here (http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/)
Back up your config files before installing. Enjoy - let me know of any problems.
Regards Duncan
Will this rpm work with Athlon processors? I just installed Asterisk on another computer but want to run it on my SME server (which is an Athlon).
Charlie
-
Charlie,
I am running it on an Athlon XP1800.
Just download Duncan's rpm and install. Works great.
Would be even more wonderful if I could compile the drivers for the cards I bought from Digium!
JV
-
I will have a go at compiling the drivers this afternoon and see whats what.
This build contains the firmware for Digiums IAXy (http://www.digium.com/index.php?menu=iaxy) unit. Unfortunatly we cant get them yet down here, but they look like being a good option for remote users.
-
Duncan,
Got through all the complie stuff, but...
the drivers won't load with modprobe. Something about the version of the kernel not matching the version of the modules.
Am sure it's something I'm doing wrong but am not a whiz on this yet.
JV
-
Where did you get the drivers from ?
-
Duncan,
Driver from Digium CVS
cd /usr/src
export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
cvs login
Then:
cvs checkout zaptel libpri
Here's what I've done so far.
1. Installed kernel source rpm from the 6.0 release. This puts a bunch of patch files
and a tar.bz2 file in /usr/src/redhat/SOURCES.
2. Uncompressed the bz2 file with bunzip2.
3. Extracted tar file. Puts files in /usr/src/redhat/SOURCES/linux-2.4.20
4. Digium "make" script needs soft link to kernel sources. So, "cd to /usr/src".
Then "ln -s /usr/src/redhat/SOURCES/linux-2.4.20 linux".
5. Check soft link with "ls -ld /usr/src/linux". Should see the link.
6. Create the kernel config file. "cd /usr/src/linux" then "make menuconfig".
At the bottom of the menu is Load alternate configuration file. Select that option
and enter "/boot/config-2.4.20-18.7". Then back to main menu and select Save alternate
configuration file. (Note: I think this may be the problem with the mismatch error
message when I try to modprobe the driver). Enter "/arch/i386/defconfig" then OK to
save. Exit menu.
7. Enter "make dep". Bunch of stuff flashes by for a minute or so. This is suppose to
create the modversions.h file that zaptel needs.
8. Then "cd /usr/src/zaptel". "make clean" then "make install" (Note: this is where
things go south on me. Get error messages about unresolved symbols)
See Digium FAQ -
http://digium.com/index.php?menu=faq#Installation_0
JV
-
You are making things a bit hard for yourself. I will do a zaptel howto (just did a quick install on a test machine) and add it to my contribs directory.
-
Zaptel howto here (http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/zaptel.htm)
Dev tools
For SME 6
http://mirror.contribs.org/smeserver/contribs/saco/contrib/devtools_SME6.0/
For SME 6.0.1
http://mirror.contribs.org/smeserver/contribs/hpe/devtools-6.01/dev-rpms/
Regards Duncan
-
Duncan,
Thanks for the great howto. Much easier than what I struggled with and more complete.
Still end up at the same spot though. Here's what I get when I try to load the driver with modprobe:
# modprobe wcfxo
/lib/modules/2.4.20-18.7/misc/zaptel.o: kernel-module version mismatch
/lib/modules/2.4.20-18.7/misc/zaptel.o was compiled for kernel version 2.4.20-18.7custom
while this kernel is version 2.4.20-18.7.
/lib/modules/2.4.20-18.7/misc/zaptel.o: insmod /lib/modules/2.4.20-18.7/misc/zaptel.o failed
/lib/modules/2.4.20-18.7/misc/zaptel.o: insmod wcfxo failed
Also when I run depmod:
# depmod -a
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7-e-smith/kernel/drivers/net/ppp_generic.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/tor2.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/torisa.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wcfxo.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wcfxs.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wct1xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wct4xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wcusb.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/zaptel.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/ztd-eth.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/ztdynamic.o
Tried to force insmod to load the driver with same result.
Suggestions?
JV
-
You probably need to look in the Kernel Makefile and get rid of the "custom" bit at the top.
Duncan
-
Duncan,
Removing the custom bit in the Makefile solved that problem.
