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smeserver-asterisk and SME7rc2

Offline arne

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smeserver-asterisk and SME7rc2
« Reply #15 on: May 04, 2006, 01:20:03 AM »
By the way, I can see that you are accessing the DISA function from internal while I does it from an external connection (via my mobile phone).

Just now I still have the old 6.0.1 installation.

When I call up the DISA function from internal it does not have a problem. When I call it from external sip it doesent work. I guess this indicate a nat/router problem.

I will now reinstall the SME 7.0 R2 and your RPM's again.

The hardware I am using for testing is a Pentium III 800 Mhz with 500 Mb RAM.

I understand it will be correct to install the identical same rpms as you even though I use an old single processor PC ?

What is selintra-sail ? (I did not install something like that, should I ?)

One other thing, when you perfomed this last test did you use a SME 7.0 R1 or a R2 ?

(Reason for this last question: I made two sequently different installations of SME 7.0 R2 and your asterisk rpms. For each of those 2 installations a "rpm -q" gave a message that the rpms were not installed, even though they were.) (So I wonder: Would it be bether to try with R1 ?)

Best reg Arne.
......

Offline arne

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« Reply #16 on: May 04, 2006, 01:42:57 AM »
I the "Sail" something like this, a configuration panel ?
http://www.dungog.net/sme/panels/Asterisk.html

Can the "Sail" rpm be downloaded somewhere ?

Exuse me for not being very updated on the sme server (the Asterisk server itself took all my spare time for a while :-)

Arne
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Offline arne

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« Reply #17 on: May 05, 2006, 12:55:01 AM »
Once more exuce me for knowing absolutely nothing about your project before jumping into it at this forum.

I have now installed what I downloaded to be a SME 7.0 R1

Configuration panels says SME 7.0 R2

uname -r says 2.6.9-34.EL

Actually, I don't know for sure if i run R1, R2 or R1.5

On this installation i installed the folloving rpms:

selintra-sail-2.1.11-187.noarch.rpm
smeserver-asterisk-1.2.3-2.i686.rpm
smeserver-asterisk-sounds-1.2.2-2.noarch.rpm
smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm

Hardware is an old P III / 800 single processor

I also have gone trough my router setup to eleminate problems here.

This time all installations went smooth and I could test your selintra-sail and the Asterisk rpms as well.

I think that there might be no doubt that you have done a great job in this project to integrate Asterisk into the sme server and then to keep the alternative to do a manual configuration as an option (By not using the selintra-module.) (I hope I have understod it right.)

What I ment somwhere above when I said that integrating the Asterisk controls into the server-manager vould mess up the SME server, I was actually thinking about the integration of Asterisk@home into the SME server that has also been done. This will allvays be "a bit messy" I think.

It looks for me like that the selintra-sail installation is made much more acording to the design priceples that the SME server is built up around.

On the other side - it must be really difficult to integrate al those technologies that is into the sip/iax telephony into one graphical control panel. I guess that this will be more easy later on as technology (and configurations) will tend to standardize (??)

But the manual configuration is still there as a option, and that is great.

When it comes to my small problems or intability with the DISA function, this seems to be related to someting more "Asterisk server deeper thing".

I get these errors in the log:

May  4 23:58:36 NOTICE[3531] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.24.214.24

The DISA function works ok when called from a local phone. It's only from my mobile phone I get some instability. The problem repieted when I tested with SME 6.0.1 and the old Asterisk RPM from 2004. On the oter hand - for some strange reason Astlinux with the identical same configuration files does not have a problem at all. I will try to find out about this a a more specific Asterisk server problem, as I think it is.

I guess that your Asterisk integration for the SME server will run in a stable way that will not affect the server in a negative way. (Will now try to test it out. :-)

Thanks for a great job.

By the way, is it possible to download your Sark iso for testing ?

(My mail: heiarne_at_gmail_dot_com)


Best reg Arne.
......

Offline SARK devs

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« Reply #18 on: May 05, 2006, 07:50:01 PM »
Quote
May 4 23:58:36 NOTICE[3531] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 64.24.214.24


Hi Arne,

You almost certainly have a sip device set so that it doesn't send silence.  In other words, it uses something called Voice Activity Detection, or VAD, to detect silence and then doesn't send any packets in order to ease bandwidth usage.  Unfortunately, it sounds weird. So...  in order to compensate, the receiving device is supposed to generate "comfort noise" (basically an ambient background burble) so that the whole thing sounds better to the human ear.  Long and short is Asterisk doesn't do it, hence the message.  So you should disable silence suppresion on the sending SIP device.

Kind Regards

Selintra

Offline arne

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« Reply #19 on: May 06, 2006, 01:55:07 AM »
I know about this problem. The xten light forinstance comes with a default setting that do not fit with Asterisk so it will have to be reconfigured.

The DISA function workes 100 % of the times I try it from my own local network.

