Topic: [Announce] SAIL-2.1.11-142 Beta

[chris burnat]

Hello Jeff (and crew)

Do you mean that the call doesn't go through or that SAIL doesn't dial it without the prefix? Or... something different.  From the Asterisk logs it looks like the calls are completing.

Internal numbers fo Astratel are in the 8888nnnn range, i.e. 88884070 or whatever. To fudge the 02 for sydney local numbers, we have done:

Adding 02 for Sydney numbers (begin 8 or 9) In transform 9:029 8:028
What its doing is looking for numbers that begin 8 or 9 and appending 02 to them.

And so now, when I dial 88884070 on Astratel, the transform does it job and add 02.
Result, it goes noway...  My bad, I should have copied the rest of the logs:

-- Got SIP response 480 "Temporarily Unavailible" back from 210.8.40.188
    -- SIP/astratel1-8fe1 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- AGI Script selintra completed, returning 0
    -- Got SIP response 480 "Temporarily Unavailible" back from 210.8.40.188
    -- Executing Hangup("SIP/5001-2888", "") in new stack

Regards, chris

[chris burnat]

Heoo Tib, thanks for the info, good stuff.  Just one point:

I didn't put anything in the trunk ... This is my setup atm and works fine ...
Astratel Line Route = "Primary Voip Out"
_0[23478]XXXXXXXX _8888XXXX _0011+ZXX.

_0011+ZXX.  is not supported by Asterisk, I believe this format (+ sign) is specific to some distros, i.e. A&H.  Tried it, got noway..  
chris

[SARK devs]

re astratel->astratel

Ah - OK, now we understand.

Two ways forward.  

1.  Throw away our transform and use Tib's(JonB's) Custom App (which we know works fine).


or how about this for a little experiment...


2.  We think you can probably define the same astratel trunk twice (same account id and so on, but don't let it register - the prime copy will already have done that - it might not even be a problem but, you never know... ).  That way you can have one route/trunk sensitive to local Sydney calls (_[98][9-7]XXXXXX) and one route sensitive to astratel-astratel calls. The astratel-astratel route (_8888XXXX)  would use the copy trunk that didn't have the transform  (not sure that made sense reading it back).  Drop us an e-mail if it didn't and we'll send some screen shots through of what we have in mind.


Best

Selintra

[Tib]

Thanks Chris,

Thats one I haven't tried yet ... international calls I will try some other time ... we don't tend to do a lot of them.

Local and interstate though we do every day now and all work fine ... I'm very happy with the setup ... now that all is stable and runing I might even go and get a TDM400 card.

selintra ... what softphone do you recomend for external extentions ... I tried cubix but it seems to crash all the time. Haven't tried it without the USB phone though ... I don't have a mike at work.
 
I can get X-Lite to login but it won't put through any calls ... there must be a way  I just can't figure it out.

Regards,

Tib.

[JonB]

Chris has another option. Move out of Sydney  



Tib, I use DIAX as a softphone for remote or road warrior type applications. Being an IAX softphone there are no firewall/RTP issues.

I found another advantage to it the other day as well. It can be installed on a USB thumb drive and taken anywhere and so long as you have headset with you and access to a PC with a broadband connection you can make calls.

Jon

[chris burnat]

"Chris has another option. Move out of Sydney"
Very tempting Jon...

Just posting a fix provided by Selintra overnight.  I had created a Route named 1100&1200 and of course it did not work, the ampersand is not a valid character - next time, I will read the documentation. ..  The way out of it is simple, delete (as root) the relevant line in:

Code: [Select]

/home/e-smith/db/selintra
Simple.  Thanks to Selintra for support.
chris

[SARK devs]

Hi all,

Just to add to Chris's last msg, because the selintra DB is a stock e-smith item,  you can mess about with it direct from the console just like you can any other e-smith database.  Just make your changes and signal-event conf-asterisk to re-gen the asterisk .conf files.  All the normal DB commands should work just fine.

Code: [Select]


    /sbin/e-smith/db dbfile keys
    /sbin/e-smith/db dbfile print [key]
    /sbin/e-smith/db dbfile show [key]
    /sbin/e-smith/db dbfile get key
    /sbin/e-smith/db dbfile set key type [prop1 val1] [prop2 val2] ...
    /sbin/e-smith/db dbfile setdefault key type [prop1 val1] [prop2 val2] ...
    /sbin/e-smith/db dbfile delete key
    /sbin/e-smith/db dbfile printtype [key]
    /sbin/e-smith/db dbfile gettype key
    /sbin/e-smith/db dbfile settype key type
    /sbin/e-smith/db dbfile printprop key [prop1] [prop2] [prop3] ...
    /sbin/e-smith/db dbfile getprop key prop
    /sbin/e-smith/db dbfile setprop key prop1 val1 [prop2 val2] [prop3 val3] ...
    /sbin/e-smith/db dbfile delprop key prop1 [prop2] [prop3] ...


