Topic: [Announce]SAIL 2.1.10-115/Asterisk-SME7 i386

[SARK devs]

Hello Everyone,

Thanks for your response to SAIL, our FTP server has been on its knees since tuesday!  

Just a quick note to let you know that we'll put up two new rpms this afternoon.  

Asterisk-SME7-1.2.3-100.i383

Build 100-i383 is an alpha release for non-686 chips, in particular the dinky little VIA C3 mini-ITX  boards.  It chugs along happily on our little VIA Eden.

selintra-sail-2.1.10-115.noarch

Build-115 fixes the MySQL logging problem reported by JonB (thanks Jon) allowing Areski Stats to work properly, plus we've cleaned up a few rough edges in the agi.

You can install this rpm straight over the old one (113).  However, if you do,  please run

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perl /etc/selintra/selnav

after the install.  This addresses a minor snafu in our previous rpms.  You shouldn't have to run this in future upgrades (we hope).

Many Thanks


Selintra

[JonB]

You will also need to do

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cd /home/e-smith/db
mv -f selintra_old selintra

otherwise you will find that everything that you set up has disappeared.

Go into extensions, choose a phone and hit update. This will rebuild the asterisk .conf then

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service asterisk restart

Jon

[SARK devs]

Well spotted Jon...  You wouldn't like to come and live in the UK would you?

Only joking.  Seriously tho' we really appreciate the input.

We've found the problem in the rpm.  Pure finger trouble on our part coupled with the fact that we were anxious to get a fix out for Stats.  We've already cut a new version and we're going to run it through a few cycle tests here, so it should go up onto ftp later today if it hangs together.

Please accept our apologies.

Selintra

[JonB]

Like all good Kiwi's I've spent time in the UK.  

No need to apologise. Thats what bug testing is all about.

Jon

[NickCritten]

Hello again,

A very minor bug:

In the iax.conf header you have got:

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srvlookup=yes
maxexpirey=180
defaultexpirey=160


These aren't valid iax.conf variables (according to voip-info.org & asteriskguru.com anyway!)

Also in several of the IAX2 IP Devices you have:

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canreinvite=no
This should be

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notransfer=yes


This really is a wicked package you guys have put together.  I can't wait until you've finished with the advanced IVR stuff!

[SARK devs]

Thanks for the input and the compliments Nick, we're blushing here!

We don't know IAX that well.  We'll look into these.  You can change your own settings just by altering them in headers.

We have IVR working and it looks nice.  We're also putting a feature in for 2.1.11 called Custom Apps.  Basically its a window onto extensions.conf where you can define and code your own contexts.  We'll document the agi at the same time to allow you to make your own calls to it so, for example, you can use our Route/Trunk and Dial logic etc. if you wish.  

Kind Regards

Selintra

[SARK devs]

Just to let you know.  We've put selintra-sail-2.1.10-116.i686.rpm up onto the ftp site now.

Many Thanks

Selintra

[dennyhalim]

i got few questions about sail:
- how it compare to a@h?
- how to upgrade? do automatic update also works for sail?
- my sme behind firewall, which ports must i open?

tia
dny

[SARK devs]

- how it compare to a@h?

Hello Danny

A@H is a collection of tools for Asterisk, one of which; AMP, is an Asterisk configuration workbench. SAIL is an Asterisk configuration workbench integrated into the SME server-manager.

how to upgrade? do automatic update also works for sail?

Right now rpm, soon, YUM.

my sme behind firewall, which ports must i open?

Same ones you would open for any asterisk system.  See here for a discussion

http://selintra.com/docs/cgi-bin/view/Main/NetInst

Selintra

[JonB]

For the Kiwi's out there running Asterisk, get the Kiwi voice prompts at

http://voiceprompt.archnetnz.com/index.php?p=archive

No longer will you have to press the "pound" key.

Jon

[JonB]

Any one got a Sipura SPA-3000 working with SAIL.

I used to have it working with A@H but can't get it working with SAIL.

The SPA-3000 configuration info would help.

Apart from the SPA everything is working perfectly. I have 2 boxes set up, home and office. SAILtoSAIL trunks work perfectly. Much easier than setting up IAX2 trunks using extensions as peers.

