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Asterisk on SME7

cyr

Asterisk on SME7
« on: September 09, 2005, 05:35:42 PM »
Hi,

If some people are interrested to have asterisk on SME, go to http://firewall-services.com and test the little contrib A4SME.

Cyr

Offline edb

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Asterisk on SME7
« Reply #1 on: September 09, 2005, 09:11:52 PM »
Do you have an uninstall script as well? Thanks
......

Offline gregswallow

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« Reply #2 on: September 10, 2005, 09:37:32 AM »
Looks cool Cyr.  What hardware are you using it with?

I assume it is easier to follow how Asterisk@Home does it, but there are some rpms here for asterisk, zaptel, spandsp, etc if you want to try to make the install more rpm-based.

http://atrpms.net/dist/el4/

Offline cno

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Asterisk on SME7
« Reply #3 on: September 10, 2005, 10:10:29 AM »
just installed SME 7 beta2, and installed
Asterisk, when I click on "APM" first I get a security warning and next "The page cannot be displayed"
........................

Offline gregswallow

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« Reply #4 on: September 10, 2005, 10:11:17 AM »
cno - did you post upgrade and reboot?  If not, try that.  No offence to Cyr, but keep in mind this would definitly be in the experimental category at the moment.  If you are trying this it should be with the intention of helping Cyr find and fix bugs.

Offline StuC

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« Reply #5 on: September 10, 2005, 11:57:05 AM »
I had a few failed attempts (Kernel Panics)
but that may be due to my old hardware, of the fact I was underclocking the PII 450, went in at third attempt. (I you hit enought time maybe it will work method):hammer:

One problem them was that I was not using the SME box (private server mode) as the proxy server or dns so the AMP page would not display from the server-manager page.

All installed and running now, one softphone (X-Lite) will register but wont accept calls, another hard phone (Grandstream GXP2000) wont register yet.

I am completely new to Asterisk so sure probs will be my lack of knowledge.
Thanks for the contrib, sure it has loads of life in it and I look forward to getting it going.

I see SugarCRM is installed but have not worked out how to get to that other than through the server-manager page yet.
Just found I can get to SugarCRM From
https://server.domain.etc/server-manager/amp/crm/
not sure how users without access to the server page will fare

Offline cno

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« Reply #6 on: September 10, 2005, 12:08:25 PM »
Quote from: "gregswallow"
cno - did you post upgrade and reboot?  If not, try that.  No offence to Cyr, but keep in mind this would definitly be in the experimental category at the moment.  If you are trying this it should be with the intention of helping Cyr find and fix bugs.


the first time I tried no, then I Do post upgrade and reboot
stil the same error.
........................

janvantil

downloading Asterisk/A4SME.tar.gz
« Reply #7 on: September 11, 2005, 01:26:03 AM »
I tried to download the file Asterisk/A4SME.tar.gz at the site, but the server refused me. Will it also work on sme 6.01 ??

cyr

Asterisk on SME7
« Reply #8 on: September 11, 2005, 05:56:43 PM »
Hi,

There seems to be a lot of questions and I'm not sure I can answere to all of you but I will try.

     gregswallow wrote :

     Looks cool Cyr. What hardware are you using it with?

I use soft (xlite) and hardphone with a sip privider for french people name wengo :-)

      I assume it is easier to follow how Asterisk@Home does it, but there are some rpms here for asterisk, zaptel, spandsp, etc if you want to try to make the install more rpm-based.

Yes, I know, but I wanted to use the great job of the Asterisk@Home team

       cno wrote :
       
       just installed SME 7 beta2, and installed
Asterisk, when I click on "APM" first I get a security warning and next "The page cannot be displayed"

Did you have a good name resolution of your sme test server from your client?
Take a look at /root/install_asterisk.log on your server, you should see at the end "asterisk started", if not try to understand why it did not start whith the log file or try to reinstall in a different connfiguration

       janvantil  wrote :

       I tried to download the file Asterisk/A4SME.tar.gz at the site, but the server refused me. Will it also work on sme 6.01 ??

I don't no why the server refused you, perhaps you should try again later.
And it will not work on sme6 because sme6 is based on rh73 with kernel 24.

Offline soprom

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Asterisk on SME7
« Reply #9 on: September 11, 2005, 10:00:19 PM »
Thanks for your efforts!

I installed on a fresh SME7b2 and also receive a blank page.
There was a message saying that amp did not start but I can't retrace it in the logs.

Also when setting permissions:
Code: [Select]

SETTING FILE PERMISSIONS
chown: cannot access /var/run/asterisk': No such file or directory
chown: cannot access /var/log/asterisk': No such file or directory
chown: cannot access /dev/zap': No such file or directory
Permissions OK
Done
touch: cannot touch /var/log/asterisk/cdr-csv/Master.csv': No such file or directory
chmod: cannot access /var/log/asterisk/cdr-csv/Master.csv': No such file or directory



I read from asterisk@home that a switch has to be set for installation on PII instead of PIII machine. That might cause a problem in my case since I'm running SME on a PII/333, 192MB-RAM

Where is the log for asterisk/amp ?

I also tried:
signal-event post-upgrade
signal-event reboot


From the install-log:
Code: [Select]
installing Zaptel Driver
-------------------------------------------
cp: cannot create regular file /usr/src/linux-2.6.9-11.EL/.config': No such file or directory
install_asterisk.sh: line 24: cd: /usr/src/linux-2.6.9-11.EL/: No such file or directory
mv: cannot stat Makefile': No such file or directory
sed: can't read Makefile.back: No such file or directory
make: *** No rule to make target modules_prepare'.  Stop.
Makefile:171: target ztdummy.o' given more than once in the same rule.
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE    -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o gendigits.o gendigits.c
make: cc: Command not found
make: *** [gendigits.o] Error 127
Makefile:171: target ztdummy.o' given more than once in the same rule.
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE    -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o gendigits.o gendigits.c
make: cc: Command not found
make: *** [gendigits.o] Error 127
Makefile:171: target ztdummy.o' given more than once in the same rule.


and a little further in the log...

