Topic: smeserver-asterisk and SME7rc2

[SARK devs]

Hi All,

As you probably know, rc2 went up onto the mirrors late on the 26th.  Among other things, it's uses the CentOS 4.3 base which means that the kernel has changed.  In 4.3 they've used the 34-ELsmp Kernel only (no UP).  

Whether you are running the older monolithic asterisk-SME7rc1 rpm or the smeserver-asterisk rpms, you should remove them BEFORE you upgrade to rc2 (you don't need to worry about SAIL - it should be fine).  To be specific do;

either

Code: [Select]


rpm -e asterisk-SME7rc1


or

Code: [Select]


rpm -e smeserver-asterisk-zappri
rpm -e smeserver-asterisk
rpm -e smeserver-asterisk-sounds


Then proceed with your upgrade...

After the upgrade completes check which kernel you are running (with uname -r).  We've upgraded three machines today and two of them happily came up with the new 34-ELsmp kernel but one came up with vanilla 34.  This is important to asterisk because the two kernels require different versions of the zappri drivers.

If you have 34-ELsmp (which is correct) then install

smeserver-asterisk-zappri-MPP-1.2.2-1

If you still have 34 and can't or don't want to fix it then install

smeserver-asterisk-zappri-1.2.2-1

Then install

smeserver-asterisk-1.2.3-2

followed finally by

smeserver-asterisk-sounds-1.2.2-2

You can get the rpms from here if you don't already have them.
 htttp://mirror.contribs.org/smeserver/contribs/selintra/RPMS/

After you're done,  reboot your system and asterisk should come up as normal.

Kind Regards

Selintra

[jazbokes]

Corrected  link to http://mirror.contribs.org/smeserver/contribs/selintra/RPMS/

I guess a typo..

[arne]

I tried smeserver-asterisk-1.2.3-2.i686.rpm, smeserver-asterisk-sounds-1.2.2-2.noarch.rpm and smeserver-asterisk-zappri-1.2.2-1.i686.rpm just now. (Had seme already tested config files that I used.)

It worked exept for that I had some problems with the DISA fuction (That can deliver dial tone to incomming connections.)

What is this module for:  smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm ?

Is it an alternative to smeserver-asterisk-zappri-1.2.2-1.i686.rpm  ?

What is the difference ?

(I use sip/iax telephony only, no zap things but it looks like that one zap module is requiered to make it all turnaround anyhow..)

[arne]

Yes, I'm using the newest SME 7.0 RC 2, I forgor to mention that.

Arne.

[arne]

... would it be more easy to make the sme asterisk server to work (fully) with the SME 7.0 R1 distro ?

[arne]

Excuse me for not reading your post well enough .. I can see it now about the smeserver-asterisk-zappri-MPP-1.2.2-1 (and I think I took the wrong one) Will test it tomorrow.

[SARK devs]

Hello Arne

Sorry to hear you've had a few problems.  You are quite correct; you must us the MPP version of zappri with RC2.  

You also deduce correctly that you need zappri even if you aren't planning to run any analogue lines or telephones.  Asterisk gets it's timing from zappri and without it,  conference rooms and music-on-hold wouldn't work.

Tell us more about your DISA problem and we'll try to help.

Kind Regards

Selintra

[CharlieBrady]

As you probably know, rc2 went up onto the mirrors late on the 26th.  Among other things, it's uses the CentOS 4.3 base which means that the kernel has changed.  In 4.3 they've used the 34-ELsmp Kernel only (no UP).

That is not true. rc2 includes both uniprocessor and SMP kernels.

[SARK devs]

Apologies, you are quite correct.   All of our upgrades (bar one, which we weren't sure was conducted correctly), came back up with the SMP Kernel.  We've checked the CentOS site and 4.3 contains both UP and SMP Kernels as always.

To recap for asterisk, if your system comes up "smp" then you should use zappri-MPP if it comes up "UP" then you should use zappri.

kernel-2.6.9-34.EL.i686.rpm use smeserver-asterisk-zappri-1.2.2-1.i686.rpm

kernel-smp-2.6.9-34.EL.i686.rpm use smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm

Best

Selintra

[CharlieBrady]

kernel-2.6.9-34.EL.i686.rpm use smeserver-asterisk-zappri-1.2.2-1.i686.rpm

kernel-smp-2.6.9-34.EL.i686.rpm use smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm

Shouldn't you install both on all systems? Or, if that is not possible, would it not be better to build one RPM which includes all to allow work with either kernel?

[SARK devs]

Shouldn't you install both on all systems? Or, if that is not possible, would it not be better to build one RPM which includes all to allow work with either kernel?

Good input. We'll do some research and get back with you.

Selintra

[arne]

It looks like there is some bugs with the DISA function. I tried a fresh new SME 7.0 R2 installation with these packages:

smeserver-asterisk-1.2.3-2.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm
smeserver-asterisk-sounds-1.2.2-2.noarch.rpm

Configuration/use of the DISA function is just stright forward:

(extensions.conf)

exten => 1,1,Authenticate(123123123)
exten => 1,2,DISA(no-password|trunkname)

This should normally be working, but it does not. It should give back a stable dial tone to the dialer (??) but it just give some unstable "wawa's" and then the caller is auto logged off after a few seconds. Messages from the Asterisk CLI:

sme7*CLI>
sme7*CLI>
    -- Playing 'auth-thankyou' (language 'no')
    -- Executing DISA("SIP/12345678-bd0d", "no-password|trunkname") in new stack
  == Spawn extension (incomming, 1, 2) exited non-zero on 'SIP/12345678-bd0d'
sme7*CLI>

Unless I do something wrong (I don't know about), I believe that this is an error that will repeat on all installations.