Now it down to this:
/sbin/depmod -a
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7-e-smith/kernel/drivers/net/ppp_generic.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/tor2.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/torisa.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wcfxo.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wcfxs.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wct1xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wct4xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/wcusb.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/zaptel.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/ztd-eth.o
depmod: *** Unresolved symbols in /lib/modules/2.4.20-18.7/misc/ztdynamic.o
[ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf
[root@zythian2 zaptel]# modprobe wcfxo
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol proc_mkdir_Rsmp_caa5a36f
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol create_proc_entry_Rsmp_43b0d544
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol __write_lock_failed
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol register_chrdev_Rsmp_d52e100f
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol request_module_Rsmp_27e4dc04
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol remove_wait_queue_Rsmp_c3bb9b34
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol add_wait_queue_Rsmp_51c5cc8e
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol __generic_copy_to_user_Rsmp_d523fdd3
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol unregister_chrdev_Rsmp_c192d491
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol sprintf_Rsmp_1d26aa98
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol __read_lock_failed
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol __wake_up_Rsmp_127fda83
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol printk_Rsmp_1b7d4074
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol kfree_Rsmp_037a0cba
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol kmalloc_Rsmp_93d4cfe6
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol remove_proc_entry_Rsmp_5607dea6
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol schedule_Rsmp_4292364c
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol __generic_copy_from_user_Rsmp_116166aa
/lib/modules/2.4.20-18.7/misc/zaptel.o: unresolved symbol __pollwait_Rsmp_c50c8dc7
/lib/modules/2.4.20-18.7/misc/zaptel.o: insmod /lib/modules/2.4.20-18.7/misc/zaptel.o failed
/lib/modules/2.4.20-18.7/misc/zaptel.o: insmod wcfxo failed
When I run the make oldconfig, am I suppose to hit enter or do I have to enter 'y' each time?
JV
-
Duncan,
Found this in the Digium lists:
Oliver,
Thanks. That was the problem. I removed the linux and asm include
directories and linked them to the kernel source directories.
Gregg
On Sun, 2003-05-11 at 06:45, The Traveller wrote:
> Yo Gregg,
>
> Did you recently upgrade your kernel?
> You might be building against the wrong kernel-headers. Check if the
> kernel you're running was actually built from the sources you have
> in "/usr/src/linux/" and if "/usr/include/linux" and "/usr/include/asm"
> are symlinks to the appropriate places in this source-tree. Some
> distributions place the files there directly instead of symlinking, which
> goes wrong when you choose to compile your own kernel.
>
>
>
> Grtz,
>
> Oliver
>
> On Fri, May 09, 2003 at 22:30:04 -0400, Gregg Lebovitz wrote:
>
> > I am building asterisk in the same way I have done it in the past.
> > Recently, I started seeing errors when the zaptel 'make install' does a
> > depmod -a.
Tried it but no go.
JV
-
Duncan,
I've learned over the years that sometimes you have to start at the beginning, so I did a fresh install of 6.01 and used your directions.
I now have green lights on the Digium cards!
Thanks again. Send me an email as I would like to get you something for your help.
JV
-
Duncan,
I've learned over the years that sometimes you have to start at the beginning, so I did a fresh install of 6.01 and used your directions.
I now have green lights on the Digium cards!
JV
Great, that lets me know that on a fresh install - the howtos and packages just work.
Thanks again. Send me an email as I would like to get you something for your help.
JV
That’s not necessary. Being a comms guy – this stuff interests me – so I am happy to help.
I am not happy with the modules.conf template - it will work but its a bit ugly. I will sort something when I get some time and amend the howto.
-
Jonvee, I have adjusted the howto to better deal with modules.conf. Note renamed zaptel => 10zaptel.
It will matter - so you should make the change.
-
im rather new in asterisk ,..and i got some problems while trying to compile zaptel.
we have the same errors that jonvee...but i dont understand the explanation.(the first part...where it says "Removing the custom bit in the Makefile solved that problem."..
i dont know what is the custom bit..cna anybody explain to me more specific how to do it?
-
to be more specific..when i do make install zaptel...it gives me this errors
fi
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/tor2.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/torisa.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wcfxo.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wcfxs.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wct1xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wct4xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wcusb.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/zaptel.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/ztd-eth.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/ztdynamic.o
/sbin/depmod -a
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/tor2.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/torisa.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wcfxo.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wcfxs.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wct1xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wct4xxp.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/wcusb.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/zaptel.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/ztd-eth.o
depmod: *** Unresolved symbols in /lib/modules/2.4.25-1-386/misc/ztdynamic.o
[ -f /etc/zaptel.conf ] || install -m 644 zaptel.conf.sample /etc/zaptel.conf
-
First things first. Whats the deal with your kernel because 2.4.25-1-386 is not the default. Have you upgraded?
-
i have done an apt-get install 2.4.25....cause when ive tried to donwload the original kernel-source(2.4) it didnt appear. so in other words i have upgraded...
pd:sorry for my english..im from argentina.
-
I just posted a nice response and had this thing timeout on me :-( .
So, The previous page has links to the zaptel howto and development tools required for compiling the modules. Might be an idea to drop back to the original kernel and follow the howto.