Code: [Select]

installing libpri driver
-------------------------------------------
gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g     -c -o copy_string.o copy_string.c
make: gcc: Command not found
make: *** [copy_string.o] Error 127


Code: [Select]

Checking current version of AMP..1.10.007
Installing new AMP files..cp: cannot stat /etc/asterisk/voicemail.conf.template': No such file or directory
OK
Configuring install for your environment..sed: can't read /etc/asterisk/cdr_mysql.conf: No such file or directory
sed: can't read /etc/asterisk/cdr_mysql.conf: No such file or directory
sed: can't read /etc/asterisk/manager.conf: No such file or directory
sed: can't read /etc/asterisk/manager.conf: No such file or directory
sed: can't read /etc/asterisk/vm_email.inc: No such file or directory
/usr/src/AMP/apply_conf.sh: line 67: /usr/sbin/amportal: Permission denied
OK
Setting permissions on files..chown: cannot access /var/run/asterisk': No such file or directory
chown: cannot access /var/log/asterisk': No such file or directory
chown: cannot access /dev/zap': No such file or directory


So I guessed I needed to install gcc (!!)
Install went better but it's a no-go to open the web interface:
Code: [Select]

127.0.0.1 - admin [11/Sep/2005:19:13:54 -0400] "GET /server-manager//cgi-bin/pleasewait?/server-manager/cgi-bin/amp HTTP/1.1" 200 390
127.0.0.1 - admin [11/Sep/2005:19:13:56 -0400] "GET /server-manager//cgi-bin/amp HTTP/1.1" 200 407
127.0.0.1 - admin [11/Sep/2005:19:15:22 -0400] "GET /server-manager//cgi-bin/amp HTTP/1.1" 200 407
Sophie from Montréal

cyr

Asterisk on SME7
« Reply #10 on: September 12, 2005, 11:20:16 AM »
Quote
I installed on a fresh SME7b2 and also receive a blank page.


You should check if you can access your server with the name and not with the IP adress

guest22

Asterisk on SME7
« Reply #11 on: September 12, 2005, 12:57:10 PM »
The topic of this post should be ' problems with AMP'  not asterisk. Asterisk itself runs perfectly. All above 'errors'  are AMP related.

Offline Franco

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« Reply #12 on: September 12, 2005, 09:16:38 PM »
It should read "I think I found a bug". These are my experiences:
-Tested in 03 different systems and they all work. The problem seems that 7b2 is not resolving the hostnames correctly and therefore "The page cannot be displayed"
-The issue with the Pentium II and Pentium III are codec related, so it's not about your machine StuC.
-I tested on a Pentium II 350mhz/196ram, Pentium Celeron 587mhz/128ram and a Pentium III 500 mhz/512ram. VoiP equip. used to test was Artdio phone, Sipura and Xlite, X100P card.
-Amp can be accessed in all of them through http://sme/server-manager/amp/ but not thru Panel, which wants to go to http://www.smedomain/server-manager/amp
-Couple of things Post Upgrade is required to work and Zaptel drivers had to be compiled by hand.
-Thank you Cyr

Offline cno

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Asterisk on SME7
« Reply #13 on: September 12, 2005, 09:59:59 PM »
Hi
I tried also to access outside server-manager with no luck

then I tried to do this
https://beta-server/server-manager/amp/
then I have access to amp

and not this way
http://beta-server/server-manager/amp
(need the last / after amp)
........................

Offline gregswallow

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« Reply #14 on: September 13, 2005, 12:03:56 AM »
Quote from: "stuntshell"
-Couple of things Post Upgrade is required to work and Zaptel drivers had to be compiled by hand.


There are Zaptel drivers here, already compiled:
http://atrpms.net/dist/el4/zaptel/

cyr

Asterisk on SME7
« Reply #15 on: September 13, 2005, 10:14:31 AM »
Hi,

Because there seems to be a lot of trouble to access amp perhaps you can guive me a hand. The only way to access the panel that I found was to make a redirection

look at /etc/e-smith/web/functions/amp

But I'm not a really good perl developer and if someone have an other idea that would be great.

Quote
Couple of things Post Upgrade is required to work and Zaptel drivers had to be compiled by hand.


I'll try to do something for this  :lol:

Offline Franco

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« Reply #16 on: September 13, 2005, 05:36:41 PM »
Thank you Cyr,
I think that just by adding the backslash at the end of the line already helps a lot. Now, I don't know for sure if it's a bug, but my server does not resolve to www.domain, which the scripts asks.
A couple of things I did is tested AAH 1.3 and 1.5 on the same machine, 1.3 has a hard time detecting the zaptel card (genzaptelcfg required) where 1.5 does it by itself.
So I downloaded the 1.5 installer and I'm trying to make it do the work like your installation. I have already failed the first one, and I'm reinstalling 7b2 (better fresh, to continue.
1.5 is compacted in few installer scripts, which may be easier to modify (not for me so far  :hammer:

Offline Franco

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« Reply #17 on: September 13, 2005, 05:45:10 PM »
Quote from: "gregswallow"

There are Zaptel drivers here, already compiled:
http://atrpms.net/dist/el4/zaptel/


Great site, thanks a lot Greg!

Offline gregswallow

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Asterisk on SME7
« Reply #18 on: September 13, 2005, 09:38:04 PM »
Quote from: "stuntshell"
Quote from: "gregswallow"

There are Zaptel drivers here, already compiled:
http://atrpms.net/dist/el4/zaptel/


Great site, thanks a lot Greg!


No problem.  Just a suggestion if you're going to try and modify A@H 1.5 - They install most everything from source because it is a single purpose server and they don't care about removing the files or upgrades, etc.  For SME, everything should be installed with rpms.  Whatever parts of the script you can remove by inserting 'rpm -Uvh xxx.rpm' instead of the steps to compile from source, do it.

cosy

VoIP Newbie
« Reply #19 on: September 14, 2005, 08:21:55 AM »
HI,

  I just like to test Voip in my SME 7.X Box.
but can someone explain to me these questions?
totaly new to this voip. :hammer:

1. To test Voip, do i need to buy any hardware/ software?

2. Can i use normal ADSL 256/64 with TPG australia?

3. What about the telephone bill for local & international cal cost and all?



thanks

cosy

VoIP Newbie
« Reply #20 on: September 14, 2005, 08:22:31 AM »
HI,

  I just like to test Voip in my SME 7.X Box.
but can someone explain to me these questions?
totaly new to this voip. :hammer:

1. To test Voip, do i need to buy any hardware/ software?

2. Can i use normal ADSL 256/64 with TPG australia?

3. What about the telephone bill for local & international cal cost and all?



thanks

cyr

Asterisk on SME7
« Reply #21 on: September 14, 2005, 11:05:15 AM »
Hi Cosy,

Quote
1. To test Voip, do i need to buy any hardware/ software?


No, you just need to install a soft phone as xlite on your client machine.

Quote
2. Can i use normal ADSL 256/64 with TPG australia?


Only if you need 1 or 2 simultaneous communication or if you buy the g729 codec license.

Quote
3. What about the telephone bill for local & international cal cost and all?


I don't know, check your voip privider.

cosy

Asterisk on SME7
« Reply #22 on: September 19, 2005, 05:51:20 AM »
HI,

 Thanks. but still not clear for my Q3. To make voip calls, do i need to get authorized from my ISP?

if so why we need voip server? they normally provide voip package now.

What is the different betwen our SME voip server and the free voip providers?

Anyway tell me someone how i can reduce my home phone bill :-D

Offline Franco

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« Reply #23 on: September 19, 2005, 06:53:33 AM »
With Asterisk on SME you'll have your own voip server, and provider if you will(if you could have gateways everywhere and bandwidth, etc.). Then you can use it to make LAN calls, or even over the internet if you open the right ports or use VPN.