The strange thing is that it some times work, some times does not work and some times "work partly" when the DISA function is used by call up from my mobile telephone (from en external extention). From the ISDN telephone at work it does not at all. From my friends IP telephone it works 100 %. It's just unriable, now and then leaving the mentioned (incorrect) error message in the log.

When I use the identical same PC, the same internet line, the same router, the same configuration files, the same of everything, exept for that I'm changing asterisk@sme with Asterisk at Astlinux, then absolutely all problems are solved and there is not a single error at all.

I don't understand it at all, how it can be that the Astlinux does everything 100 % OK while asterisk@sme does not do it reliable with all conditions set to be identical the same.

When I tryed the old aterisk@sme from 2004 (and SME 6.0.1) the problem were exactely the same. The asterisk@sme did not do a reliable DISA function, while Astlinux does it all without a problem all the time.

The strange thing that I does not understand is that the error og asterisk@sme anno 2006 is identical the same as the error of asterisk@sme anno 2004 (When it receivew DISA calls from the outside world it start to talk about "Comfort noice" and things does not work as they should. Then put int he Astlinux flash ram installation with identical configuration, and everything works without a problem.

The Astlinux installation is a minimized Asterisk on Linux installation only filling approx 28 MB. It all runs from RAM and it consists of 17 running processes, thats all. asterik@sme is something like 10 times as big. One thing I ask myself about is if the problem could have something to do with performace ? I dont know, but the Astlinux is so small and so little demanding that it can run on allmost anything.

(By the way, the DISA function also works 100 % (here at my place) with the asterisk@home installation, and that is not a very litle intallation. It is only asterisk@sme that gives me this rather strange problem.)

It would be interesing to know if there is some other that has experienced some problems like this.

Best reg Arne
......

Offline arne

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« Reply #20 on: May 07, 2006, 01:30:20 PM »
I have now tested with a asterisk Digium sourcode installation on a Clarkconnect 3.2 Home.

It apeared that the error or instability with the DISA function vere exactely the same on this installation, so it obviosly has nothing to do directely with your rpms.

Its rather strange. When I use Astlinux everything works ok. When I use anything else there is some instability.

Problem is not there when the DISA function is called up from a local extention. When called up from an external telephone it is there.

Could it be that there is some default parameters releted to the compilation of the Asterisk server that chould be changed ?

(Or could it be just some other configuration item ..)

(Don't know ..)

Best reg Arne.
......

Offline JonB

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« Reply #21 on: May 07, 2006, 02:48:04 PM »
Arne,

I have no problems with using DISA either from an incoming Gateway number or from a hidden option in an IVR menu using the smeserver-asterisk rpms.

I don't actually use DISA on an incoming Gateway line as it is a waste of a line but I have tested it sucessfully.

I do use it as a hidden option in an IVR message.

This what I have used in the past on A@H installations and have used as a custom IVR application for smeserver-asterisk installs


[custom-disa]
exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,Authenticate(****) ; replace **** with a numerical password
exten => s,5,DISA(no-password|from-internal)

Jon
...

Offline arne

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« Reply #22 on: May 07, 2006, 07:37:13 PM »
Thankyou for your information, JohnB !

I have finaly traced down the problem in the end (at least I hope it is the end ..)

It apeared that my nat router was set up to support UPnP.

This gave some strange behaviour for the communication related to the Asterisk server. Some times and mostly it worked OK, but some times also it made some rather strange behavoir and inncorrect error messages.

It took a while, but now I know :-)

I have tested the smeserver-asterisk rpms without the configuration rpm, so I know that can work.

It was just this strange instability that actually aperared to be the UPnP function of the NAT router (Thomson Speedtouch 516) that played a few trics. (And to confuse me even more, my Asterisk intallation on Astlinux did not have a problem at all with the UPnP function.)

One reason that I use the DISA is because I like to have the Asterisk dial tone available at my mobilephone. (this usually saves me 90 pst or more for the international calls.) One other reason is that I have some friends with "old telephones" that are allowed to make free international calls trough my server. (And I don't have an incomming POT line at all, anymore.)

I used to use the telecard system at asterisk at home, a2billing for this function, but I found out that Astlinux with the use of the DISA function could do a bether and more easy job.

The next step were then to try to integrate it all into my SME web/mail server.

Bu the way, I think all the SIP/IAX telephony vendors I use are accepting parallell calls on the same account, so that the only practrical limitation is the bandwith of the network connection. Until now this has never been a problem at all. So when people are calling up and using my Asterisk server via the DISA function, the only way I can see it or notice it is from the log. (Refer to the Astlinux server I have been using until now.)


Best reg Arne.
......

Offline SARK devs

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« Reply #23 on: May 07, 2006, 08:58:45 PM »
Hello Arne

Glad you sorted your problem out.  Might we ask a favour?  We've just acquired a small 400Mhz diskless thin client box that we'd like to install AstLinux on.  Any notes or pointers you could give us would be much appreciated.