The custom bit is the word custom that exists at the top of some redhat makefiles.
i.e.
pico /usr/src/linux/Makefile
VERSION = 2
PATCHLEVEL = 4
SUBLEVEL = 20
EXTRAVERSION = -18.7custom
Hope that helps
Duncan
-
hi there.. and thanks for the howto
When i try to
rpm -Uvh http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/smeserver-asterisk-1.0-RC1.i686.rpm
i get
package smeserver-asterisk-1.0-RC1 is for a different architecture
*EDIT*
Noob here... i just sit and wait for the i386 version
-
Hi there, it's me again.
I have solved the other problem, but now when I do an insmod wct4xxp it's appears to me this error>
router:/usr/src/zaptel# insmod wct4xxp
Using /lib/modules/2.4.18/misc/wct4xxp.o
/lib/modules/2.4.18/misc/wct4xxp.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
router:/usr/src/zaptel#
thxs
-
That module is for an E1/T1 card - http://www.digium.com/index.php?menu=wildcard_te410p
Is that the card you are using?
-
Is the addmailbox command missing in the rpm or am I blind ?
/Mats
-
Its missing. I will put it - and a couple of other things in the next release.
If you add the context to voicemail.conf the mailbox will be built the first time you leave a voicemail.
-
I just run the cvs command and copied into /usr/sbin and now its up and running.
cd /usr/src
export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
cvs login (password is anoncvs)
cvs checkout asterisk
cp /usr/src/asterisk/contrib/scripts/addmailbox /usr/sbin
But thx anyway, you have done a tremendous work...
/Mats
-
FYI.
It's possible to run asterisk under windows in a CoLinux linux installation.
Download and read the files in
http://ftp://ftp.nacs.net/asterisk/astwind/
/Mats
-
FYI
Release Candidate 2 of 1.0 is released
http://ftp://ftp.digium.com/pub/asterisk/asterisk-1.0-RC2.tar.gz
http://ftp://ftp.digium.com/pub/asterisk/asterisk-sounds-1.0-RC2.tar.gz
/Mats
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@duncan,
It seems a link is missing between the following directories when u want to record the "buzy", "not here" or "name" announce in the voicemail.
ln -s /var/spool/asterisk/voicemail /var/lib/asterisk/sounds/voicemail
Isn't It ?
thks
PK
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Yep, something I need to fix on the next build - however the linking goes like this
ln -s /var/spool/asterisk/voicemail/default /var/spool/asterisk/vm
ln -s /var/spool/asterisk/vm /var/lib/asterisk/sounds/vm
Regards Duncan
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Interesting.....
DeStar is a web front end for the open source PBX Asterisk.
http://openfacts.berlios.de/index-en.phtml?title=DeStar
screenshots are very interesting.... :-D
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really nice.
Have you actually tried it?
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I am about to but support on configuration would be great... its very hard to figure out where to strat!!
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These guys are doing the same for MacOSX
http://www.sunrise-tel.com/, Í also was unable to get it to work, got stuck on the quixote install :cry:
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yeah i got suck on quixote as well, i found some good how-to's online that ill post up later on. I have my oztell and FWD acounts and i just ordered a Sipura-2000 box so next will ill begin to seriously play with it
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I think the Sipura is a much better choice than what I got, I wanted to save money and bought an used Tellian that only does h.323. If I decided to use it, I'll have to implement open-h323 and have asterisk do the translation. I've been able to play with asterisk at the moment by editing the conf files and it sure is a lot of work, if you have any progress on quixote, please let me know. Thanks!
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hi there.. and thanks for the howto
When i try to
rpm -Uvh http://mirror.contribs.org/smeserver/contribs/dthomas/smeserver/6.x/Beta/Asterisk/smeserver-asterisk-1.0-RC1.i686.rpm
i get
package smeserver-asterisk-1.0-RC1 is for a different architecture
*EDIT*
Noob here... i just sit and wait for the i386 version
I get the same error. Any work around?
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I recently setup a new SME Server and am yet to install Asterisk.
I'm just wondering, should i use the rpm, or do it the old way:)?
thanks.
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The Asterisk rpm takes care of some SME specific items. It also gets around having to install some dev tools.
For those looking for a 386 rpm - it might be quicker to build from source. I really dont have anything in the way hardware to test the builds at the moment.
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Hi fellows,
Looks like we need a good client for configuring all options of asterisk now. I've been searching for possibilities and it looks that this one:
http://www.ifrance.com/belikewater/code/actos.html
it's the most complete and gives access to all features.
I already installed it and python 2.3 but it doesn't seems to work.
Anyone have an idea why?