Quote
Anyway tell me someone how i can reduce my home phone bill

There are too many ways to list here, the options are numerous!

cosy

Asterisk on SME7
« Reply #24 on: September 20, 2005, 01:37:28 AM »
HI, Thanks.

   If you can explain 1 or 2 methods we can use to reduce our home phone bills thats going to be good help for everyone i think.

Anyway What do I need to have a VoIP Line?

 I think i need to have some sort of a free VOIP Provider (ISP) to connect?

Offline Franco

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« Reply #25 on: September 20, 2005, 04:51:55 AM »
Cozy,
Methods on how to reduce phone bills:
1- Free calls over the internet
2- Lower rates when calling anywhere in the world.

Asterisk + SME is a great combination in a sense where you don't need to be running 02 systems. The advantages of this set up, be for home user or small business are numerous, and I recommend checking http://www.voip-info.org for all the needed info.

In my case I use at home and at the office. At home I have my own PBX (how fancy!) and calls comming in can be answered on the regular phones or using desktops, laptops (running Xlite) or IP Phones. They can be upstairs, downstairs or anywhere my wireless setup allows me. I have my own attendant that greets my guests and directs them to the right extension. They also can be put on hold, transferred, or go into virtual meetings, all thru an automated system that includes my favorite background music.

To make local calls, we use the regular PSTN system, long distance gets routed to a VoIP provider, 1-800 calls are routed to http://www.FreeWorldDialUp.com that gives me free calls anywhere in the world.
Incoming calls are taken as if I only had one number/line, when in fact I use Gismo http://www.gizmoproject.com , FWD, my e164 number, my local number and my Voip provider number.

Calling home or office when I'm at a client (or anywhere in the world actually) is free with the help of VPN, I could also use FWD or anything without the need of VPN, but I like quality and I can also make calls as if I were dialing from the office or home, when I'm not physically there, paying as if it was a local call.

There's also flexibility, control, reporting and other things that make asterisk great. The price to implement a perfect system (including PSTN cards, IP phones and your time of course) can be high initially, but it's a good investment in my view.

Enough of me already! ;-)

Offline torrestech

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VOIP TPG australia
« Reply #26 on: September 21, 2005, 10:56:12 PM »
Hi.
I run VOIP on my own network. At this time i am not using ASSTRIX but just a hardware phone adapter. Plugs into my existing PABX and gives me 2 voip lines. Works well. Just needed to install QOS from Swertz Knudsen's website and find a good VOIP provider. I use siphone. www.siphone.com.au which gives me 1 cent per minute calls anywhere in australia.
Regards,
Adam
...

matsk

Asterisk on SME7
« Reply #27 on: September 23, 2005, 11:12:59 AM »
Quote from: "gregswallow"
Quote from: "stuntshell"
Quote from: "gregswallow"

There are Zaptel drivers here, already compiled:
http://atrpms.net/dist/el4/zaptel/


Great site, thanks a lot Greg!


No problem.  Just a suggestion if you're going to try and modify A@H 1.5 - They install most everything from source because it is a single purpose server and they don't care about removing the files or upgrades, etc.  For SME, everything should be installed with rpms.  Whatever parts of the script you can remove by inserting 'rpm -Uvh xxx.rpm' instead of the steps to compile from source, do it.


Isn't the prefered installation source:
1. YUM
2. RMP
3. Sourcecode

/Mats

kangkc

Asterisk@Home too heavy for SME!
« Reply #28 on: September 25, 2005, 04:20:23 AM »
I have been following this thread to see where it's heading. I have both Asterisk@Home 1.5 and SME 6.x implemented in my SOHO environment and it has been working fine. Ging forward instead of running from two servers, I would like to eventually combine this into a single server to save on hardware upgrade cost.  

I'm not with the idea of moving Asterisk@Home into SME as I feel it's way too heavy for SME. In fact I feel that it's like putting SME into Asterisk@Home.  

What I'll be trying to do it to install Asterisk on it's own in SME without AMP, copy the config files from AMP generated over to SME. Going forward I'll be looking into using IPManager or other Asterisk configuration software to configure the server.
Of course what I'll be missing is the nice reporting that AMP provides which at the moment the least priority. Prety sure this can be managed and collected out from Asterisk with other tools.

At the end of the day, SME Server is to provide a secure, stable platform for production environment and I don't think I want to change this.

matsk

Re: Asterisk@Home too heavy for SME!
« Reply #29 on: September 25, 2005, 06:46:34 PM »
kangkc Could you clarify ! First you say :
Quote from: "kangkc"
I would like to eventually combine this into a single server to save on hardware upgrade cost.

And then :
Quote from: "kangkc"
I'm not with the idea of moving Asterisk@Home into SME as I feel it's way too heavy for SME. In fact I feel that it's like putting SME into Asterisk@Home.

Do you wan't to add asterisk into your SME server or not ?

And lets keep our reasoning based upon facts not "feelings".

When choosing strategy, one server with all services or several servers with different services should be based upon facts like, hardware, application work load, personell experience and so on.

To mix A@H in SME isn't that big contradictiction, they are based upon the same strategy, a web interface that controlls the underlying work horses/services.

And I don't understand your reasoning about putting SME into A@H or vice versa. Adding * onto a SME server can be disastrous if you dont have hardware that can keep up with the workload because * needs resources when it requests it but this is if you have calls that is teminated into * or that * is a media gateway.

And to get hard facts I always include MRTG or rrdtools on my systems so I can see the trends on the workload and from that I can take descissions about upgrades, investments or strategy changes.

And to add AMP & FOP as panels into SME is a neat way to include a application fast and to avoiding the need to convert the PHP/Flash to Perl CGI (SME interface).

I'm running SME and A@H on the same machine for my SOHO (4 users) without any problems on a 1,7GHz, 256M RAM machine.

Quote from: "kangkc"
At the end of the day, SME Server is to provide a secure, stable platform for production environment and I don't think I want to change this.

Trus me Asterisk is as important as the SME server and is as stable and secure!

I'm also looking into XEN that will realy give you the tool to controll your resources and you can maximize the utilization on your resources.

/Mats

kangkc

Asterisk on SME7
« Reply #30 on: September 26, 2005, 03:12:20 AM »
To put it in perspective, yes Asterisk on SME is ultimately what I want and I have done that some time back with a rpm contribs from Duncan on a test machine, minus FOP, AMP and the rest. Just plain Asterisk.

But to just use the install script from Asterisk@Home onto SME is probably a no for me as most of the distributions are not rpm and templates based. Putting dev tools on production server is also not recomended.

Anyway, this is strictly my personal opinion. I guess end of the day you can decide what you want to do with your SME server and this just the way I see it; sticking to fundamental SME guidelines, rpm and templates.  