Now to get Jive Messenger integrated via the Asterisk plugin and I should be home and hosed.

Jon

[SARK devs]

Hi Jon,

Glad it's running well.  We're particularly pleased with the SAIL-to-SAIL trunking, it works very well.  We'll spin up an spa3K today and give you our settings (we've got an spa2K lying around somewhere so we'll spin that up as well).


New release (2.1.11) at the weekend all being well.

Kind Regards

Selintra

[SARK devs]

Any one got a Sipura SPA-3000 working with SAIL.

I used to have it working with A@H but can't get it working with SAIL.

The SPA-3000 configuration info would help.

Hi Jon,

We've put an spa3K up this morning without too much trouble, however, it has shown up a few things in the ata handling that we don't like.  This area of the code was retrofitted from Version 1 of SAIL and, to be honest, we didn't test it as well as we should have.   Here's the plan....

We will put a full mod into 2.1.11 which is currently in packaging.  In the meantime, if your requirement to get the spa3K running is urgent then send us an e-mail at admin@selintra.com and we'll send you a temporary config which will get you up and running.  We'll also put a section into the wiki showing how to set the SPA series up - all of 'em; 1K, 2K, 3K and 841.

Best

Selintra

[NickCritten]

Hi Everyone,

By and large everything is great!
However I'm having a few problems and I've been Googling them for a few days with no luck so here goes:

Setup:
ADSL Router =>
SME7pre4 in Server-Gateway mode (Aquarius) with static IP=>
LAN (192.168.30.0/24) =>
SME7pre4 in Server-Only Mode with Asterisk & SAIL installed (avatar)

PortForwarded UDP4569 from Outside to avatar


Problems:
1) Both SIP & IAX Softphones: (I don't have any hardphones for testing yet )
Start a conversation, it starts off fine, Lag is tiny... gradually over the course of a few minutes, the Lag from one side of the conversation increases all the way up to nearly 10Seconds!, and the quality of the audio goes really bad.  I've check and doublechecked all the settings for the devices on the server and on the PC's that are running them but I'm stumped.  What kind of testing / troubleshooting should I be doing?  I wanted to get this going myself, but after nearly a week its still a problem.


2) IAX Client from Off-Network (Coming in over Internet)
IAX Softphone when on the LAN registers with absolutely no problems.
If I try to connect over the internet, the asterisk server just does not respond to the registration packets.
I've been running TCPDump and I can see the UDP4569 Packets hitting the asterisk server, but it doesn't seem to respond at all.
I've tried turing on IAX Debug and running the asterisk CLI at full verbosity but it doesn't seem to react at all to the registration request.

My only theory is that the SME firewall on the asterisk box is dropping the packets, but without a firewall log, how do I check this? (ive searched contribs for 'firewall log' 'denylog' 'dropped packets log' and a few others)


I'm sorry if theres something obvious, but i really have tried... just boot me in the right direction please!

[SARK devs]

Hello Nick,

Sounds like you've been in the wars!

gradually over the course of a few minutes, the Lag from one side of the conversation increases all the way up to nearly 10Seconds!,

We've never seen this happen with asterisk, however there are a few posts on the BB's regarding GSM over internet.  Try running your tests with SAIL set to fidelity mode.  On the face of it, it looks like a resource problem somewhere.  We have a few questions....

You say "one side of the conversation", do you mean one particular party is lagging but the other is OK? Please clarify.  Are these lan-to-lan calls or is one party coming in over the internet? If so, what bandwidth do you have and what codec are you using? What is your cpu usage during the lag phase?  How much memory do you have on the box and what cpu speed?

I've been running TCPDump and I can see the UDP4569 Packets hitting the asterisk server, but it doesn't seem to respond at all.

Have a look at what's happening by running ipchan on the asterisk box.  It will show you the packets that your machine is seeing in realtime.  If you have a traditional router, try removing the server-gateway machine from the scenario and use your router to forward the 4569 packets directly to your server-only machine.   To check whether 4569 is open run either...


netstat -anp | grep 4569

or

lsof -i | grep 4569

These will tell you the port state and the PID of the application that opened it.

Let us know how you get on.

Selintra