Best regards,
organetic
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;-) A LOT more info needed ont his one pls
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;-) A LOT more info needed ont his one pls
as you may have noticed, i was going down the lest configure asterisk via a GUI path. But now that i understand the config files i see no use for the GUI, and im the kind of person that still struggles with the Linux command line interface!!!!
There is probably no need to start talking about how to configure asterisk in this forum as there is plenty of help out there, try and find the how to guides and follow them.
This was the most helpful for me
http://voxilla.com/voxstory39-nested-order0-threshold0.html
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Duncan,
just wondering if your rpm installs asterisk 1.0.0?
If not, is it worth doing a manual install to get 1.0.0?
Anyone used it yet? I'm sure there's not a huge difference, but still interested.
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Hi,
YFYI
1.0.0 is pretty stable (current 1.0.2) and has all the basic PBX functionality. There are some differences witn the 0.7.x series, but not so much compared to the 0.9.x (mostly bug fixes)
RequestedDeletion
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Has anyone successfuly implemented the h.323 channels on asterisk? I'm trying to compile pwlib after doing flex and bison. I get stuck after ./configure when trying make:
[root@brio pwlib]# make
make[1]: Entering directory /root/pwlib'
set -e; for i in /root/pwlib; do make -C $i debugdepend debug; done
make[2]: Entering directory /root/pwlib'
Created dependencies.
set -e; make -C src/ptlib/unix debugdepend; make -C tools/asnparser debugdepend;
make[3]: Entering directory /root/pwlib/src/ptlib/unix'
g++ -DP_LINUX=2.4.20-18.7 -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -DP_USE_PRAGMA -g -D_DEBUG -DPMEMORY_CHECK=1 -DPHAS_TEMPLATES -I/root/pwlib/include/ptlib/unix -I/usr/include/pwlib -I/root/pwlib/include -M ../../ptclib/asner.cxx >> /root/pwlib/lib/obj_linux_x86_d/asner.dep
In file included from /root/pwlib/include/ptlib/contain.h:222,
from /root/pwlib/include/ptlib/unix/ptlib/contain.h:120,
from /root/pwlib/include/ptlib.h:139,
from ../../ptclib/asner.cxx:290:
/root/pwlib/include/ptlib/object.h:334:22: iostream.h: No such file or directory
/root/pwlib/include/ptlib/object.h:336:21: iomanip.h: No such file or directory
make[3]: *** [/root/pwlib/lib/obj_linux_x86_d/asner.dep] Error 1
make[3]: *** Deleting file /root/pwlib/lib/obj_linux_x86_d/asner.dep'
make[3]: Leaving directory /root/pwlib/src/ptlib/unix'
make[2]: *** [debugdepend] Error 2
make[2]: Leaving directory /root/pwlib'
make[1]: *** [libs] Error 2
make[1]: Leaving directory /root/pwlib'
make: *** [debuglibs] Error 2
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Has anyone had any experience with installing any of the various GUI configuration scripts/executables for Asterisk? Am and Destar seem to be the most likely suspects.
Cheers,
Adam
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I installed Asterisk using Duncan's RPM, but it hasn't installed vmail.cgi.
Any idea how i would go about getting this?
I know i have to use make vwebmail, but not sure where to run this.
Thanks
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create an i-bay and put the cgi file into the correct dir. vmail is an option, not a default functionality.
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create an i-bay and put the cgi file into the correct dir. vmail is an option, not a default functionality.
I don't have vmail.cgi on the server though. i have to do "make vwebmail", but i'm not sure where to do this since i never did "make install", having used Duncan's RPM instead.
Cheers.
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Just download a copy of * from CVS and copy vmail.cgi from it, it's just a script ;-)
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Got it...:)
thanks.
Now if i can work out why it is coming up this error after i log in, then i'll be happy:)
malformed header from script. Bad header=<pre>Bleh, no /etc/asterisk/vo:
I swear, i'm never chaning my server again, its so annoying to get every back to the way it was...:P
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Thanks to this wonderful post and I'm able to set up a Asterisk server within a short time on a test SME 6 server with two SIP softphones talking to each other. I'm pushing ahead to get a card to connect analog line to my Asterisk server (or SME server more specifically). Instead of rushing out to purchase the high end card, I intend to do more testing with a low end card and X100P seems to be the one. Any idea whther the X100P has the compatible driver that works with SME 6? I'm not a experience builder from source hence rpm disto based driver will be idea.
Anyone has a used/working X100P card willing to sell/ship to a country in Asia?
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There are zaptel packages here (http://ftp://ftp.nacs.net/asterisk/rh73/RPMS/). I have not tried them - prefering to compile my own.
