Topic: SAIL -414

[Franco]

A couple of things:

I'm not sure why the symlink would go missing but I'll have it checked on a test server.

I can confirm that as well, Hervé's suggestion fixes it.
----------
Is the fxotune feature  only available to the TDM cards? I have a X100P clone that I would love to eliminate the echo
----------
I had once again to do:

Code: [Select]

[root@sme ~]# chmod 755 /home/QoS/qos
[root@sme ~]# /etc/init.d/qosd start
Starting QoS:                                              [  OK  ]

----------

N.B. You will be prevented from modifying any of the SAIL-"managed" files (the "SAVE" button will not be displayed). They can only be changed by changing various settings within SAIL itself.

There are a lot of things I had modified on the General Edit Menu that did not come with the upgrade, and now I cannot change them anymore? Volume on the zaptel cards, voicemail instructions and the fromto are just a few.
---------
Is it possible NOT to use a Pre Select two digit code when dialing through an analog line?

[sonoracomm]

I began looking into QoS this last weekend and I was having trouble seeing it work.  Now I know why.    

I too had to run the commands listed above to make it work.  I used

Code: [Select]

/etc/init.d/qosd status
before and after running the commands to see what changed.

Now I need to do some testing...

Comments from a newbie concerning QoS:

The QoS Server-Manager panel seems like voodoo.  There might be more explanatory text to help the uninitiated.

A testing regimen (example) would be nice too.

Again, thanks so much for all your work and help,

G

[gippsweb]

I have been using the SAIL Qos for a couple of months and it seemed to be doing the job quite well until late Dec when we changed from 1500/256 to 8000/384.
I left the settings as they were but call quality has dropped of dramatically.
I've been playing around with the figures, but as mentioned by Sonoracomm, the qos panel is a tad vague. we can't seem to find a happy medium so I have it disabled at the moment and only run any large downloads overnight when noone is here.

[sonoracomm]

It sounds like you went from DSL to cable?

At least around here, DSL offers much better all-around performance than cable.  DSL is often a MUCH more direct connection to the ISP than cable and may offer much better routing and more controlled latency...if not the fastest downloads.  As long as you aren't more than about 15K feet from the CO...

With cable, you are most certainly sharing the connection to the head-end with a few thousand of your closest neighbors.

I have customers with remote IP phones that report problems over cable that I haven't seen with DSL connections (sorry, not SME/SAIL YET! But I'm working on it...)

G

[Tib]

A lot of providers here in Ausi land are going to 8000/384 on the ADSL instead of ADLS2.

I'm switching to 8000/384 in 2 weeks as well ... I'll see what happens.

Although I don't use the QOS that comes with sail ... I use wondershaper.


Regards,

Tib

[hervep]

Is it possible NOT to use a Pre Select two digit code when dialing through an analog line?

Maybe I do not understand your question as I should, but why don't you use a route instead of ( or in addition with ) your trunk pre-select ... ?
In my case ( Belgium ) the common access code for 'outside' is '0'. As a result, just made a route to my analog provider trunk (X100P) using '_0.'.
Just means 'send everything that begins with 0 to that provider trunk, regardless of what follows'. In this particular case, it works a little bit as a 1 digit 'pre-select'.

Kind regards,

Hervé

[hervep]

I began looking into QoS this last weekend and I was having trouble seeing it work.  Now I know why.    

I too had to run the commands listed above to make it work.  I used

Code: [Select]

/etc/init.d/qosd status
before and after running the commands to see what changed.

Now I need to do some testing...

Comments from a newbie concerning QoS:

The QoS Server-Manager panel seems like voodoo.  There might be more explanatory text to help the uninitiated.

A testing regimen (example) would be nice too.

Again, thanks so much for all your work and help,

G

Don't forget that 'QOS' concept is much more complicated that what we can do onto SME/SAIL. Real QOS concept should be end to end, it means that your provider should be aware of the QOS level you want. This is currently (most of the time) not possible, certainly in combination with 'low-cost' internet access. The only thing you do by setting up 'QOS' onto SME is to avoid somebody on YOUR network to take 'full' bandwith to the internet. Conceptually, it is more efficient on 'DSL' connections, just because you can more or less control your traffic to the first 'router-concentrator' of your provider. after that 'router', the quality you will have is very unsure. In case of cable connection, this is always unsure, just because your neighbour is using the same pipe to the first 'router' without being aware of your QOS concerns.
Based on those facts, the only thing you can do is to tell your 'SME' you would like to be sure that some part of your global internet bandwidth is reserved for 'Voip'. The panel offers you the possibility to tell which part in both directions.

There are other ways to try to improve the voip 'audio' quality (I.E. MTU size, codec choice & payload ... ), but ideal situation remains to fully control the bantwidth, jitter, latency end to end = 'real QOS'.

Hope this can help ...

Kind regards,

Hervé

[SARK devs]

Wow - you guys have been busy!

Let's see....

Stuntshell - General Edit panel.  This panel was thrown together quickly to allow edit of those files which SARK/SAIL doesn't touch.  However, the "headers" panel still exists for those files that we "share" with you.