Lastly, what Dungog.net have done with Asterisk is to me a good start.  

I rest my case here.
[/quote]

Offline arne

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Asterisk on SME7
« Reply #31 on: September 27, 2005, 12:37:36 AM »
Thanks for a great work.

I used the automated installation from the link in this tread and it just went right just in. (Whith a few minor bugs already mentioned above.)

No I just wonder how to use it in a practical way ..

As I thought this was a little different qestion so I started a new tread on how to use it.

By the way .. I could never have made this installation without this automated script, so thanks a lot !!

Arne.
......

matsk

Asterisk on SME7
« Reply #32 on: September 27, 2005, 02:31:26 AM »
Quote from: "kangkc"
But to just use the install script from Asterisk@Home onto SME is probably a no for me as most of the distributions are not rpm and templates based. Putting dev tools on production server is also not recomended.

A minor correction, Asterisk@Home is a mix of RPM's and source code. Regarding the dev tools you have the option to remove them after you have completed the installation of A@H and thereby avoid the security issues with having dev tools installed.

But in generall I agree that they should be avoided and sources from YUM is prefered not RPM.


<IMHO>
Mats

Offline arne

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Asterisk on SME7
« Reply #33 on: October 02, 2005, 12:30:45 AM »
I made one other question, on this board, to find out what i can be used for .. after a few days of thesting I think I have found out that .. it is something like a local ip telephone telephone sentral. You can set up telephoned to internal ekstensions, call between local telephones, record and listen to messages, etc ..

I think this configuration guide is very good:
http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm

I have installed Asterisk on the sme gateway according to info in this tread and I have also set up one dedicated Asterisk server based on their Centos bundle.

I think the gateway (sme) and the server (centos) installation of the Asterisk installation works quite simular, but I have one BIG problem.

I can get the local lines to work and the "syntetic voices" on the server and so on, but I have not been able to configure a "trunk" (connection out) in such a way that I have been able to call out. Among other I have tried to connect to the free pulver.com server to call their free time info on 612. It should work, but it doesn't.

Have any of you made succsessfully configurations for calling out ? Any ideas / advices ?

Best reg Arne.
......

Offline psoren

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Asterisk on SME7
« Reply #34 on: October 02, 2005, 11:45:27 AM »
Quote from: "arne"
I made one other question, on this board, to find out what i can be used for .. after a few days of thesting I think I have found out that .. it is something like a local ip telephone telephone sentral. You can set up telephoned to internal ekstensions, call between local telephones, record and listen to messages, etc ..

I think this configuration guide is very good:
http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm

I have installed Asterisk on the sme gateway according to info in this tread and I have also set up one dedicated Asterisk server based on their Centos bundle.

I think the gateway (sme) and the server (centos) installation of the Asterisk installation works quite simular, but I have one BIG problem.

I can get the local lines to work and the "syntetic voices" on the server and so on, but I have not been able to configure a "trunk" (connection out) in such a way that I have been able to call out. Among other I have tried to connect to the free pulver.com server to call their free time info on 612. It should work, but it doesn't.

Have any of you made succsessfully configurations for calling out ? Any ideas / advices ?

Best reg Arne.


Hi Arne,

Do you have a router in front of your servers, in that case you need to open some ports for asterisk. For SIP it's UDP 5060. If more SIP accounts then, 5061 and so on. Maybe you also have to open ports UDP 10000 to 20000 for RTP.
I have used both my SME server and Asterisk@Home and they both worked well. I used a danish company called Musimi (Owned by Telio in Norge :-) )
Have you tried the combination of SME7 and Asterisk@Home?

Per

Offline arne

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« Reply #35 on: October 02, 2005, 01:56:02 PM »
About NAT: According to some info on the net forwarding should not be required. I think that the real and true answer in that case is that forwarding will be required some timens and some times not, depending on the server technology of the ip telephone vendor. I have tried different clients and also to dump the trafic with ethereal, and I think they do not do it the same way, allways.

I have made one installation of Asterix@Home at my SME 7.0 Gateway and one other at a PC on lan. To day I have finally managed to hack my adsl router, so it is not a router any more but a bridge. The SME gateway now receive the external ip.

I think it is right to say that a central part of the functionality of the Astersis server is to work as a sip telephone proxy server and a a local sip telephone switcboard. In this way it has the nature of a client releative to the external sip server you are connecting to and as a server for your local sip telephone. In this way it lookes like it can "translate technology standards" so you can have one internal protocol standard in your home/office while different oubound lines (trunks) can run different protocol standards (Also at Asterisk@home, or will it be needed "Asterisk normal" ??) while your clients have only one communication setup.

I think that if you have only one ip telephone you will have rather little use for a Asterisk proxy/telephone switchboard. You can connect your telephone client directely to the external server or via the Asterisk proxy/switchboard. There should not be the big differences. (exept for some increased functionalities of the Arsteisk like telephone ques, music while you are waitning, anwsering machine etc.)

One other big and important reason will of cource be if you want to learn the new technology.

Because of "the client nature" of the Asterisk "proxy server" I believe that the case will be that if you will have to make a forwarding to make a client to work you will also have to do the same if you are running a Asterisk server at lan or a kind of dmz.

On the other hand, if you are running the Asterisk on a gateway, it might be situations where you also will have to open aditional ports on the gateway to make it work.

I use the norweigan sip telephone wendors Telio and Televoip but I have not come so far, yet that I have been able to connect to their sip servers via Asterisk.

I have until now just tried with some free services like pulver / freeworlddialup. This does not work for me so far.

I just try to explain things in such a way that I am able to understand it myself  :hammer: Please correct me if I should be wrong.

By the way, I think the SME version of the Asterisk@home and the orginal downloadable "Centosbundle" work rather the same way. Initially revision status seems to be approx the same, but the "Centosbundle" did a lot of updating using the command "yum uptdate".

Hopefully "someone" will correct me if I'm wrong about "technology and understanding". (For me one or two ip telephone clients will be enough so the Astersik server part of it is mainly for learning - and learning that will normally be to do some mistakes and inncorrect conclusions on the way.
......

Offline arne

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« Reply #36 on: October 02, 2005, 02:27:37 PM »
I have tried to set up a "trunk" (extension out) via freeworld according to this guide:

http://www.voip-info.org/wiki/index.php?page=AsteriskAtHomeFWD

For some reason it does not work for me. I can just hear a "syntetic voice" of the Asterisk server that says something like "All cirquits are bussy, please try later".

In outbound routing, dial paterns I just fill inn 0|. expecting that 0612 should give me the 612 watch function of freeworld, but it does not work. (Have tried with 0|. and nothing at trunk configuration, Outgoing Dial Rules.)

I must admit that I have not opened aditional ports on the gateway Asterisk@home or forwarded ports to the lan Asterisk@home installation. (Due to the fact that the ordinary freeworld client does not need any portforwarding.)

