Topic: Add a carrier profile to SAIL

[gippsweb]

Hi Selintra
I've just installed SAIL onto my SME box and am slowly getting my head around the click and go config.

Would it be possible to get Faktortel added to SAIL, it's config for Asterisk is on the following pages;
http://www.faktortel.com.au/support-asterisk-iaxconf.shtml
http://www.faktortel.com.au/support-asterisk-extensionsconf.shtml
and
http://www.faktortel.com.au/support-asterisk-sipconf.shtml


Thanks

Actually, now that I think about it. Acouple of carriers I'd like to have in there is sipbroker and voxalot. That'll give us direct enum lookups and calls between carriers, which is rather difficult at the moment.

[SARK devs]

Would it be possible to get Faktortel added to SAIL, it's config for Asterisk is on the following pages;

Hi gippsweb

Yes of course.  We'll put them on the list.  There's a release due at the end of this week so we'll try to get them into that.  

 

...sipbroker and voxalot. That'll give us direct enum lookups and calls between carriers, which is rather difficult at the moment.

Can you expand on this?  What is the difficulty you face? Let's see if we can help.

Kind regards

Selintra

[gippsweb]

Can you expand on this?

At the moment from home we have a Billion 7402vgp modem, it is configured as follows;
Line 1 Astratel
Line 2 My a@h box(now SME sail)
Gateway Sipbroker
Having sipbroker im my dial plan allows me to easily ring people who are with providers other than my own with ease. I was never able to configure it properly on my a@h box.

PS I'm also seeing the following errors from sail


May 22 13:20:55 WARNING[6823]: chan_zap.c:3931 zt_handle_event: Didn't finish Caller-ID spill.  Cancelling.
May 22 13:32:29 WARNING[6953]: chan_zap.c:3931 zt_handle_event: Didn't finish Caller-ID spill.  Cancelling.
May 22 15:25:49 WARNING[8435]: chan_zap.c:3931 zt_handle_event: Didn't finish Caller-ID spill.  Cancelling.
May 22 15:26:00 WARNING[3544]: chan_zap.c:1583 zt_set_hook: zt hook failed: Device or resource busy

[SARK devs]

HI

No problem. I've just had a look at Sipbroker and it looks fine.  I'm on the road today and tomorrow but I'll ask soemone to have a play with it back at the ranch.

May 22 13:20:55 WARNING[6823]: chan_zap.c:3931 zt_handle_event: Didn't finish Caller-ID spill. Cancelling.

This is Asterisk dumping your CID info.  Usually it's because it doesn't like your ring tone (if you'll pardon the expression).  Which country are you in?  

Kind Regards

Selintra

[gippsweb]

We're in australia, I've had the card working since the day they were a-ticked in a box with a@h 1.3 and it hasn't missed a beat.
I've only installed sail onto our upgraded sme box this morning, so I am expecting a few teething problems....

[SARK devs]

Doh! - says Aus next to your userid - sorry.

Do you have Aus  (au) set in Globals?

Check that zaptel contains

Code: [Select]


loadzone = au
defaultzone = au




Selintra

[gippsweb]

Yep, I definately have them both in zaptel.conf

I rebooted to be sure and it seems to have loaded in aus mode now.

Although I don't seem to be getting anything in stats although it is set to on in global.
And we are getting the following;

indications.c.505 ast_unregister_indication_country: Removed default indication country 'au'
setup_zap: Ignoring caller_id
setup_zap: Ignoring signalling

[SARK devs]

I rebooted to be sure and it seems to have loaded in aus mode now.

What brings you to this conclusion? What do you see when you ztcfg -vv and what is in /proc/zaptel/1

Looking at the BB's for the TDM400, the advice they give for Australian operation is to add the following to /etc/modprobe.conf...

Find the line

Code: [Select]


install wctdm /sbin/modprobe --ignore-install wctdm && /sbin/ztcfg


...and change it to

Code: [Select]


install wctdm /sbin/modprobe --ignore-install wctdm opermode=AUSTRALIA fxshonormode=TRUE boostringer=1 && /sbin/ztcfg


Apparently there are also hang-up detection issues with the TDM400 in Aus so you can add the following to zapata.conf.  

busydetect=yes
busycount=6

I'll try all this back at the ranch tomorrow to make sure it doesn't break anything.  


Kind Regards

jeff@selintra.com

[gippsweb]

ztcfg -vv gives me

Zaptel Configuration
===============

Channel Map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 04: FXO Kewlstart (Default) (Slaves: 04)

2 Channels configured.

/proc/zaptel/1   has

Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"

      1 WCTDM/0/0 FXOKS (In use)
      2 WCTDM/0/1
      3 WCTDM/0/2
      4 WCTDM/0/3 FXSKS (In use)

I have put that line into modprobe.conf, although apparently it is only opermode that is needed now as the rest is automatic as of kernel 2.6

I did notice that in a@h 1.3 in dmesg.log it use to show channel 1 and 4 in Australia mode, now it shows channel 1 in auto/dpo mode whatever that is.

I actually think it is hanging up the pstn line better now than it was on a@h with those lines in zapata.conf
Also it seems to be answering pstn calls before the first ring has finished, we need to get to the end of the first ring to get the caller id info.

[SARK devs]

Also it seems to be answering pstn calls before the first ring has finished, we need to get to the end of the first ring to get the caller id info.

From what I can gather, your caller-id info arrives betwen the first and second ring so I think you are right.  I'll do some more research today.  This is purely an asterisk issue so it's just a question of figuring out what we've missed in creating the rpms or missed from the config.

Best

[SARK devs]

Hi again,

We put up an "Aus" server to test some of this stuff.  It looks like we might be taking the call too early (our fault).  If we put up a fix do you think you could test it for us?  - the BT CID we get here arrives very early.

Kind Regards

[gippsweb]

No problems, I'm more than happy to do any testing I can to help 'the cause'.

[SARK devs]

Hi Again

Fetch up server-manager and click on Headers in SAIL PBX.  
Click the edit button next to  "zapata.conf".

Scroll down in the freeform window and add this to the bottom on a new line...

 

Code: [Select]

sendcalleridafter=2

You may also have to remove the "callprogress=yes" tuple.  It may be OK but it causes us problems here in UK when we are trying to detect CID.

Save your changes and then restart asterisk and give it a go.

Let us know how you get on.


Best

Selintra