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Asterisk work

achandra

Asterisk work
« on: February 22, 2006, 11:40:06 PM »
I have seen the directions for SAIL and Asterisk implementation on SME 7.

Which version is more stable? I had some issues with installation of the Asterisk Modularized version complaining of log file locations missing on install. This then caused Asterisk not to start up.

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« Reply #1 on: March 03, 2006, 12:23:03 AM »
Hi,

Which log files were missing and which asterisk rpm were you using?

As far as we know the .85 build -  asterisk-SME7-1.2.3-85.i686.rpm is good to go. We've done twenty or thirty cycles with it now and we've not had problems like the ones you describe (doesn't mean they aren't there, just that we havn't seen them in testing).  However, we do stipulate that you must not install it over a previous manual or rpm asterisk install.  This is because we don't build new etc/asterisk if it's already present on your system.  

For SAIL, the .90 build is OK but we will be putting up a new rpm on friday/saturday which includes queues, agents and graphic stats.  This will be a 2.1.10 release, probably build ninety-something depending upon how it goes in final test - the bigger the number, the more crap we found - :-)

Best

Selintra

achandra

Hello
« Reply #2 on: March 03, 2006, 01:01:04 AM »
Hi,

Thanks for the answer. I looked through the Selintra guide quite a bit. what I couldnt understand from it ( likely my own confusion), is where to get the packages for install.

But as you've stated you're likely going to post these in a download section?

As for the other little tid bits of sip natting the 5060 and the other necessary ports thats all good to go?

Any QOS work in the works?  :-D

Finallyy...(phew...thanks for your patience), how about security?? -open vpn integration or??

[/u]

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« Reply #3 on: March 03, 2006, 01:35:46 AM »
Quote
As for the other little tid bits of sip natting the 5060 and the other necessary ports thats all good to go?

Any QOS work in the works?  

Finallyy...(phew...thanks for your patience), how about security?? -open vpn integration or??


The guide discusses port management in some detail, however in broad outline, SIP NAT is not an issue as long as your carrier is running session border control.  A remote handset coming in SIP through NAT is still a challenge since we haven't any SBC of our own (although we are looking at it).  We have made inter-node setup and management extremely easy so you can iax trunk off of remote SARK/SAIL instances to help get around the problem.  Only other solution is to run an IAXy or an IAX softphone.

We do set the QOS bits as part of SIP generation but we aren't running any chokes, although we might in the future.  We've played with various versions of shaper.  Problem with QOS is you can only police your own downstream traffic, you have no control once you let the packet go :-(. Furthermore, our experience with UDP seems to suggest that it may not be necessary for us, at least against TCP congestion. UDP is usually a big enough bully to get through even against heavy traffic.  Having said that, there's no issue  if you want to fit a shaper to SME yourself.

Tell us what security you would like and why.  

Thanks for your interest.

Selintra

achandra

RTP
« Reply #4 on: March 03, 2006, 01:43:17 AM »
Thanks for the detailed explanation.

I guess these are the complexities of dealing with UDP and RTP in a layer 7 type scenario.

My main concern if any is that Im attempting to tunnel two asterisk boxes in to remote places. In this fashion I dont have to deal with issues such as certain countries not allowing sip communications, etc. Also since its tunneled it has a "higher" level of security.

The problems encountered here are posing more of an 'ah ha' type scenario and the complexities of what I get or dont get with trying to install AAH on an all inclusive box.  

QOS and Security in terms of tunneled connection  are important to the small business I am dealing with.

Im not sure if you have any suggestions on the design of this.

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« Reply #5 on: March 03, 2006, 12:37:50 PM »
Quote
Im not sure if you have any suggestions on the design of this.


I'm not sure we have to be honest.  We shied off tunneling VOIP bacause we saw a few horror stories re thruput and, for most users, it's a fairly tough call to create such a comms infrastructure.  With SAIL's inbuilt Sibling support, to set up an internode IAX2 link with two SAIL instances requires just 4 fields to be filled out and a route to be created. Having said that, we'd be delighted to have you give it a go with SAIL because it does as you say, solve a lot of problems with SIP/NAT.[/quote]

Offline gregswallow

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« Reply #6 on: March 04, 2006, 09:44:29 AM »
Where is asterisk-SME7-1.2.3-85.i686.rpm from?

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« Reply #7 on: March 04, 2006, 10:36:19 AM »
It's ours.  We built it specifically for CentOS 4.2 with udev.  It's really there to support SAIL but it is pure Asterisk - you can use it as-is if you just want an asterisk install for SME or CentOS.  It includes Asterisk, the asterisk add-ons, asterisk sounds, zaptel and libpri.  There's a small mod from us to the daemon to modprobe for TDM & X100 boards (if it doesn't find any, it loads ZTDUMMY) and then to wait while udev builds the device trees.  Other than that, it's vanilla.  Only word of warning is that it was built against the -22 kernel which means that it won't run on sme 7.0 releases earlier than pre2.  If in doubt, uname -r your kernel to check.

Offline gregswallow

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« Reply #8 on: March 04, 2006, 08:49:03 PM »
Sorry, I meant where can it be downloaded?

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« Reply #9 on: March 04, 2006, 11:01:38 PM »
Sorry Greg :-)

It's on our ftp site...

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

Please be patient - it's a big rpm and its mounted on a spare ADSL line we have so it will take about 10 minutes to download assuming you are the only person on-line.

On a clean Pre3 system Asterisk should just come up and run after the install.  

If it's of general use, you are welcome to it as a contrib.

The 2.1.9 SAIL beta is also on the ftp site so we guess some of you might want to give it a whirl.  It's not mandatory but we'd really appreciate it if you'd register with the Wiki before you download it.