You can contact us at admin@selintra.com

Kind Regards

Selintra

Offline arne

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« Reply #24 on: May 08, 2006, 01:10:22 AM »
Tonight I think I tried to sort out thousends and millions of rpm's the old fashioned way to set up an environment on the SME 7.0 so it vere able to compile the Asterisk sourcode. But it it did not work. Somewhere in the middle of the asterisk sourcecode compilation it crached. (But it worked on the Clarkconnect 3.2 Home.)

On the other hand your rpms went in nice and smoothly today, and everything worked ok after I found out that my router played a lot of strange triks with me with the UPnP function.

When it comes to thin clients and Asterisk Astlinux is absolutely the right thing. Their web page actually contains a lot of the info you need.

http://www.astlinux.org/

They also has a mail list (mentioned on this page) that I can recomend. I have just been using their Astlinux version 0.3.0 until now. They has just released version 0.4.0 that is sligthly bigger (increase from 32 to 48 MB foir the complete installation with Linux OS.)

The way I did it when I first installed it on a PC with a standard flash ram was just to download the Asterisk ISO This contained the required tools to instyall the OS with Asterisk to the flash memory from an ordinary Windows PC. (I just used the standard memory and the memory reader that I usually use for my digital camera.)

Later on I also bought a used thin HP client to also check out this. This was more difficult, because the 32 MB flash memory was embeded into the mainboard, and there was no CD provision.

The way I did it was to use two standard USB pen memories an another Linux distro called Slax http://slax.linux-live.org/?lang=en

There is also a program called myslaxcreator:
http://myslax.bonsonno.org/download.php

Myslaxcreator is able to make a booable USB pendrive from a SLAX CD.

I think I used Slax version 5.0.3 as I believe that Myslaxcreator does not support the newer verions of Slax.

So I booted the thin client from the one USB drive with slax while I had a image of the Astlinux on the other USB stick.

Then when Slax (Linux) had started I used the standard Linux dd command to write over the Astlinux image from usbb to the embeded flash memory on the thin client.

If you join the Astlinux mail list, you can just send a mail to this list and I will receive it.

You have to register on the list before it really will work.


Best reg Arne.
......

Offline jester

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chan_misdn
« Reply #25 on: May 08, 2006, 07:19:35 PM »
Selintra,

I'm new to the whole Asterisk scene so bare with me if i'm being stupid. I've been reading (http://www.voip-info.org/wiki/view/chan_misdn) that misdn/chan_misdn is part of asterisk 1.2 so does this mean it is also supported by your Asterisk RPM's and SAIL? I've read that HFC based cards are supported by misdn so i wondered if i could use Asterisk/SAIL for my cheap hfc card to experiment with asterisk as PBX.

Regards,
jester.

Offline SARK devs

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« Reply #26 on: May 08, 2006, 08:41:31 PM »
Quote
i wondered if i could use Asterisk/SAIL for my cheap hfc card to experiment with asterisk as PBX.


Hello Jester

You pose a good question.  We would like to do something for passive BRI cards but unfortunately, asterisk is a bit of a mess in this area with several competing (it seems to us) solutions, none of which is an obvious choice.  We intend to do some work in this area over the next few weeks so watch this space....


Kind Regards

Selintra

Offline arne

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« Reply #27 on: May 09, 2006, 10:56:11 PM »
I have just installed and setup a new SME 7.0 R2 server with your astrisk rpms. Looksl like it is working quite OK.

Just a small thing - there is no automated startup after reboot, is it ?

It has to be started with "asterisk start" ?

I dont use the Selintra configuration tool. (Could this be the reason ?)

Best reg Arne.
......

Offline SARK devs

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« Reply #28 on: May 09, 2006, 11:12:37 PM »
Hello Arne,

Hope you are well.  

You are quite correct.  The asterisk rpms are deliberately very simple so that users may do what they wish with them.  There are automatic start-up and shut-down routines in the SAIL workbench/generator but not in the asterisk rpms.   It is not difficult to create a start-up routine of your own usng the tools SME Server provides for server management.  There is also an example of an asterisk start-up routine in the asterisk sources on the digium site.  It is called safe_asterisk.

Kind Regards

Selintra

Offline SARK devs

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« Reply #29 on: May 09, 2006, 11:45:33 PM »
Hello again Arne!

Please accept my apologies, the above answer is not quite correct.  I just downloaded the source rpm to check.  The start-up and shut-down routines are included in the rpm. Look at /etc/rc.d/init.d/asterisk.  We simply don't create the data structures in SME to have SME-server automatically invoke them.    To automate the system you simply need to do the following:-

Code: [Select]

mkdir -p /etc/e-smith/db/configuration/defaults/asterisk
echo service >  /etc/e-smith/db/configuration/defaults/asterisk/type
echo active >  /etc/e-smith/db/configuration/defaults/asterisk/status


Run console save to force sme server to rebuild the database ( there is a less expensive way, but I can't remember it at the moment).
Code: [Select]

/sbin/e-smith/signal-event console-save


That's it.  Asterisk should now start and stop automatically.

Kind Regards

jeff@selintra.com