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

[NickCritten]

Hello Nick,

Sounds like you've been in the wars!

Aye!

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.

Is already 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.  

Correct. One Side is pretty much OK, the other goes Laggy and horrible

Are these lan-to-lan calls or is one party coming in over the internet?

LAN-LAN on apretty decent Switch,  One of the Machines is on Wireless and I can check tonight if that is a problem, but I've not had issues with anything in the past..

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?

Asterisk Box is a P4 1.7Ghz 384MiB DDR Machine.. All good quality components (Intel branded Mobo, Branded memory etc.)
According to top, the cpu barely wakes up during the calls.

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

[/quote]The router is a traditional No-NAT variety (Not 'standard' ADSL modem/router PAT stuff) and its not blocking anything, but i can set up a psudo-DMZ and connect over an ethernet connection to the outside IF of aquarius to test if it a problem...
I'll try your tests and get back to you this evening.  Thanks for the support guys!

[SARK devs]

One of the Machines is on Wireless

Before you do anything else, try it without the radio.



Selintra

[NickCritten]

One of the Machines is on Wireless

Before you do anything else, try it without the radio.



Selintra

Will do, the only reason I haven't before now is that I've had other VOIP systems (all Cisco stuff) running over the wireless without any problems whatsoever.

[Tib]

Hi

Has anyone got asterisk conected with SPA3000 I've found a few instructions on the web but none seem to work for me ... the SPA3000 is supposed to work as good if not better than the TDM400 etc as a pstn gatway.

If someone has had any joy please let me know the config ... especialy with the sail ver of asterisk ... thats what I'm using.

Regards,

Tib.

[SARK devs]

Has anyone got asterisk conected with SPA3000

Hello Tib,

The spa3k runs fine with asterisk.  We've got code in SAIL 2.1.10 which is supposed to make it easier to set up but we're not that keen on it so we've put a revised version into 2.1.11 which we are much happier with.  If you aren't using SAIL, the spa is still relatively easy to set up (at least as far as asterisk is concerned).  The trick is to treat it as two distinct devices; a trunk and an extension.  Unfortunately, the number of variables in the device is absolutely bewildering if you are unfamiliar with it.  The other problem is that the default mode of the device is quite some distance from that required for asterisk gateway operation.  We will put up a how-to on the www.selintra.com/docs wiki this weekend in preparation for 2.1.11 and this will cover the set-up, with screen shots.  Where we can't help is with disconnect detection for different countries.  This is a complex area and you may need to speak to your telco provider to get it running properly.

the SPA3000 is supposed to work as good if not better than the TDM400 etc as a pstn gatway

No, it's not as good as a well set-up TDM board, at least in our experience, but it's not bad once you find a set-up that works for you.  However, despite the bewildering array of settings, there are still some things which it just doesn't seem to be capable of, one of which is its apparent inability to detect monotone disconnect tones (as opposed to DTMF), something which many cable based systems give instead of the more traditional polarity reversal or CPC.

We'll put a note up here when the how-to is ready.

Best

Selintra

[JonB]

I managed to get my SPA-3000 working (well almost) with SAIL yesterday.

I can get * to answer the incoming PSTN call and forward to the appropriate extension.

Disconnects from the PSTN work fine and drop the call on the extension , however if the VOIP extension terminates the call first the PSTN caller does not get disconnected and eventually gets passed on to the operator extension voicemail.

Almost there tho.

Jon

[Tib]

Thanks guys,

I'll keep my eyes open for any advancements.

Ohh and this is a great bit of work you guys are doing selintra ... very nice setup .
I found this post yesterday ... it may help someone but I could get things to work with it even though they recon it does.

http://www.geekgazette.com/index.php?option=com_content&task=view&id=28&Itemid=26&limit=1&limitstart=2

Regards,

Tib

[i4daniels]

Hi,

Very, very, very nice contrib to SME7! thanks a million!

Is it possible to switch the behaviour of the Queue so it emits a ring rather than play the MP3 tracks? how would you remove the "Please hold while we connect you?" and just immediately ring the group of phones!

I've used Asterisk for the last three years and know my way round the configs but changes i'll make would probably be wiped out on the next GUI config update.  Could we just edit the files through the GUI?

I'm running this on a ML370 G2 using Aastra 480i VOIP phones and Telipliant (uk) and it runs without error.

Pure class!

Dan Scannell
www.i4tech.co.uk

[SARK devs]

Is it possible to switch the behaviour of the Queue so it emits a ring rather than play the MP3 tracks? how would you remove the "Please hold while we connect you?" and just immediately ring the group of phones!

Hello Dan.  

Short answer is... don't know, I'm the only one in the office today (Sunday) and I didn't write the queuing stuff.  I'll ask the lead developer to drop a line in here tomorrow.

Could we just edit the files through the GUI?

Er...Not sure we understand, you can directly change just about everything we control;  SIP, IAX, Voicemail, MOH, conference rooms, features, agents & indications can all be modified directly through the GUI.  Simply change, delete or add tuples in the string-editor box.   We will also expose extensions in the 2.1.11 release so you can code your own extensions.conf contexts.  Which particular file(s) did you want to edit and we'll see if we can't help?  In 2.1.11, we've upgraded our text editor which means we can bring the entire /etc/asterisk directory under user-updateable control very easily.  That will probably happen in 2.1.12 and it will just mean the extra entities will show up in the Headers section.

Hope this helps

Selintra

[i4daniels]

Now that's what I call service!

You usually change the option in the extensions.conf file so that it returns a ring rather than play music on hold

From the manual:
Queue -- 'r' — ring instead of playing MOH

So, I would imagine I could edit the extensions.conf file using the server-manager component, add the r switch into my conf file and this would update my /etc/asterisk/ files ?



Dan Scannell
www.i4tech.co.uk

[SARK devs]

So, I would imagine I could edit the extensions.conf file using the server-manager component, add the r switch into my conf file and this would update my /etc/asterisk/ files ?

Hello Dan,

Pretty much, although changing extensions.conf isn't an option because there aren't any dial or queue statements in there.  So... we're putting a little mod into the queues panels in 2.1.11 to expose the options so you can set them yourself.

Hope this helps

Selintra

[JonB]

Selintra,

SME7rc1 breaks Zaptel due to the new kernel 2.6.9-34.EL.


Jon

[hanscees]

hi, two things.
- Is there a possibilty someone could make an howto for dummies (or point me to one). I don't know much about.....asterix. What is the fuctionality? Why do I need it?

- If you have performance issues in the howto section for sme7 I have posted  a howto on traffic shaping. I heard it makes your calls better, since it gives packets with specific bits a quicker response. At my site I also have an iptables script you can use that does show dropped packets.

Hans-Cees

[SARK devs]

SME7rc1 breaks Zaptel due to the new kernel 2.6.9-34.EL.

Thanks Jon,

We downloaded RC1 overnight.  We'll recompile zaptel with the new kernel and get a compliant rpm up sometime today.  


Selintra

[SARK devs]

Is there a possibilty someone could make an howto for dummies (or point me to one). I don't know much about.....asterix. What is the fuctionality? Why do I need it?

Hello Hans

There is a very good forum here

www.voip-info.org

You can find everything you need to know about asterisk and VOIP in general on this site.

- If you have performance issues in the howto section for sme7 I have posted a howto on traffic shaping.

Thanks for this, we'll take a look.  In general, because asterisk uses UDP, which is a bit of a packet "bully", we've found that packet congestion has not been an issue.  However, we do plan to revisit this area.


Kind Regards

Selintra