Topic: [ANNOUNCE] selintra-sail "Cologne" alpha

[SARK devs]

Hi all,

selintra-sail-2.1.12-239 Cologne is now ready for you to test/play with/break.  There are two large new features in Cologne so you WILL find errors/bugs.

Cologne currently requires a new set of asterisk rpms which are prefixed "anabri".  You will find them on contribs here

http://mirror.contribs.org/smeserver//contribs/selintra/RPMS/

Cologne can be downloaded from here

ftp://81.149.154.14/

Installation notes can be found on the docs site here

http://81.149.154.14/docs/cgi-bin/view/Main/SysVersion2Release1Issue12

You MUST read the release quicknotes before attempting to install Cologne.

NEW FEATURES

Cologne supports ISDN HFC cards and DiD number ranges.  If you have an ISDN line then please give the software a go.

Cologne supports Universal Provisioning.  In theory, any IP device can be configured using the new templating sub-system.  Go on - see if you can break it!  We'll be delighted if you do because that will mean we're all still learning.

Kind Regards

Selintra

[jester]

I see i've got a funfilled weekend ahead  , thanx guys!

But watch what you're saying about breaking stuff.... this noob is good at that!
(one nutcase can break more than 10 wizzards can fix)

jester.

[jester]

Hi Selintra,

I don't know if maybe the servers are not in sync yet, but the anabri-asterisk-zappri-x.xx.xx file is not on there... so no testing this weekend...  

jester.

[SARK devs]

Hi Jester

Zappri went up to the mirrors a little late.  Should be there soon.



Best

selintra

[chris burnat]

jeff,
the alpha quicknotes state:
You must first remove any existing asterisk rpms from your system e.g.
rpm -e selintra-asterisk
rpm -e selintra-asterisk-zappri-MPP

What I have on my system at the moment are:
1 all project 31860 Jun 7 09:03 perl-Term-ReadKey-2.21-1.1.el3.rf.i386.rpm
1 all project 3355261 Jun 7 09:03 smeserver-asterisk-sounds-1.2.2-2.noarch.rpm

1 all project 15962267 Jun 7 09:03 smeserver-asterisk-1.2.3-2.i686.rpm
1 all project 1848688 Jun 7 09:03 smeserver-asterisk-zappri-MPP-1.2.2-1.i686.rpm

1 all project 1329215 Jun 24 15:11 selintra-sail-2.1.11-214.noarch.rpm

And so I should leave the 1st and 2nd rpms on the list alone alone, remove the 3rd and 4th rpm and reinstall using the anabri versions, and upgrade the 5th with 2.1.12-239, correct?

Followed by:
/etc/e-smith/events/actions/navigation-conf

Thanks
chris

[SARK devs]

Hi Chris

perl-Term-ReadKey-2.21-1.1.el3.rf.i386.rpm  - not used in 2.1.12 - leave or remove, up to you.
smeserver-asterisk-sounds-1.2.2-2.noarch.rpm - fine as-is

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

selintra-sail-2.1.11-214.noarch.rpm - fine as-is

Yes do the navigation-conf run (I'll update the alpha notes).

Best

Jeff

[chris burnat]

Thanks jeff, done it and working fine, meaning nothing appears broken.
One question, what was the difference between zappri and zappri-MPP?
I notice that you onl;y have zappri, not MPP in the new rpms.
Best.
chris.

[chris burnat]

Jeff,
(i) I did a restart just a moment ago because I had carried an update for SME - full reconfiguration in fact..  OK, asterisk would not restart. I have no card, all the same went throught the motions detailed about PCI setup, no go. The trick was to save global settings, and all is well now.

(ii) In global settings, there is a warning:
Operation status report -  TFTP server is not installed!
It would be good having some guidance on this,
Regards
chris

[SARK devs]

One question, what was the difference between zappri and zappri-MPP?
I notice that you onl;y have zappri, not MPP in the new rpms.

Hi Chris

The new rpms are based upon the Junghans "BriStuff" asterisk.  Currently there is no MPP version of their work.  This means that ISDN is UP only for now.  We'll put out a new set of ordinary asterisk rpms (next week I hope) for 2.1.12 users who don't want or need ISDN HFC BRI.  They will have MPP support.  