Anybody that have tried to set up the freewold sip connection and that have made this work ?? (I wonder if the config guide is for an older version of Asterisk@home, not the 1.3 version.)

Some other config guides:
http://www.voip-info.org/tiki-index.php?page=Asterisk+at+Home+Examples

Best reg Arne.
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Offline arne

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« Reply #37 on: October 02, 2005, 02:49:26 PM »
According to the commersial sales arguments this telephone can be connected to your lan telephone server or to the same telephone server from whereever you are in the world where it is a open wlan. (So that the telephone costs from airports etc will be zero.) I guess that it is right that the data transport to the "local telephone server" can go via lan or via internet (provided proper firewall configuration.)

I guess that telephones "home" will be for free to the local extensions while "telephone calls out" will be with the same prices as "when home" (If your Asterix is connected to a "external sip server".)

http://estation.dk/product.asp?product=83&sub=446
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Offline psoren

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« Reply #38 on: October 02, 2005, 03:58:26 PM »
Quote from: "arne"
According to the commersial sales arguments this telephone can be connected to your lan telephone server or to the same telephone server from whereever you are in the world where it is a open wlan. (So that the telephone costs from airports etc will be zero.) I guess that it is right that the data transport to the "local telephone server" can go via lan or via internet (provided proper firewall configuration.)

I guess that telephones "home" will be for free to the local extensions while "telephone calls out" will be with the same prices as "when home" (If your Asterix is connected to a "external sip server".)








http://estation.dk/product.asp?product=83&sub=446


Well, if everything you write in your last tree posts are correct, i can't say..... but it looks like you got the picture...
I initially started with an Asterisk installation on my SME server/Gateway which worked fine. I then tried the Asterisk@Home and i had a lot of trouble until the current version came out. It then worked fine and i even got a X100 card installed (clone card) and got my PSTN line to work through Asterisk, cool... It just didn't give me the caller ID. I have then lately installed it with SME7 and just loaded the back up file i had from Asterisk@Home and it then worked fine again. How ever i have also experienced the "All lines are busy now try again later" problem.
I hope there will come new versions off Asterisk@Home that will solve those two things, and then i will start to play with it again when the cold, dark and rainy season hits us again.
Have you had a look at the forums at musimi.dk? There is a lot of good stuff there (It's all in danish).

Per

Offline arne

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« Reply #39 on: October 02, 2005, 04:55:02 PM »
".. and i even got a X100 card installed (clone card) and got my PSTN line to work through Asterisk, cool."

Ok, so this x100 card is one of your "trunks" (or outgoing connections from the server) ?!

When I red the info about the Asterisk server I got the understanding that such cards could be used only for connecting traditional telephones without a sip adapter directely to the PC.

But it can also (or only ?) be used to set up your own trunk/bridge from ip telephony to tradisjonal old telephone lines ??!!

By ther way, thanks for info !

Arne.
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Offline arne

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« Reply #40 on: October 02, 2005, 05:07:28 PM »
I found some info in some other tread that I copy over for further referance. If the Asterisk server is on lan, then ports can be forwarded as/if neccessary via the server-admin panel. If the Asterisk is running on the Sme gateway, I guess there might be situations where it will be neccessary to open ports on the gateway for access to server functions on the gateway server. I just copy over this info from the other tread:

#mcedit /etc/e-smith/templates/etc/rc.d/init.d/masq/42AllowTS
add the folloing:
/sbin/iptables --append INPUT -p udp --dport 8767 -i $OUTERIF -j ACCEPT
followed by:
#/sbin/e-smith/expand-template /etc/rc.d/init.d/masq
#/sbin/e-smith/signal-event reboot

(How to open udp port 8767 to the local server functions at the gateway server.)
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Offline Franco

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« Reply #41 on: October 02, 2005, 05:23:30 PM »
Arne, it would be too difficult to troubleshoot your set up, if you're getting the asterisk prompt it means you forgot something (DID?).
You don't have to open any ports, unless you want to be a provider (you want other people to connect to your asterisk)

These tutorials here work: http://geekgazette.com/index.php?option=com_content&task=category&sectionid=4&id=13&Itemid=31
Different asterisk@home versions (1.1,1.3,1.4,1.5...) require different settings, but they're quite alike.

Offline arne

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« Reply #42 on: October 02, 2005, 08:10:13 PM »
Thanks for the link !

I have set up nothing else than the standard things found on the guides on internet, yet, so it should be not to difficult. Also deleting one and one setup to to only one certain test at the time.

I think the need of opening incomming ports will be required if you want to call your own server via internet. Lets say you log into your own server and call extension 200 in the living room or 201 in some other room. Of cource all calls to your home/office will be for free for instance with this telephone that is also free:  http://www.xten.com/ (Have not checked it out yet, but I believe that this should be an easy one. There might though be some security things to think about.)

Other potencial needs for opening/forwarding ports will be if you use a sip telephone vendor that use a some "unstandard" connection. (I think one of my vendors do that.)

Best reg Arne.
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Offline arne

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« Reply #43 on: October 02, 2005, 08:51:24 PM »
Ref to the link above.

This Gizmo alternative looks rather interesting.

The article about the configuration is rather new, and the rates for calling "the old telephone net" is not to bad. Nothing to pay per month.

I think this should be tested:

http://geekgazette.com/index.php?option=com_content&task=view&id=33&Itemid=31

http://sipphone.com/

http://sipphone.com/minutes/rates.html

One other thing .. I think such testing projects for sip telephones arrangements should be done as a "pay in advance" service and not "pay per bill" due to security resons (If somebody should be able to empty your 10 dollar, jo can live with that.)

Thanks for info !

Best regards Arne.
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Offline arne

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« Reply #44 on: October 02, 2005, 08:54:33 PM »
Look at this. Not to bad for us testers (if it works).

http://sipphone.com/5free1/
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Offline arne

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« Reply #45 on: October 03, 2005, 12:11:14 AM »
The Gismo setup did work. I came first out to the Gismo network, and then out to telephones in Europe within my free 5 min limit. Had to pay nothing to get it up and running.  :hammer:  Think I will buy an inbound number and some more call credits ..

All the small details in the login info had to be right, so I had to experiment a bit.

Please leave a question if you want to login and if you get problems.
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Offline arne

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« Reply #46 on: October 03, 2005, 05:46:18 PM »
Configured and connected to broadvice.com as well.
http://www.broadvoice.com/

Used this guide. It worked ok for both outgoing and ingoing telephones:
http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm

Problem: I think these server/trunks is located in the USA. The prices for calling are rather low, but the quality of the sound is a bit poor with some delay. It is cheap enough but not good enough for making local calls here in Norway.

I were not able to do a full test on the Gizmo connection. My credit card did not work on their site, so I could not buy a number. My impression is still that the Gizmo project has a bit bether sound quality.