The X100P works well enough - although there are some delays (maybe a couple of rings). Perhaps you could try ATP (http://www.austechpartnerships.com/atp/) for the card.
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I completed the How To's and everything seemed to be ok. I started testing and this is what i find.
When i connect to asterisk using
asterisk -r or -vvvvvc : I note no errors
Then i make a call from the extension i note this :
cli > WARNING [17424]: channel .C.1816 ast_request: No Channel Type registered for 'ZAP'
cli > Notice [17424]: app_dial c:696 dial_exec: unable to create channel 1 of type 'ZAP'
Does anyone know the fix for this.
modprobe zaptel = no errors
modprobe wcfxo = no errors
ztcfg -vvv = no errors
I followed the how to for zaptel and asterisk located here, house keeping and all. e-smith 6 with x100p pci card single.
thanks
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I would guess you have a problem with your Zaptel or Extension configs.
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term.c:1439: warning: assignment makes pointer from integer without a cast
make[1]: *** [editline.o_a] Error 1
make[1]: Leaving directory /home/erkang/asterisk/voicepet-single-x100p/asterisk/asterisk/editline'
make: *** [editline/libedit.a] Error 2
compile error of asterisk... does this tell anyone anything.. what can i do next
Zaptel loads on startup with no errors and shutdown dismounts correct with no errors. drivers load and everything seems ok untill i try and make a call. then i see the no registered channel for ZAP... i downloaded and tried to compile asterisk and get the above errors. when i am in cli> i note no zap commands ie... ZAP show peers say no command ZAP. Also note i have completed a search for chan.zap.so and i dont seem to have this file. i think this issue might be something like i am missing a dep. but i have installed all dep i can find relating to asterisk and zaptel.
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Thanks for the great how to and detailed posts.
I am only left with on small issue
how do i get wcfxo to run auto on startup. when every i restart the server. Zaptel is enabled and so is asterisk but nothing will work till i run.
modprobe zaptel
modprobe wcfxo
ztcfg
im sure i am missing something simple
thanks
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ok i want to try this and know this isnt a plug and play setup. wish one of the script writers would create a nice install script and a server manager template. anyway i have heard allot of good things about thisthim im going to try to set this up sometime this week.. anyone have any things they would do differently??
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I have spent some time trying to get this going. Some fresh insight into this dying thread would be appreciated.
I have a couple of soft SIP phones running internally OK, with voicemail, etc.
When I tried to get a soft SIP phone (xten) connecting from the outside, I had to first install a port opening rpm (UDP ports 3478,5060, 8000 and 8001) and open a few ports. I could then connect.
If I called voice mail I could listen to messages. If I called an internal xten SIP phone, I could hear them but they could not hear me.
Any ideas on what I am doing wrong?
Chaloner Hale
email@chaloner.ca
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The Asterisk rpm takes care of some SME specific items. It also gets around having to install some dev tools.
For those looking for a 386 rpm - it might be quicker to build from source. I really dont have anything in the way hardware to test the builds at the moment.
Duncan,
I never tried to make RPMs, but i need asterisk for my Mini ITX server. Can i use the source RPM from your build, to make my own RPM?
Per
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Chaloner
Your problem is a well known SIP thru NAT issue. The easy solution is to either use an IAX client (http://www.virbiage.com/firefly/index.php) or run some sort of Vpn.
Per
To build an Asterisk rpm - you need the source code and Dev tools (including openssl - bison). So why build the rpm - just compile (http://no.longer.valid/phpwiki/index.php/Asterisk%20HowTo) from source.
You will need to edit the Makefile to build for your processor type. Check the Asterisk forums for more details.
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To build an Asterisk rpm - you need the source code and Dev tools (including openssl - bison). So why build the rpm - just compile (http://no.longer.valid/phpwiki/index.php/Asterisk%20HowTo) from source.
You will need to edit the Makefile to build for your processor type. Check the Asterisk forums for more details.
Well, there are some RH 7.3 RPMs out there, but they fail on dependencies:
[root@perserver RH73]# rpm -Uhv asterisk-1.0-0_rh73.i386.rpm
error: failed dependencies:
libasound.so.2 is needed by asterisk-1.0-0_rh73
libgdk-1.2.so.0 is needed by asterisk-1.0-0_rh73
libgtk-1.2.so.0 is needed by asterisk-1.0-0_rh73
libodbc.so.1 is needed by asterisk-1.0-0_rh73
libpq.so.3 is needed by asterisk-1.0-0_rh73
libtonezone.so.1 is needed by asterisk-1.0-0_rh73
To me, it looks like stuff i don't need. I am not gonna use GUI or zaptel, but some voicemail i think...
Can i ignore these (or some) and install with --nodeps?