Operation status report - TFTP server is not installed!

Apologies.  The provisioning subsystem needs tftp-server to be installed for it to work.  You system will still run without it but you won't be able to provision any phones or ATA's.  To fix it simply enable CentOS base and  install tftp-server.

Code: [Select]

yum install tftp-server

Best

Jeff

[jester]

Jeff/Selintra,

Just in advance: i'll probably be asking a lot of stupid questions because i'm new to this.... so here's the first one:

Is it possible to assign more PSTN/DID numbers than 2 to an ISDN telephone line?! In holland we get 4 numbers with an ISDN2 line.
 
Btw if we find some bug (not that i have yet), my guess the best place is the bugtracker?! If so it might be good to state that at the top of this post before everything get's incoherently dumped in these forums.

Love it up till now by the way! Still figuring out how to get my line up yet but still love it

jester.

[SARK devs]

Hi Jester

You can create as many DiD's as you wish.  Simply create a new trunk and choose a carrier type of PTT_DiD_Group.  It will lead you through setting up the rest of your DiDs.  Have a look here in our docs...

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter095

Re Bugs - we are still deciding whether to use the good folks at contribs' bug tracker or to make ours available to everyone.  

Best

Selintra

[jester]

Selintra/Jeff,

Is there a thought behind creating 4 trunks of a ISDN line in FOP when the detected pci card only has 2 channels? This will clutter FOP really quickly on larger configurations.


Kind regards,
jester.

[SARK devs]

Hi Jester,

Is there a thought behind creating 4 trunks of a ISDN

Some, but not much .  To be honest, we haven't had the time to devote to FOP that we would have liked.  The idea behind the four trunks was simply a guess on our part as to how many active inbound lines you might have at any one time on any one trunk.  If you only create one trunk per connection type/VOIP Account, FOP won't let you do "drag-and-drop" transfers except for the first call.  

The FOP screen is really just eye-candy in our opinion and not good enough to deploy in a production environment.  Sometimes it works, sometimes it doesn't and because it's all written in one big perl module plus Flash, it's not an easy thing to pick up and modify.  We will shortly look at HUD to see if that does a better job.

Are you making and taking calls OK over ISDN?  What kind of call quality are you getting?

Best

Selintra

[jester]

Hi Jeff/Selintra

If you'd cut down the number of trunks to the number of channels of the ISDN line would seem sensible... but if FOP is going to be thrown, just a waste of time.

I'm happily receiving calls, but i can't figure out how to make calls... all i'm getting is a Dutch operators voice saying that i'm trying to call an unknown number... i think my brain is failing me, i thought i knew my own mobile number.   . If i look at the *CLI lines passing by it seems to be the correct tel.no.:

Code: [Select]

   -- AGI Script Executing Application: (Dial) Options: (Zap/g2/0666666666)
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g2/0666666666
Jul  2 18:41:12 WARNING[7793]: chan_zap.c:8498 zt_pri_error: 1 updating callstate, peercallstate 2 to 1
    -- Zap/1-1 is proceeding passing it to SIP/5000-b58b
    -- PROGRESS with cause code 100 received
    -- Zap/1-1 is making progress passing it to SIP/5000-b58b
    -- Hungup 'Zap/1-1'

It's probably just me messing something up.

I'm calling with an Tiptel usb phone through a wireless connection from my laptop to my mobile.... so it's not fair to say anything about the sound quality. I hope to get a chance of testing a set-up at my work with a proper phone and connection. I'll report it back if i do so.

Kind regards,
jester.

[SARK devs]

Hi Jester,

Excellent work.  Thanks for this.  We've used your output t find what we think may be a small bug in the Jungenns code.  We'll fit it tomorrow and get back with you.