Is Asterisk a good and reasonable practical installation for the sme server ? If you want to learn and know the technology, I think it is a great installation. (and for such a use the small bugs does not mather.)

If you want to make a best as possible and practical use of ip telephony I think you will not need a Asterisk at all, It will be much more easy and practical solution to set up ip telephones on the lan and then to connect directely to external servers. (I think the standard and unmodified sme server works exelent for such a purpose.)

Personally I will still use the Asterisk installation for a while, but it is not because I like to speak in the telephone.

Arne.
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Offline arne

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« Reply #47 on: October 03, 2005, 07:52:36 PM »
I hope that anybody else in some project or circumstanses will find this info usable or interesting ..

I have now just checked out the theory that the Asterisk server works in such a way that "your local switchboard" can be also accessible via internet.

By default the sme 7.0 firewall will block for external login to the Asterisk server. If you open for the proper ports of the firewall, your local ip telephone clients can be anywhere in the world, they do not need to be at your lan.

Just tried now to download the X-lite client on a web cafee. No problem to log on to the Asterisk server at home and to call local numbers for free and to call out to "old telephones" via the Broadvoice or the Gizmo connection.

One of the strong sides of the Asterisk server is that it can handle many outgoing sip telephone connections, and then you for instance can configure it to select the cheapest (or best) connection for each country. It is also passibele to set up a password for each connection out. (If you open for external telephone it will be rather neccessary.)

After having tested a bit with one Aterisk on the lan and one at the gateway it its my impression that the "sme variant" at the gateway performes bether than the "orginal Asterisk bundle at the lan. (But it could also be a qeustion about a bit poor hardware.)

Thanks a lot to Cyr that made this modification for the sme server and that made all theese new "ip telephone alternatives" possible on the sme gateway.  :hammer:  

(But it is a rather big mod and it changes quite a lot on the sme gateway. It is not completely the same after the mod.)
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Offline Franco

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« Reply #48 on: October 04, 2005, 07:07:59 AM »
Quote
(But it could also be a qeustion about a bit poor hardware.)

Codecs not only will give you better usage of your bandwidth, but also better performance from your machines (Some people say 50%  :-o ):

http://www.aussievoip.com.au/tiki-index.php?page=G729

Offline arne

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« Reply #49 on: October 04, 2005, 02:06:08 PM »
Psoren ->

"It then worked fine and i even got a X100 card installed (clone card) and got my PSTN line to work through Asterisk, cool... It just didn't give me the caller ID"

Does this mean that you did set up a trunk for the PSTN ?? (How to do that !?).

stuntshell ->

I have allways thought about the codex had to do with something like or simular to encoding and decoding with the only right (or wrong) codex. Could a codex be "more ore less good" as long as you dont have controll over the coding in the orher end ? (I don't know .. but possibly it could be like that.)

Also I have allways thought about the use of coding / use of codex is something that happen at the client only. Right/wrong ?
 
If you receive a steam of udp packets that contains a sound signal, could the sound quality be more or less good depending on the codex you use at the client side ? (Have been thinking rather little about the codex problem before, and that might be "wrong" ..)

Arne.
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Offline psoren

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« Reply #50 on: October 04, 2005, 02:28:44 PM »
Quote from: "arne"
Psoren ->

"It then worked fine and i even got a X100 card installed (clone card) and got my PSTN line to work through Asterisk, cool... It just didn't give me the caller ID"

Does this mean that you did set up a trunk for the PSTN ?? (How to do that !?).



Arne,

The X100 card allows you to recieve PSTN calls. That means you can recieve PSTN calls into the asterisk server. There are some tasks in the Asterisk@Home command line to install the card. You don't have to set up anything extra for it as it is there by default, it's the Zap gateway. It will just send the call to your default trunk. I don't have asterisk installet at the moment since i am using my testserver for other things so i don't remember things in detail.

Per

Offline Franco

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« Reply #51 on: October 04, 2005, 02:58:24 PM »
Installing the X100P should be easy, genzaptel on asterisk@home is a script that does it all for you. On the combination SME+A@Home you can modprobe wcfxo, then zaptel followed by ztcfg -v.
Quote
I have allways thought about the codex had to do with something like or simular to encoding and decoding with the only right (or wrong) codex. Could a codex be "more ore less good" as long as you dont have controll over the coding in the orher end ? (I don't know .. but possibly it could be like that.)

Codecs are compression schemes, very useful when you need to make calls over the internet.

Quote
Also I have allways thought about the use of coding / use of codex is something that happen at the client only. Right/wrong ?

It needs to happen on both sides, most new phones are smart enough to detect which one to use.

Quote
If you receive a steam of udp packets that contains a sound signal, could the sound quality be more or less good depending on the codex you use at the client side ? (Have been thinking rather little about the codex problem before, and that might be "wrong" ..)

Exactly the point, with the use of the right codec you use less and smaller packets, easy to traverse and keeping the sound quality.

Offline arne

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« Reply #52 on: October 04, 2005, 03:48:40 PM »
stuntshell -> Thanks for info !

psoren ->

"The X100 card allows you to recieve PSTN calls. That means you can recieve PSTN calls into the asterisk server. There are some tasks in the Asterisk@Home command line to install the card."

But the Aserisk is quite much something like a "Computerized local telephone switchborad". If you can receive calls shouldn't you also be able to call out from the local clients and via the X100 card ??

(Important question .. because if it can do that, it should also be possible to set it up like a private link (bridge) between the sip network and the PSTN network.)(Hiden question behind: Is the Aterisk "just" a "local switcboard" or is it also actually able to work like a "sip telephony server" ?) (Receive sip callas from anywhere and then tranfer them over to the PSTN network.)
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Offline psoren

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« Reply #53 on: October 04, 2005, 04:24:14 PM »
Quote from: "arne"

But the Aserisk is quite much something like a "Computerized local telephone switchborad". If you can receive calls shouldn't you also be able to call out from the local clients and via the X100 card ??

(Important question .. because if it can do that, it should also be possible to set it up like a private link (bridge) between the sip network and the PSTN network.)(Hiden question behind: Is the Aterisk "just" a "local switcboard" or is it also actually able to work like a "sip telephony server" ?) (Receive sip callas from anywhere and then tranfer them over to the PSTN network.)


Yes, Asterisk can do all that but you need the right hardware. X100 is a FXO card. If you want to call from asterisk to PSTN then you need a FXS card. But why would you want that when your VoIP provider does it for you  :-)
It's more expensive hardware, too expensive for just fun.....

Per

Offline arne

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« Reply #54 on: October 04, 2005, 05:53:17 PM »
He, he .. I dont have a PSTN line at all, only a ADSL connection and a number of sip telephone "lines". Thats the reason I have to ask as I am not able to carry out those tests myself.

I believe that the "technology class" or "type of technology" that Asterisk represent will have a lot of impact on businesses and on societies in the relatively near future.