Per
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ftp://rpmfind.net/linux/freshrpms/redhat/7.3/alsa-lib/alsa-lib-0.9.0-fr0.rc7.0.rh73.1.i386.rpm
ftp://rpmfind.net/linux/redhat/7.3/en/os/i386/RedHat/RPMS/gtk+-1.2.10-15.i386.rpm
ftp://rpmfind.net/linux/redhat/7.3/en/os/i386/RedHat/RPMS/unixODBC-2.2.0-5.i386.rpm
ftp://rpmfind.net/linux/redhat/updates/7.3/en/os/i386/postgresql-libs-7.2.4-5.73.i386.rpm
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Thanks Chal
Fixed most but not all:
error: failed dependencies:
libpq.so.3 is needed by asterisk-1.0-0_rh73
libtonezone.so.1 is needed by asterisk-1.0-0_rh73
libpq.so.3: If i get a never version, then there is more dependencies.
libtonezone.so.1: For this i can only find som Mandrake stuff which i'm not allowed to download.
I don't think i need them, so i will try without.
Per
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Per
I really think you are setting yourself up for some heartache. Trust me - compiling from source is not that hard. There is no guarentee that a .386 rpm will work for you.
Duncan
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Per
I really think you are setting yourself up for some heartache. Trust me - compiling from source is not that hard. There is no guarentee that a .386 rpm will work for you.
Duncan
You're probably right Duncan :-D
But if one doesn't work then i try the other.
I always thought a .386 rpm was able to run on anything...
I think i will try to compile on a spare machine first, then i don't break too much :hammer:
I just don't understand, all the failed dependencies, won't it need them if i compile myself?
Per
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I just don't understand, all the failed dependencies, won't it need them if i compile myself?
Per
No, The dependencies are based on that particular build of the rpm - something additional that the author included. Building from source - and adjusting the dev tools to suit your version of SME will work as per the howto. However - you need to adjust the Makefile to suit your processor.
Regards Duncan
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I just don't understand, all the failed dependencies, won't it need them if i compile myself?
Per
No, The dependencies are based on that particular build of the rpm - something additional that the author included. Building from source - and adjusting the dev tools to suit your version of SME will work as per the howto. However - you need to adjust the Makefile to suit your processor.
Regards Duncan
Thanks a lot!
I will try that as soon as i get time. (Have to work too....;-) )
Per
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Per
I really think you are setting yourself up for some heartache. Trust me - compiling from source is not that hard. There is no guarentee that a .386 rpm will work for you.
Duncan
OK, i got it up running now!!
You were right Duncan, it wasn't hard to compile. I just forgot to uninstall all the crap RPMs, that i installed when i tried that RedHat RPM, so i had to do it twice. But who cares now....... :hammer:
Per
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Hi all,
After i read the post, i like to use asterisk on my server. I am new with linux so i only use rpm´s. I had install the rpm´s and they work well, now i like to use a cheap isdn card. What card should i use, i have a avm 2.0 pci card (germany), how can i install that card ? I have also a AVM PBX with S0-Bus.
Sorry for my bad english !
regards
fpausp
austria
sme 6.0.1-01 + updatescript + asterisk + avm isdn card
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fpausp, check out these forums for Asterisk help:
http://asterisk.xvoip.com
They are asterisk specific so you will get more help there.
Tristan
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Hi
just poped back to thank everyone for the help and a great forum thread.
I completed the basic install and configuration of 5 extensions. One as a remote extension for my office PC to PBX at home. The x100p card tests and works fine. I went one step further and added oh323 support and created an endpoint that now connects to a VOIP providers Gatekeeper and i can use there cheap rates to call worldwide and interact with there Voip Clients PC to PC. Asterisk answers the calls from there services and transfers to the correct extension.
Anyway just wanted to update and thank everyone for making it posible. E-smith / Asterisk rocks.
The owners of my office have played with the solution and are now interested in having it as there new office solution. Bonus For Linux - E-smith and Asterisk.
Cheers
Skydiver
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I gave this a whirl http://sourceforge.net/projects/asteriskathome/
installs centos3 taking over the whole hd much as a sme install and then installs and compiles asterisk
I then did yum update and updated the os.
I havent hammered it or added any hardware just played with sip an asterisk demo config but it seems very stable.
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Very nice, I wonder if xPL alone would work with SME. Time to build another box and try ;-)
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The hub certainly should compile on a SME, the installer includes a compiled version and installer. It fails on my SME. The service is reported to start but status says its stopped.
Got any automation project in mind ?
I would like to move my VB6 stuff to linux but have no c skills and havent seen any other language that does sockets and serial that i can grasp easily.
