Topic: [Announce] Selintra-sail-2.1.13-256

[SARK devs]

hi guys

2.1.13-261 went up to the ftp site today.  More bug fixes etc etc.  

2.1.13 is very different to 2.1.11 and before.  You must read the docs before you try it and you must run it with either the anabri asterisk releases or the 1.2.9 releases.  There are still a few bugs which we will fix over the next day or two but it should come up OK with X100P, TDM and ISDN HFC Cards.  We'll try to get the docs up to date over the next day or two, particularly the screen shots etc.

2.1.13 is more or less the shape that the final V2 release will take.  There isn't much more functionality we want to put in before V3.  Please give it a go and see if you can break it.

Kind Regards

Selintra

[jester]

Selintra/Jeff,

I've installed the 2.1.13-261 version of SAIL. Where do i put the subscriber numbers (MSN) for my HFC card?

Kind regards,
jester.

[SARK devs]

Hi Jester

Where do i put the subscriber numbers (MSN) for my HFC card?

Go into trunks and choose "create".  On the next screen, from the drop down choose a carrier of "PTT_DiD_GROUP".   This will allow you to create a DiD Span of one, or more, contiguous MSN's.  Sail will then create a separate trunk for each MSN in the group so you can route each one as you wish.   If your DiDs aren't contiguous then create as many groups as you need to record all of your MSNs.

Many Telcos do not deliver the full dialled number (DNID) when delivering MSNs.  BT only delivers the last six digits of the DNID, others truncate leading zeroes and so on.  Your DiD Numbers should match what your Telco delivers.

p.s. Don't forget to go into the PCI cards panel and run initialize and probe/commit, the first time you bring sail up.

Kind Regards

Selintra

[jester]

Hi Selintra,

I'm sure i've tried this last night, but besides the auto-generated Zap1-1 and Zap2-1 trunks i don't recall seeing a newly created PTT_DiD trunk back in the 'path' dropdown menu's when creating a route.... but i'll verify this tonight to be sure it ain't my brain melting down due to the heatwave over here.

Kind regards,
jester.

[SARK devs]

HI Jester,

don't recall seeing a newly created PTT_DiD trunk back in the 'path' dropdown menu's when creating a route....

    No, you won't see them in the route.  (sorry mate, this is our fault for releasing the code before we got the docs up-to-date).  MSNs/DiD's are logical constructs.  In truth they are nothing more than entries in extensions.conf which route the inbound call to the correct destination.

For outbound, you choose either the zap group, which will use the first free zap circuit,  or the actual zap circuit(s) you want to use (zap1-1 or whatever).

Hope this helps.

[jester]

Selintra,

If that's the case, what MSN number is send along when dialing outbound ?! ... i don't see the logical link between a trunk and the number used when dialing out over that trunk.

kind regards,
jester.

[SARK devs]

Hi Jester

I was betting you were going to ask that.  

We need input on this.  In general, we don't send CLID on outbound.

comments?


Best

Selintra

[jester]

Hi Selintra,

The problem is that i have to little understanding of the workings of linux/asterisk/telco stuff. So i'm not able to 'see' the programming trouble behind this and am probably not seeing the bigger picture... but in my laymans thinking:

It's about a logical place where to (be able to) define an outbound caller id.

My first guess would be a outbound caller id field when creating a route, maybe even for every path, that would leave the most flexibility.

Otherwise i'd figure when adding a PTT_DiD trunk also assigning it to a (or more) channel or group. When creating outbout routes beeing able to select a PTT_DiD would hold the information where to place the call and what to use for the Caller ID. But this would probably mean trouble with PTT_DiD number ranges and less flexibility.

Probably my laymans thinking would screw up the entire PBX but anyway... just trying to help the best i can.

Kind regards,
jester.

[SARK devs]

Hi Jester

Thanks - this is good input.  