For 10 years ago, I called a person in some far away country and it costed me 20 dollar just to know that he was not there. Today I can speak with the same person for hours and hours for allmost nothing.

The interesting part of it is to try to explore and find out the real potencial of this software.

Like the web browser started "a new generation of internet", I believe that Asterix and this class of programs also will be "a new generation of something".

One thing that I belive will/should come is more or less free worldwide videotelephones, ofcource behind sme gateways ..

By the way  .. thanks for the info.

Arne.
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Offline arne

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« Reply #55 on: October 06, 2005, 01:23:48 AM »
Some practical info: I tried this vendor. Very beautiful web page, but very poor quality in sound in this part of the world. http://www.broadvoice.com/

I found this other one on the net. My first impression is really cheap prices ond also ok quality:
http://www.iax.cc/ (Even though long distases to american servers)

By the way I think there is one practical and very usable way to use Asterisk, also for privat persons:

You can set up different external "routes" so you can be connected to different telefhone companies at the same time. Then you can program in which vendor should be used for which trafic. If you have a local provider vith high quality and cheap prices on local trafic, then the local trafic can flow via this vendor. Then you can program in that more distant trafic should be carried by a vendor that can do this job the best way (iax.cc !!) YOu can block access to mobile telephones if you think "mobile to mobile" is bether etc.

All this smart thinking about prices and things like that can be programmed into the Asterix and you can have access to more than one telephone operator from one telephone. Everything can be programmed so it automatically finds the ceapest or best way out.

I think that one of the challanges of Asterix@home is to find usable and practical way to use it. I think this ability to chose between telephone vendors is a very good one.
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Offline arne

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« Reply #56 on: October 07, 2005, 03:49:49 PM »
Possible major bug in the SME Asterisk installation (??!!)-

There is a number of smaller bugs, but possible also a major one.

The bug that apeared in my installation was that it were not able to register for incomming calls on sip servers. It worked for inn and out trafic against iax servers but only for outgoing trafic against sip servers.

After trobleshooting a bit I foung that the reason were that it did not register or renew registration on sip servers. This function worked ok on iax servers.

After making a new installation on an other PC using the "Centos 3.5/Asterisk bundle" I could see that the rgistration process now worked ok both for sip and for iax servers.

(Dont know of course for sure if the bug I had with the Asterisk/SME variant were due to some error from my side or not.)

I think that the SME/Asterisk installation is great for testing and learning, but if it is a question of a stable and troble free installation it is requied to use two PC's, one for the sme gateway and one for the Aterisk/Centos 3.5 installation.

The Sme server (7.0 Beta) does on the other side a great job as a sip/iax telephone gateway. (My experinece is that many cheap routers can have problem with sip/iax trafic.)

By the way if it is a strong criteria to use only one PC and to run the Asterisk on the SME Gateway, this wendor http://www.iax.cc support iax and the SME server installation for telephone trafic inn and out. The vendor has also an easy configuration guide that work on the sme server and they also answer e-mail and gives technical support in a way that I think is OK.

My persoanl opinion is that the Asterisk server is a great and a quite usable thing, but it should be runned on its own server for its own purpose.
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Offline jester

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« Reply #57 on: October 07, 2005, 05:07:06 PM »
I've heard/read a view times that A@H has some security issues due to AMP.
Has anyone considered just installing Asterisk and using IPManager and
IPSwitchBoard (http://ipmanager.thorben.dk/) from a WinDoze-PC ?!

BTW can someone point me out on setting up an environment on SME7beta4
for compiling Asterisk?! I keep getting stuck on missing kernelsources, headers,
symlinks on the right places... i want to try to install Asterisk with the Junhanns
drivers for cheap ISDN cards.

Regards,
Jester.

duncan

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« Reply #58 on: October 08, 2005, 01:40:20 AM »
Quote from: "jester"

BTW can someone point me out on setting up an environment on SME7beta4
for compiling Asterisk?!


You could try

Code: [Select]
yum install kernel-devel

Offline arne

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« Reply #59 on: October 09, 2005, 11:38:53 AM »
A few things to be mentioned ..

The Asterisk server installation now works just fantastic.

I first tried diverse soft telephones among other X-ten and SJPhone. X-ten was most easy to configure, but SJPhone had best sound quality. SJPhone had some strange problems loosing connection to the Aterisk server after 6-7 hours in such a way that "the only" way to recover vere to reboot the server.

I then tried a Grandstream Handytone adapter. This really improved everything.

The quality of the telephony via the Aterisk installation is now equal to any other telephone connection. The most surpricing side of the story, as I will see it is that the quality of an american "sip telephone line" is very "ok and usable" ( http://www.iax.cc/ ) (Actually it is not a sip but a IAX connection. This seens to give superior quality for the Asterisk server.)

After had read about the Aterisk server "allmost" for the first time in this tread, I am now definitively an "adicted Asterisk user".

The tarif for calling my relatives abroad used to be approx 1 dollar per minute. Now it is just 5 cent and I still use the same handhold wireless analogue telephone and the call up is still much the same as before.

I believe that the Asterisk@home for SME is quite useable "thing" with some "minor bugs" that could be corrected ??

A one more "thank you" to Cyr and everybody else that has supplied with info in this tread.
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Offline Franco

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« Reply #60 on: October 10, 2005, 07:25:24 AM »
Quote from: "jester"
I've heard/read a view times that A@H has some security issues due to AMP.
Has anyone considered just installing Asterisk and using IPManager and
IPSwitchBoard (http://ipmanager.thorben.dk/) from a WinDoze-PC ?!

Regards,
Jester.


This is great @jester!!!
It's not yet perfect and lacks database support, but for basic configuration It makes it very easy!

Thanks,

Offline arne

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« Reply #61 on: October 16, 2005, 02:05:31 AM »
Just as an experiment, I tried to rename the two main configuration files (that AMP is making) and then to try to edit some easy configuration by hand.

My idea were just, if there were som "more easy and light" Asterisk installation available, rpm based or whatever, without all that owerhead of configuration things that is in Asterisk@home, possibly such a installation could be fitted into the sme server without to much "noise", and then the configuration could be some more "easy framework" edited by hand.

I just tried some testing on the Asterisk@home with these two files, setting up one external telephone line and 3 internal extensions. The 3 internal extensions can call each other. When a call is received from the outside all the telephones are ringing.