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I got asterisk working on my 6.0.1 server today!!!! has anybody loaded asterisk on the new 6.5?
thanks
Texasboy
:-o 8-)
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I got asterisk working on my 6.0.1 server today!!!! has anybody loaded asterisk on the new 6.5?
thanks
Texasboy
:-o 8-)
It works a treat on SME 6.5
:)
Cheers
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Careful with asterisk sources at the moment - there is an error in chan_zap.c:2879
Line reads if (p0->tranfer && p1->transfer
Spot the deliberate???
It should, of course, read
if (p0->transfer && p1->transfer
Otherwise you will get the error==>
[snip]
chan_zap.c: In function zt_bridge':
chan_zap.c:2879: structure has no member named tranfer'
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory
[snip]
Other than that it all runs fine on 6.5 provided you install the correct rpms in the correct sequence....
Just for completeness - here they are....
kernel-headers-2.4.9-34.i386.rpm
cpp-2.96-113.i386.rpm
glibc-devel-2.2.5-44.i386.rpm
gcc-2.96-113.i386.rpm
kernel-source-2.4.20-37.7.legacy.i386.rpm
bison-1.35-1.i386.rpm
bison-devel-2.0-4.i386.rpm
ncurses-devel-5.2-26.i386.rpm
openssl-devel-0.9.6b-36.7.legacy.i386.rpm
zlib-devel-1.1.4-8.7x.i386.rpm
then you can get the latest asterisk sources from CVS:
export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
CVS login (passsword=anoncvs)
cvs checkout zaptel libpri asterisk
fix the bug which you will find in
asterisk/channels/chan_zap.c
then do your make clean ; make install for zaptel then libpri and, finally, asterisk.
Phew :pint:
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I have followed this thread with interest.
Having played with Asterisk@home and being very impressed, I thought how nice it would be to have it on SME. So are you all saying it is now working? are your SME boxes setup for gateways or servers? As it seems to me (Who is stuck behind a MS ISA server that won't pass SIP) that a solution with SME with some type of firewall would be good (I am saying this out of total lack of knowledge on SME)
Interested to know the latest..
Drift.
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Imho, have SME in Server/Gateway mode and asterisk@home behind it.
If more than one system isn't an option, edit things in asterisk by hand instead of using AMP.
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Imho, have SME in Server/Gateway mode and asterisk@home behind it.
If more than one system isn't an option, edit things in asterisk by hand instead of using AMP.
Thanks for the info, I did read about the problem with AMP and having access to the shell. I really must make the time to read up on Asterisk,I assume you are saying that SME will pass the required ports? or does that require configuring on SME? Most of the problems I have had is that I have an *@home setup and we at work are behind an MS ISA server, which flatly refuses to play nice with SIP.(MS say it won't work) So it looks like your suggestion is the route to go.
Thanks again.
Drift.
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The ports would still need to be open/forward for SIP, and SIP uses a lot of ports (may be the reason they say it won't work?). But again *@home isn't limited to sip and IAX2 works right after setup, even behind the ISA Server, without the need to open ports.
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The ports would still need to be open/forward for SIP, and SIP uses a lot of ports (may be the reason they say it won't work?). But again *@home isn't limited to sip and IAX2 works right after setup, even behind the ISA Server, without the need to open ports.
Mmm very thought provoking, I just downloaded the DOcs for Asterisk proper, and will start to have a read up on IAX2, sounds like that will solve the problems with ISA.
Thanks ever so much.
Drift.
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It works a treat on SME 6.5
:)
Cheers
Do you an example of your setup you can share? (extension,sip,oh323,zapata,..., conf files)
I see you were able to load OH323 as well! (An How-To on that would be great)
I'm trying to have the same setup running on a 6.5 system, zapata gets loaded but I'm in a point where you once got stuck before ast_request: No channel type registered for 'Zap'.
Thanks,
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Good Afternoon,
Still pulling hair here due to the fact that I can't find some of the files needed by various aspects of Asterisk. The links are broke and/or the files are no longer available. The how-to and files provided by duncan are straightforward even though Asterisk has been updated to 1.2, I'm willing to settle with the older version to learn it.
I'm trying to get asterisk and amp working together. When I went through all the stuff last night, I finally got down to step 18 but ran into errors with the script apply_conf.sh when it attempted to apply some permissions to a folder in the var folder. I've reformatted my SME box for the third time and am going at it again. I'm very appreciative of all the effort by others that have more knowledge and such to get as far as we have with Asterisk. So, round the mulberry bush we go again.
Here are a few files from the AMP how-to that I'm not able to get so I went up to pre11 instead of pre4. I don't know if that will affect the whole installation or not.