So if you do get a rogue row, you can always do a delete on it.  We'll put it on the list to publish the structure but in general very few rows are interdependent.  The ONLY departure from standard e-smith practice is in the AGI.  Because we can't afford to run any Perl code under Asterisk (you can do it, but it barks), we have our own DB parser/reader implemented in C.


Go on....  get hacking.

Best

Selintra

[SARK devs]

re astratel->astratel and 02 Sydney prefix

Here is a transform which works for all cases (thanks to Sam for thinking it up and Chris for testing it in anger).

9:029 8:028 028888:8888

That's it.


 

Kind Regards

Selintra

[psoren]

Hi,

Has anyone succeeded on getting the SPA2000 to call out?
I can recieve calls with no problems, sound both ways.

I have installed:
asterisk-SME7386-1.2.3-100.i386.rpm
selintra-sail-2.1.11-162.noarch.rpm

It's running on a VIA mini ITX PD1000 (Which is why i use the i386 rpm)

I'm not sure if it is the SPA or my carrier setup which is the problem
I can dial 0 and get the operator extension, but everything else gives me busy tone.

I have tried working on the dialplans, but no luck and i don't know much about that at all...

My carrier is the danish Musimi.

I would apreciate a push in the right direction if possible

[SARK devs]

Per,

Can you locate the dial plan in the spa2000 and show it here please?

btw - 686 runs fine on VIA PD10000 - we have several here - they use the Nehemiah chip which is true 686.

Selintra

[psoren]

Per,

Can you locate the dial plan in the spa2000 and show it here please?

btw - 686 runs fine on VIA PD10000 - we have several here - they use the Nehemiah chip which is true 686.

Selintra

   You are rigth..... it does.. DUH!! i always took this to be i585. I also have an older one which i use to test with, A@H for example..... That's i585 bacause i always have to recompile to make it run. I really feel stupid!!!
Well that's one thing less to worry about then

For the dial plans on SPA i have tried this:

(*x.|*xx*|x.)

And this:

(xxxxxxxx|112|00x.|*31*xxxxxxxx|*31*00x.|*xx)

And BLANK

Blank is the worst, the other two allows me to get the operator extension.

Am i missing a route out somehow?

Per

[psoren]

Hi again,

I have just set up a soft phone - X-lite.
It's the same problem. But i can make calls between the two.

Per

[SARK devs]

Hi per

Usually we recommend (*x.|*xx*.|x.) so that is the one you should use.

Have you set up a route in SAIL to send the outbound call to the trunk?  What does the route dialplan look like?  

What do you see on the asterisk console when you attempt to dial out?

Extensions don't need a route, they are set up automatically so this is why they can call each other.


Also,  is musimi a VOIP carrier or a regulat phone company?

Selintra

[psoren]

Hi per

Usually we recommend (*x.|*xx*.|x.) so that is the one you should use.

Have you set up a route in SAIL to send the outbound call to the trunk?  What does the route dialplan look like?  

What do you see on the asterisk console when you attempt to dial out?

Extensions don't need a route, they are set up automatically so this is why they can call each other.


Also,  is musimi a VOIP carrier or a regulat phone company?

Selintra

Hi Selintra,

Musimi is a VOIP carrier, it is one of the pioneers in denmark and they want to provide VOIP for "the people". So they are cheap, but all support is in the musimi forum and done by "the people". They have good howto's for Sipura and other good hardware but when using * and so, one have to relie on the forums. I think only a few actually uses SME servers if any.
I have it running on a A@H server and i remember having trouble with that as well, the first versions didn't even want to register. But now i want to use SAIL and then only have one server running to cut down on the power bill.
I have copied the trunk settings and reg. string from that A@H to SAIL

The route dial plan looks like this:

_XXXXXXXX _00XXXXXXXX_112

And the Asterisk output when trying to call is this:

Connected to Asterisk 1.2.3 currently running on perserver (pid = 4592)
Verbosity is at least 5
-- Executing AGI("SIP/5000-7bb8", "selintra|OutRoute|musimi-out") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
-- AGI Script Executing Application: (SetCallerID) Options: (46928840)
-- AGI Script Executing Application: (Dial) Options: (SIP/61658710@46928840)
-- Called 61658710@46928840
-- SIP/46928840-a49f is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- AGI Script selintra completed, returning 0
-- Timeout on SIP/5000-7bb8
== CDR updated on SIP/5000-7bb8
-- Executing Busy("SIP/5000-7bb8", "") in new stack
== Spawn extension (internal, t, 1) exited non-zero on 'SIP/5000-7bb8'
-- Executing Hangup("SIP/5000-7bb8", "") in new stack
== Spawn extension (internal, h, 1) exited non-zero on 'SIP/5000-7bb8'

I apreciate your help.

Per

[psoren]

IT'S WORKING !!!!!

I found the problem..

DID Number
SIP/IAX User

     Has to be the phonenumber to registrer.

SIP/IAX
Peer      

      Was also the phonenumber but that has to be something else like "musimi-in" or else it will conflict somehow.

Thank's for the help to make me see in the right direction.

Per