Full documentation is at www.selintra.com/docs

Let us know how you get on.  

Selintra

Offline gregswallow

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« Reply #10 on: March 05, 2006, 08:34:07 AM »
Thanks, I registered and downloaded it.  Looks great, I think it's the only rpm-based Asterisk contrib for SME.  People have been trying to modify Asterisk@Home for SME, but it's a pretty clumsy scripted install.

Can I download the src rpms as well from somewhere?  By the way, someone can set you up a ftp directory on contribs.org if you'd like to save your bandwith.  Just send an email to staff_at_contribs.org and ask.

Also, if you want, the bug tracker on contribs.org has a category for contribs now.  I would really encourage you to take advantage of that (just create a bugzilla account and requst a subcategory for your contrib).  The bug tracker is much more efficient for getting feedback for users of your contrib than asking people to email you with comments, and another side-effect is you'll get suggestions and help from developers of SME.

Offline JonB

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« Reply #11 on: March 05, 2006, 09:19:27 AM »
Selintra,

This is nice work.

A couple of comments though.

I installed on a SME7 pre3 system that did not have a zaptel card. ztdummy did not automatically load. This of course meant that meetme conference did not work. I had to run

Code: [Select]
# modprobe ztdummy

twice before ztdummy loaded.

You do not expand httpd.conf and restart httpd during the install of the SAIL rpm. This means that http.conf does not include the template fragments for Flash Operator Panel so FOP is not available.

On a similar note your FOP template fragments assume a local IP address of 192.168.1.0/255.255.255.0. You may be better using the variable $LocalIP.

I noted also that none of the IAX2, SIP or RTP ports are opened on the server. This is fine if one only wants the server as a local PBX with no external access.

It can be rectified by

Code: [Select]
config set IAX2 service status enabled access public UDPPort 4569
config set SIP service status enabled access public UDPPort 5060
config set RTP service status enabled access public UDPPort 10000:20000  
signal-event remoteaccess-update


Keep up the good work!

Jon
...

achandra

Now were getting some where
« Reply #12 on: March 05, 2006, 09:35:55 AM »
Sweetness :-D

This is all good news and I'll give it a whirl . Thanks for answering all my questions.  8-)

Looks like a lot of hard work went into this.

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« Reply #13 on: March 05, 2006, 12:54:14 PM »
Quote
I installed on a SME7 pre3 system that did not have a zaptel card. ztdummy did not automatically load. This of course meant that meetme conference did not work


Hi Jon thanks for your interest.  Apologies - we should have referred everyone to the Quickstart installation guide on our docsite...

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

We recommend doing console-save after the install.  This is to re-instantiate httpd and to force a rebuild of configuration because we use the new 7.0 configuration tree method of describing and enabling the services - more of which below.

Can we ask if you installed SAIL against your own version of Asterisk?  The reason we ask is that the Asterisk-SME7 daemon (in our rpm) does do the modprobes and loads for the Digium Hardware.  The rpm also creates and enables the service.  We don't do this in the SAIL rpm, hence the question.

Quote
On a similar note your FOP template fragments assume a local IP address of 192.168.1.0/255.255.255.0. You may be better using the variable $LocalIP.


Oops! - guilty as charged - we'll fix it Jon :-)

Quote
I noted also that none of the IAX2, SIP or RTP ports are opened on the server. This is fine if one only wants the server as a local PBX with no external access.


The console-save should open the ports during the configuration rebuild.  Once again, we do that in the Asterisk rpm.  However, we have another guilty secret; in the 85 rpm there is a little bug....  We spelt the name of the bloody file wrong!  Instead of UDPPorts it's in there as UDPorts.   :cry:  ( You just can't get the staff these days - the execution is set for tomorrow morning at dawn).  

It's fixed in the 100 build. If you've already downloaded the 85 build, don't bother downloading the 100 just do

mv /etc/e-smith/db/configuration/defaults/asterisk/UDPorts \
/etc/e-smith/db/configuration/defaults/asterisk/UDPPorts



Thanks very much for the input Jon - we really appreciate it.

Selintra

Offline JonB

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« Reply #14 on: March 05, 2006, 01:28:42 PM »
Selintra,

Thanks for the reply. I did have a look at the Quickstart but missed the console-save. Teach me not to RTFM properly :oops:

I used your .85 rpm to install *.  I had previous problems with getting ztdummy to load on this server with A4SME7.

It is a brand new ASUS MOBO with a Via vt8237 chipset. Under lsmod there is no usb_uhci module loaded.
The usb modules loaded are uhci_hcd and ehci_hcd however ztdummy should load with these usb modules.

It took 2 attempts to modprobe ztdummy. The first time there was errors and the second time was ok.

Maybe it would be worth while looking at zaprtc instead of ztdummy. zaprtc uses the Real Time Clock built into all modern motherboards instead of using the USB hub timing.

Look forward to the new rpm.

Jon
...

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« Reply #15 on: March 05, 2006, 06:02:52 PM »
Quote
Can I download the src rpms as well from somewhere?


Greg

We've loaded the Asterisk-SME7 srpms (85 & 100) to our ftp site for you.

Selintra

Offline gregswallow

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« Reply #16 on: March 05, 2006, 09:21:40 PM »
Hi, the sail src rpm too please.  Thanks.

achandra

Looking Good
« Reply #17 on: March 06, 2006, 09:05:12 PM »
Hi,

I spent time reading through the "how to/install guide". It looks very very good.

The section that was VERY VERY useful was the excellent writeup on lossy vs. lossless compression codecs. In addition the breakdown chart is very very useful as well.

Id like to ask about the SARK iso that you have to offer...is that sme and sail all on one disk? If so is it omptimized specifically for Asterisk/PBX type setup? - AND....where can I get the ISO??