By the way - It works 100 % OK for the "ordinary telephone functions" and there is no problems (that I have fount until now) dealing with ordinary incomming and outgoing telephony functions.

(I wonder if I red that the newest A@H has some problems with the DISA function as well. Could it have something to do with the newest Asterisk server itself ?)

And one other thing - I think that it is a right approach to do a Astereisk impementation to the SME server to chose a rather "minimalistic approach" like these Aterisk rpms. To bundle something like A@H will destroy your SME installation completely. I think that if "the Asterisk part" of the SME server should have a graphical interphace at all, it should be something like a easy test editor for the configuration files, like the Astlinux  http://www.astlinux.org/

(Reason: When configuring Asterisk there is so many factors outside the server itself that you will have to take into consideration. If you make an easy standard configuration via a graphical shell a la server-manager this might fit with one vendor and one kind of equipment, but not all of them. Because of this there is no way around to let the user manually edit the configuration files itself as long as the sip/iax technology is just so variated and unstandardized like it actually still is today. Editing the asterisk configuration files could be done via ssh shell or via a simular web shell a la Astlinux. With such a "minimalistic aproach" it should be possible to run the Asterisk process(es) in the sme box without messing it all up. )

Best reg Arne.

[SARK devs]

Shouldn't you install both on all systems? Or, if that is not possible, would it not be better to build one RPM which includes all to allow work with either kernel?

We've had a look at this.  Turns out that the guys who did the MPP rpm (Lloyd Keen and his buddy down in Aus) are a lot smarter than we are at Selintra.   They've already included both kernel paths in the work.  So...  The upshot is that you can run the MPP rpm with either kernel.  Well done Lloyd and well spotted Charlie. Oh... and not so well done Selintra - must try harder.

Best

Selintra

[SARK devs]

It looks like there is some bugs with the DISA function.

Hi Arne

We can't simulate your problem.  DISA excutes correctly under SAIL/smeserver-asterisk. Here's our test server set-up.

Code: [Select]

[root@old1 ~]# rpm -q smeserver-asterisk
smeserver-asterisk-1.2.3-2
[root@old1 ~]# rpm -q smeserver-asterisk-zappri-MPP
smeserver-asterisk-zappri-MPP-1.2.2-1
[root@old1 ~]# rpm -q smeserver-asterisk-sounds
smeserver-asterisk-sounds-1.2.2-2
[root@old1 ~]# rpm -q selintra-sail
selintra-sail-2.1.11-181


Here is a SAIL DISA call going through...

Code: [Select]

-- AGI Script Executing Application: (Authenticate) Options: (XXXX)
    -- Playing 'agent-pass' (language 'en')
    -- Playing 'auth-thankyou' (language 'en')
    -- AGI Script Executing Application: (DISA) Options: (no-password|internal)
    -- AGI Script Executing Application: (Hangup) Options: ((null))
    -- AGI Script selintra completed, returning 0
    -- Executing AGI("IAX2/217.14.132.185:4569-4", "selintra|InCall") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (DBget) Options: (dbVal=cfim/7000)
    -- DBget: varname=dbVal, family=cfim, key=7000
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=cfim/7000)
    -- DBget: varname=dbVal, family=cfim, key=7000
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (DBget) Options: (dbVal=ringdelay/7000)
    -- DBget: varname=dbVal, family=ringdelay, key=7000
    -- DBget: Value not found in database.
    -- AGI Script Executing Application: (Dial) Options: (SIP/7000|10|t)
    -- Called 7000
    -- SIP/7000-4757 is ringing


What you are seeing in the log is SAIL accepting the DISA password and then entering the DISA application and accepting a number to dial (in this case an internal extension of 7000)  then pushing it onto the stack to be executed.  You then see the CFWD processing code trace followed finally by the phone ringing.  

If this is an asterisk problem then it's not one that seems to affect SAIL.  Sorry we can't help further but this looks like a syntax or set-up problem.


Kind Regards

Selintra

[arne]

Thankyou for your help !

This problem with the DISA setup is a rather strange one ..

Just for fun and to check it out I installed SME 6.0.1 and the old Asterisk rpm from 2004:

http://mirror.contribs.org/smeserver/contribs//dthomas/smeserver/6.x/Beta/Asterisk/

It apeared that the DISA problem was exactely the same using this software.

When I make a copy paste of the configuration files over to a Astlinux installation (running from a flash memory on another PC), there is no problem with the DISA function or anything else.

I wonder if the problem with the sme 6.0.1/7.0 and Asterisk can be related to the setup of my nat router so I will try to troubleshoot that .. (But still .. the Astlinux doesn't have a problem, running trough the same router, so it's rather strange ..)

Best reg Arne.

[arne]

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.

[arne]

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

[arne]

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.

[SARK devs]

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

[arne]

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

[arne]

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.

[JonB]

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

[arne]

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.

[SARK devs]

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

[arne]

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.

[jester]

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.

[SARK devs]

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

[arne]

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.

[SARK devs]

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

[SARK devs]

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