Have not been testing to much, but I think it works at least "by part".


sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = sip_guest
srvlookup = yes
language = no

register=>20202020:123456@213.160.242.135/20202020

[400]
accountcode=400
callerid = "navn" <400>
type = friend
host = dynamic  
username = 400
secret = 1234  
context = fra_40X
dtfmode = rfc2833

[401]
accountcode=401
callerid = "navn" <401>
type = friend
host = dynamic  
username = 401
secret = 1234  
context = fra_40X
dtfmode = rfc2833

[402]
accountcode=402
callerid = "navn" <402>
type = friend
host = dynamic  
username = 402
secret = 1234  
context = fra_40X
dtfmode = rfc2833

 
[ip24_utgaaende]
type=friend
secret=password
username=20202020
host=213.160.242.135
fromuser=20202020
insecure=very
fromdomain=213.160.242.135  
dtmfmode=rfc2833  

[ip24_inngaaende]  
type=peer
host=213.160.242.135
insecure=very
canreinvite=no
context=fra_ip24
dtmfmode=inband  
disallow=all
allow=alaw



extensions.conf

[fra_ip24]
exten => 52907527,1,Dial(SIP/400&SIP/401&SIP/402,,)  

[fra_40X]
exten => 400,1,Dial(SIP/400,1,)
exten => 401,1,Dial(SIP/401,1,)
exten => 402,1,Dial(SIP/402,1,)
exten => _XXXX./400,1,Dial(SIP/${EXTEN}@ip24_utgaaende,,)  


By the way, made with some help from some other norwegian discussion forum.


Doea anyone know some "small and easy Asterisk variant" with "manual configuration" that eventually can be fitted into the SME server without to much trouble. Does it for instance exist a Centos 4.1 variant ??

Such an "easy installation" might not be able to do all the things that Asterisk@home are able to, but it might be able to do some of the basic things about ip-telephony ? With some "premade" configuration files that should be developed further on, it should be able to do some basicc things.


Best reg Arne.
......

Offline arne

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« Reply #62 on: October 16, 2005, 02:13:36 AM »
"Does it for instance exist a Centos 4.1 variant ??"  .. Well I did not think about the Asterisk@home this time ..
......

Offline arne

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« Reply #63 on: October 16, 2005, 02:22:22 AM »
Could this be such a "easy one" maintaning some basic functionality afer some manual configuration ???

http://dag.wieers.com/packages/asterisk/

Anybody who knows ??
......

Offline Franco

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« Reply #64 on: October 16, 2005, 05:26:36 AM »
oh no,
if you want that way look for either Duncan's packages & how-to (CVS and RPM), or Dungog's rpm. If you're talking 7B, then yum will do.

Offline arne

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« Reply #65 on: October 16, 2005, 12:00:49 PM »
OK. Yum will do .. I will try that.

The problem with the "Asterisk@home into sme server" modification is that it don't give you the functionality of Asterisk@home and the SME server. It still seem to leave the system in a rather unstable and buggy condition. (That might not be a problem for an "experimental homeserver")

The interesting thing, as I would see it, vere if it were possible to collect some of the functionality of the Asterisk server, while still keeping ordinary functions and ordinary stability for the SME sever itself.
......

Offline arne

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« Reply #66 on: October 16, 2005, 12:11:12 PM »
"Duncan's packages & how-to (CVS and RPM), or Dungog's rpm."

Do you have a link ? Would be interesting to take a look as these as well.
......

Offline Franco

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« Reply #67 on: October 16, 2005, 09:00:56 PM »
http://mirror.contribs.org/smeserver/contribs/

Duncan=dthomas

These will be for 6 and 6.5. Dungog sells the interface that will modify Asterisk like AMP does.

Offline arne

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« Reply #68 on: October 17, 2005, 12:19:47 AM »
Thanks. I tried the dungog variant. It did install, but I wer not able to get it started. (Tried different config but it just crached every time.)

I also tried the dthomas variant, but it did not install because of C3 processor on the PC.

The dthomas variant looks promicing so I will try it on another P3 tomorrow.
......

cyr

Asterisk on SME7
« Reply #69 on: October 17, 2005, 11:27:21 AM »
Hi,

Sorry for my long silence, but I tried to make a contrib for SME more attractive.

So you'll can get the new contrib A4SME7.tar.gz at http://firewall-services.com

The news,

** the installation is based on rpm from the ate rmp repository :-)

** You now have an uninstall script

** This contrib is derived from AAH beta4 with a newer version of AMP

** The user www is now in the group asterisk so there is no need to use an apache user with a shell.

Offline arne

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« Reply #70 on: October 18, 2005, 01:08:13 AM »
I have now made a test installation of actually all of those variants that is mentioned above, inkluding the last two rpm and sourcecode based variants.

For the testing I have been using my rateher old "Linux test boxes" 1. P III/700/512 laptob 2. AMD K6-II/450/512 3. C3/1000/512 4. Duron 1200/512 (Also have some newer hardware, if required.)

I were not able to make one single reliable and stable installation for any of the available asterisk variants for the SME server.

On the other hand all test installation of the Asterisk@home/Centos 3.5 bundle has been working perfectly on the P3 Laptop and the Duron 1200. It also did apear, as far as I can see it, that the it is rather easy to disable the graphical interphase of Asterisk@home and turn it over to be a manual configurated Asterisk installation.

The last two new "Asterisk for the sme server" variants were tested on the P III laptop with SME 7.0 B4 for some minuttes ago. (Have just downloaded B 5 now.)

It apeared that the rpm based version of the new asterisk installation crashed the server-manager panel. Uninstall worked but did not bring back the server-manager panel. The source code based variant did not crash anything, but the asterisk server were not able to start up.

When it comes to the problems with the two newest "Asterisk for the SME" server variants, I would guess that this might be related to the hardware that I used, a P III Laptop, that might not be the most typical server hardware. (Even thouh this runns Asterisk@home without a problem.)

If any of you get the new "asterisk for SME server" to work well enough could you please send a post and inform about hardware. (If it's a good chance for success I will find another and betther PC). A fully working SME server with a fully working Asterisk server would be really a nice "thing"  :hammer:

Until now my conclusion will have to be that if you use two PC's you can have a stable and nice working SME "multiserver" and stable and really flexible Asterisk@home telephony server. If you try to run it on the same hardwere it will not work. (Sorry, I don't like this conclusion at all. Will proceed the testing imidiately if I can see a chance for success.)

Best reg Arne.
......

cyr

Asterisk on SME7
« Reply #71 on: October 18, 2005, 10:26:05 AM »
for the crash goto http://forums.contribs.org/index.php?topic=29275.0

I'll try to make a fix today

cyr

Asterisk on SME7
« Reply #72 on: October 18, 2005, 05:54:09 PM »
bug fixed

enjoy

cyr

Asterisk on SME7
« Reply #73 on: October 18, 2005, 06:08:39 PM »
Jester said :
Quote
A@H has some security issues due to AMP


I think that is because in A@H they used a httpd user with a shell.
But it is not necessary.
In my first contrib, because of that, I used the admin user which is the httpd-admin user and I encounter a lot of trouble with amp link in the server manager.
However, in my new contrib A4SME7, I just create a user and group asterisk wich is the user of the asterisk deamon and I add the user www in the group asterisk.
I know this is just a start but I think AMP is a really good interface to manage asterisk ;-)