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_rxfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_txfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/apps_makefile.patch
The links above are not working and/or I'm not able to get to opencall.org to get these files. Anyone know of any other location where I might be able to get them?
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Good Evening,
OK, I need the following files:
cd /usr/src/asterisk/apps
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_rxfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/app_txfax.c
wget ftp://ftp.opencall.org/pub/spandsp/spandsp0.0.2pre4/apps_makefile.patch
patch < apps_makefile.patch
These are mentioned in the AMP how-to. I've gotten all the way down and compiled asterisk but it gets the following when I execute it because I'm using pre11 files instead of the pre4 mentioned in the how-to:
[app_rxfax.so]Ouch ... error while writing audio data: : Broken pipe
Any help??
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Good Afternoon,
Arrgggh, I've tried to strip out all of the references to those files I need from opencall.org but evidently haven't found it yet. Still pecking at it.
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I sent on email!
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Good Evening,
OK, I'm stopping at this point. There are two things in the AMP how-to guide that are giving totally hellacious fits. They are both in Step 18.
#1 It states to edit the chown_asterisk.sh file with vi and execute the following command while in vi: :g/asterisk:asterisk/s//www.shared/g Well, I execute this command but also notice three lines toward the bottom as follows:
chmod u+x /var/www/html/admin/*.pl
chmod u+x /var/www/html/admin/*.sh
chmod u+x /var/www/html/panel/*.pl
I change these lines to this:
chmod u+x /var/www/html/amp/admin/*.pl
chmod u+x /var/www/html/amp/admin/*.sh
chmod u+x /var/www/html/amp/panel/*.pl
Now this is not stated in the how-to that you should do this but I'm assuming that it should be done based upon changes that I've had to make to other files up to this point. The folder /var/www/html/admin doesn't exist but /var/www/html/amp/admin does. Am I correct with changing this? Is this an oversight in the how-to?
#2 Now after modifying the apply_conf.sh file as it states to do in the how-to, I execute it and receive the following:
Unable to connect to remote asterisk
su: warning: cannot change directory to /var/lib/asterisk: Permission denied
-bash: /var/lib/asterisk/.bash_profile: Permission denied
-bash: /var/www/html/amp/admin/retrieve_op_conf_from_mysql.pl: /usr/bin/perl: bad interpreter: Permission denied
su: warning: cannot change directory to /var/lib/asterisk: Permission denied
-bash: /var/lib/asterisk/.bash_profile: Permission denied
op_server.pl: no process killed
So I think, well let me see if asterisk works and if I can connect to it manually. with the asterisk -vvc command which I seen in another contrib how-to that had a pre-packaged rpm. I believe it was duncan's. I get the following on the screen:
[chan_zap.so] => (Zapata Telephony)
== Parsing '/etc/asterisk/zapata.conf': Not found (No such file or directory)
== Unregistered channel type 'Tor'
== Unregistered channel type 'Zap'
[root@smebox asterisk]# Ouch ... error while writing audio data: : Broken pipe
Now, I can live without this zapata stuff I think as I think, not sure on this, as I believe it deals with zapata hardware and digium card stuff. I don't have any hardware installed for asterisk and will be using software phones. How can I strip this out of asterisk and/or how do I re-compile it without it or fix the above error since I won't be using the hardware anyways?
I am SO STINKING close that it is driving me insane. Any ideas would be very helpful.
On another note, at the beginning of the how-to, it lists a ton of file to wget and then install in that specific order. Well when attempting to follow this order with a simple rpm -ivh command, I found that the second and third file needed to be reversed in order. So instead of:
rpm -ivh glibc-devel-2.2.5-44.i386.rpm --- and then
rpm -ivh glibc-kernheaders-2.4-7.16.i386.rpm
it should be
rpm -ivh glibc-kernheaders-2.4-7.16.i386.rpm
rpm -ivh glibc-devel-2.2.5-44.i386.rpm
The reason I state this is that the how-to doesn't say anything about using the --nodeps option when installing the first set of files.
Well, I'm stuck at Step 18 and can't go any further. I appreciate all the help that I've received from stuntshell and others. More help and/or ideas would be much appreciated as I'm trying to get something setup now instead of waiting on the official release of SME7.
Trivia of the day: Doolittle of WWII was the only American to receive both the Congressional Medal of Honor and the Medal of Freedom.
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Good Evening,
Ok found this at http://www.wlug.org.nz/AsteriskNotes
This error is from mpg123, not asterisk. When asterisk dies, mpg123 complains. The error that killed * is either above that message or available using GDB if it is a segfault etc. You probably have a configuration error.
Still digging around trying to figure out what to do with it. When I did a ps aux I found four instances of mpg running. Rebooted the system and am still getting the same error. Well, I'll keep digging.