Kind Regards

Selintra

[marcelb]

Is there anyone help me debugging

The asterisk message file give me the following lines:

Jul  3 13:54:21 WARNING[13189] chan_zap.c: Ignoring switchtype
Jul  3 13:54:21 WARNING[13189] chan_zap.c: Ignoring signalling
Jul  3 13:54:21 WARNING[13189] chan_zap.c: Ignoring pridialplan
Jul  3 13:54:35 WARNING[12187] chan_zap.c: 1 TEI remove TEI = 127
Jul  3 13:57:35 WARNING[12187] chan_zap.c: 1 TEI remove TEI = 127
Jul  3 13:57:35 WARNING[12187] chan_zap.c: 1 TEI remove TEI = 127
Jul  3 14:00:35 WARNING[12187] chan_zap.c: 1 TEI remove TEI = 127
Jul  3 14:03:35 WARNING[12187] chan_zap.c: 1 TEI remove TEI = 127

Regards Marcel

[SARK devs]

Hi Marcel,

From your post we can't tell if you have a problem or not and what the symptoms may be.  Can you tell us a little more about what is happening?

Thanks

Selintra

[marcelb]

Hi Marcel,

From your post we can't tell if you have a problem or not and what the symptoms may be.  Can you tell us a little more about what is happening?

Thanks

Selintra

Selintra

We make a start from the begin.
I’m using a Dell server with Sweex ISDN board.

As the documentation told us:

Menu option PCI Cards

-   Start Initialize OK
-   Probe OK (no telephone numbers recognized)
-   Load OK
-   Stop and Start asterisk OK

Start with a new trunk and give it the telephone number OK, Dialling this number from another network give a busy tone. Dial out is not possible, warning 4230 unable to forward voice, hangup Zap/2-1

After this dial out test the D-Channel is down as I described below. This test is about 1 minute, after that the D-Channel is up.

D-Channel test
The asterisk monitor showing me 14 times == Primary D-Channel on span 1 down
After 14 times the next time give me == Primary D-Channel on span 1 up. After that we can see 1 received TEI check request for TE1 = 65 or 80

When we wait about 5 minutes the Asterisk monitor give us:
Warning[3869] chan_zap.c:8498 zt_pri_error: 1 TEI remove TEI = 127
This line 2 times and after that
== Primary D-Channel on span 1 up

Direct access to play around on my test system is possible!

Regards Marcel

[marcelb]

Hi Marcel,

From your post we can't tell if you have a problem or not and what the symptoms may be.  Can you tell us a little more about what is happening?

Thanks

Selintra

Selintra

We make a start from the begin.
I’m using a Dell server with Sweex ISDN board.

As the documentation told us:

Menu option PCI Cards

-   Start Initialize OK
-   Probe OK (no telephone numbers recognized)
-   Load OK
-   Stop and Start asterisk OK

Start with a new trunk and give it the telephone number OK, Dialling this number from another network give a busy tone. Dial out is not possible, warning 4230 unable to forward voice, hangup Zap/2-1

After this dial out test the D-Channel is down as I described below. This test is about 1 minute, after that the D-Channel is up.

D-Channel test
The asterisk monitor showing me 14 times == Primary D-Channel on span 1 down
After 14 times the next time give me == Primary D-Channel on span 1 up. After that we can see 1 received TEI check request for TE1 = 65 or 80

When we wait about 5 minutes the Asterisk monitor give us:
Warning[3869] chan_zap.c:8498 zt_pri_error: 1 TEI remove TEI = 127
This line 2 times and after that
== Primary D-Channel on span 1 up

Direct access to play around on my test system is possible!

Regards Marcel

Selintra

The latest results of the beginners test:

Dial-IN is working now and it is perfect! I was searching what went wrong with the telephone numbers, it looks like the leading zero, but there was no leading zero at all in the received number. (it is 32 degrees in here)

So, When I dial out the D-Channel test start over and over.

To be continued.

Regards Marcel

[SARK devs]

Direct access to play around on my test system is possible!

Hi Marcel,

Thanks for this we'd love to get access to your system.  As you know, we only have an ISDN simulator here, we have no real ISDN lines.

Please e-mail details to me at jeff@selintra.com

We'll arrange a time to come on to the system either by e-mail or over the phone.

Thanks again.  This is great!

Kind Regards

jeff@selintra.com