Sonoracomm - fxotune -m parameter. I have updated the fxotune panel to allow you to input your own parameters - but this is a bit dangerous since we will execute whatever you put so I want to do a little more work on it.  In the meantime, if you want to change your version of SAIL to issue -m 14 then open file /etc/e-smith/web/functions/sarkPCI and search for "fxotune".  Simply change the command string in the Perl script and save it back.

Everyone - asterisk doesn't start automatically.  We have tracked this down to the rpm and the strange way in which rpms work during update and there isn't an easy way to fix it.   Without going into the detail, if you issue an rpm -Uvh to update smeserver-asterisk, you will lose the start-up  link.  To avoid this from happening;  first remove the old rpm with rpm -e and then add the new one with rpm -Uvh.

Everyone - QOS Settings.  Yes QOS is a bit voodoo.  SAIL is using a version of Wondershaper called HBC Wondershaper.  It responds to the TOS (type of service) bits we set in sip.conf and iax.conf to prioritise VoIP traffic outbound and inbound.  However, Herve's description is absolutely correct - we can only control traffic within the scope of the SME box.  Once it leaves us we are in the lap of the Gods.  This is why you should look for a high uplift line with LOW contention and also consider using a terminator rather than a VoIP carrier.  Here for testing we run an 8000/440 ADSL2 circuit at 1:5 and a 2000/256 ADSL circuit at 1:20.  We do not run cable because the uplift here is very poor.

re QOS settings - you can see the docs here

 http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter19

Basically you should throttle your uplift at about 75% and your download at 90-95%.  However, you can play around with these numbers to see what works best for you.  Set a big upload away (or maybe send a massive e-mail)  while using the phone and see what the effect is.  If you are getting jitter then you should throttle back a bit, if not then you can ease the throttle forward until you get jitter and then back off.  Don't run the test with IAX and the iax jitter buffer running (Doh!).

[SARK devs]

Hi Guys,

We've just released -418 up to the ftp site.  It has an improved fxotune algorithm and you can specify the line break character and the -m value at run time.  Also we've added an Early/Late terminate switch to Globals...  Back sometime in the -300 releases, we implemented a late terminate algorithm (some of you were asking to take the call later in order to minimise call costs to the caller).  Well it seems we did a good job...  a little too good.  As a result, for inbound mobile calls on ISDN lines, we terminate the call so late and so fast that the mobile network can't keep up and it can lose the first half second or so of the opening conversation.  So we've now had a request from one of our customers to put a delay in.  


So...  For TDM lines you can elect to have an early or late termination.


Kind Regards

Selintra

[Franco]

How should we upgrade?

Many thanks,

[SARK devs]

Hi Stuntshell,

How should we upgrade?

Any way you like .  

The issue with symlinks (covered above) is an smeserver-asterisk problem, not a SAIL problem.  You can rpm -Uvh or yum localinstall, your SAIL rpms.  We would recommend the following...

yum localinstall selintra-sail-2.1.14-418.noarch --enablerepo=base

This will also load any dependent modules you may be missing; for example, the latest versions of 2.1.14 require nmap to do their network sniffing.


Kind Regards

Selintra

[hervep]

Hi Guys,

We've just released -418 up to the ftp site. Selintra

Hi Selintra,

Just updated my miniitx test machine.
Maybe something wrong at my side, but 'custom app' does not seems to work anymore. Previous version I used (-414), was OK.

I see [system] to be included into [default] context.
Should be included into [internal] ... not ?

Code: [Select]


;extensions.conf

....

[internal]

include => parkedcalls
include => internal-presets
include => extensions
include => conferences

exten => _X.,1,agi(selintra,OutCluster,${EXTEN})

[default]

include => parkedcalls
include => internal-presets
include => extensions
include => conferences
include => system
exten => _0.,1,agi(selintra,OutRoute,Belgacom)
exten => _2.,1,agi(selintra,OutRoute,Main server)
exten => _5.,1,agi(selintra,OutRoute,Main server)
exten => _4.,1,agi(selintra,OutRoute,Main server)
exten => _6.,1,agi(selintra,OutRoute,Main server)

exten => t,1,Hangup

exten => h,1,Hangup

exten => i,1,Background(invalid)
exten => i,2,Hangup


....

;
;#####################################################################
;
; Customer Supplied Contexts below this line (if any).
;
;#####################################################################
;

;
; Customer Supplied Context system
;
[system]
; just a test

exten => *999*,1,Wait(1)
exten => *999*,2,System(/usr/sbin/asterisk -rx 'reload')
exten => *999*,3,Hangup;


Kind regards, and many thanks for your work

Hervé

[SARK devs]

Hi Herve,

Thanks for this.  I will look at this later today.

Best

Jeff

[SARK devs]

Hi Herve,

The reason this doesn't run is that it is preceeded with an *; -

*nnnn numbers are reserved for SAIL functions.  If you use a regular 3 or 4 digit number, it will work fine.

Best

Selintra

[sonoracomm]

Maybe I missed something...I usually do...

The Admin Guide, Chapter 16 page 1 suggests we use *nnn* (three digits) for custom presets.

G

Respectfully,

G