Topic: Asterisk on SME7

[arne]

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.    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.

[arne]

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.

[arne]

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.    

(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.)

[Franco]

(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%   ):

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

[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 !?).

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.

[psoren]

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

[Franco]

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.

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.

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.

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.

[arne]

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.)

[psoren]

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

[arne]

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.

[arne]

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.

[arne]

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.

[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 ?!

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]

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

[arne]

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.