Topic: Asterisk on SME7

[kangkc]

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]

[arne]

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]

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

[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.

[psoren]

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

[arne]

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   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.

[arne]

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.

[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

[psoren]

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

[arne]

".. 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.

[arne]

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.)

[Franco]

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.

[arne]

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.

[arne]

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.

[arne]

Look at this. Not to bad for us testers (if it works).

http://sipphone.com/5free1/