Were working on this at the moment.  In the meantime, can you make calls over your ISDN circuit without including the CLID?  If you just specify the Zap group for example?


Kind Regards

Selintra

[jester]

Hi Selintra,

No i can't make calls over my ISDN trunk, whatever number i try to call i allways get the telco's (KPN) voice saying: 'This number is not in use...' . Looking at the asterisk CLI it is a correct telephone number (my own mobile nr.) :

    -- Executing AGI("SIP/5000-a433", "selintra|OutRoute|ISDN Out") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (Zap/1/0612345678)
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called 1/0612345678
Jul 23 11:33:20 WARNING[3103]: chan_zap.c:8498 zt_pri_error: 1 updating callstate, peercallstate 2 to 1
    -- Zap/1-1 is proceeding passing it to SIP/5000-a433
    -- PROGRESS with cause code 100 received
    -- Zap/1-1 is making progress passing it to SIP/5000-a433
    -- Hungup 'Zap/1-1'
  == Spawn extension (internal, 0612345678, 1) exited non-zero on 'SIP/5000-a433'
    -- Executing Hangup("SIP/5000-a433", "") in new stack
  == Spawn extension (internal, h, 1) exited non-zero on 'SIP/5000-a433'


Kind regards,
jester.

[SARK devs]

Hi Jester

Looking at the bulletin Boards this seems to be a known problem with Bristuff 3.0.  We'll regess to an earlier version of bristuff (which seems to be the cure), however there is also a "fix" of sorts also available but it isn't published by junghanns.  

Are you able to take inbound calls on the ISDN lines?

Best regards and thanks for your help on this alpha, it is invaluable.

jeff@selintra.com

[jester]

Hi Jeff/Selintra,

Yes, i can receive calls but i haven't tested it thoroughly though.

An other thought: i dont' know what the status is of the mISDN but i wonder if these drivers wouldn't be a better way to go. I've read that Junhanns has placed a ROM check so it will work only with their original hardware (see: http://www.voip-info.org/wiki/view/zaptelBRI under 'known issues') maybe this is/will be causing trouble.

As with mISDN you don't have to patch anything, so as i understand it there is no need for a seperate cologne asterisk version. Also mISDN supports Junghans cards, but Bristuff does not support Beronet cards (see: http://www.voip-info.org/wiki/index.php?page=difference+junghanns+beronet) ... but as i said, i don't know the status of the driver, but Beronet promotes the use of it for their cards so they should be usable.[/url]

[SARK devs]

Hi Jester

Re mISDN - when we first looked at ISDN, there were quite a few reports that mISDN was not very stable with the 2.6 kernel so we went with junghanns simply because it was a relatively (sic) easy install.  As we learn more, we're coming to the conclusion that you may be correct.  

At the moment the junghanns implementation doesn't seem to be that far away.  You (and others) all seem to be able to receive inbound calls and outbound works insofar as you can get audio back from the network (even if it won't dial the number for you).  

I think we'll take it a couple of iterations further and also re-evaluate mISDN and perhaps put up a challenger.

Best

Selintra

[del]

Hi Jeff,
I have installed asterisk and it seems to be OK. I have setup a couple of extensions and I am using softphones for testing. One thing I can't work out is howto retreive voicemail.    I can dial *50* then 1111 (default password, I beleive) but nothing happens. I have tried 1111#, 1111*, #1111# and *1111* but it keeps telling me that login is incorrect, what I am missing here? Thanks.

Regards,
Del

[SARK devs]

Hi Del,

The initial password to retrieve voicemail is the extension number of the phone.  So, if you are retrieving voicemail at extension 5000, the password is also 5000.  

You can retrieve voicemail for a different extension to the one you are dialling on by doing *51* in which case the autoattendant will ask for the extension number as well as the password.  

Passwords can be changed using the voicemail advanced functions.

You can find a list of all of Sail's telephone keypad operations here...

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter20

Kind Regards

jeff@selintra.com