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[Announce] SAIL-2.1.11-142 Beta

Offline SARK devs

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[Announce] SAIL-2.1.11-142 Beta
« on: March 24, 2006, 05:34:37 PM »
Hi Everyone,

SAIL-2.1.11-142 is now available for download from our ftp site at;

http://selintra.com/docs/cgi-bin/view/Main/DownLoadPages

This is a biggish release with a lot of new functionality so you will almost certainly find bugs in it.

Highlights include;
    Full, Multi-Level IVR
    Custom Applications
    Fully automatic TDM/X100 board detection/configuration
    chan-spy implementation
    Message management panels
    Updates to the templates for phones and carriers
    Simplified Sipura ATA3000 support
    General panel cleanup and bugfix


This release has been tested with the rc1 releases of asterisk (asterisk-SME7rc1), we have not tested it with any earlier releases so if you are running SME7 pre rc1 you may want to wait.  There's no particular reason why it shouldn't run but just be aware that we haven't done any testing here.

Next out will include ISDN BRI support, probably EICON DIVA Server as a minimum.

Kind Regards

Selintra

poe

Asterisk
« Reply #1 on: March 24, 2006, 11:08:16 PM »
Just wantet to say youre doing a great job on with SAIL.

I have only one small problem that has to do with asterisk itself.

Somewhere along the way (updates/reinstalls) asterisk stopped starting at boot time.

Could you tell me the best way to make it start at boot time, while maintaining compability with the asterisk rpm you use?

If you should have some spare time one day - a description of best practice when upgrading SAIL would be nice. My server complains about the existing database when doing a rpm -Uvh.

Best regards, and thanks again.

Preben

Offline SARK devs

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« Reply #2 on: March 25, 2006, 02:01:54 PM »
Quote
Somewhere along the way (updates/reinstalls) asterisk stopped starting at boot time.


Hmmm sounds like you've lost a link.  We've seen this happen a couple of times but haven't manage to track down the reason why.

Do
Code: [Select]

ls  /etc/rc.d/rc7.d

and look for S93asterisk.

I suspect its not there.

If it isn't, do
Code: [Select]

cd /etc/rc.d/rc7.d
ln -s /etc/rc.d/init.d/e-smith-service S93asterisk


That will cure your start-up problem.

Quote
My server complains about the existing database when doing a rpm -Uvh.


Don't worry about this, it's benign.  We just weren't smart enough to figure out whether the database had been created or not in the rpm.  I think they fixed this in the later releases but I'm not 100% sure.  Either way, it's not a problem.


Regards

Selintra

Offline xboxer21

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Greetings Upload
« Reply #3 on: March 25, 2006, 04:30:28 PM »
How do we upload a greeting?

Thanks
......

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[Announce] SAIL-2.1.11-142 Beta
« Reply #4 on: March 25, 2006, 04:55:12 PM »
Quote
How do we upload a greeting?


Go to any phone and dial *60*nnnn (where nnnn is the number of the greeting you want to record).  Follow the voice instructions.  You'll have to input a password - it's the "4-digit Password for KEY Ops:" in Globals panel, default is 1111.

Once saved it will appear in the greetings panel and you can optionally give it a description.

If you have a sound recording you've made externally then you can save it directly in

/var/lib/asterisk/sounds

as a .gsm file with the name

/var/lib/asterisk/sounds/usergreetingnnnn.gsm

where nnnn is the greeting number.


For a complete list of keypad operations see

http://selintra.com/docs/cgi-bin/view/Main/SysKey

Kind Regards

Selintra

Offline xboxer21

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[Announce] SAIL-2.1.11-142 Beta
« Reply #5 on: March 25, 2006, 06:21:32 PM »
Thanks Selintra
......

dswillia

[Announce] SAIL-2.1.11-142 Beta
« Reply #6 on: March 26, 2006, 04:38:56 AM »
Excellent Job guys, this last beta actually made me dump AMP.  I do have a couple questions though.

It seems as though calls are automagily answered and put into a queue and then ring to the operator extension.  I have 3 GXP 2000's I would like to all ring at the same time, no operator.  Is there a way to do this without having to create a queue and logging in all the users from the phones?

Also is there a way to pass caller id to the incomming calls, as of right now my gxp's are just showing the name of the pbx (account name)  This was working fine in AMP

Regards,

Offline SARK devs

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« Reply #7 on: March 26, 2006, 08:45:03 AM »
Hello ds - glad you like what we're doing.

Quote
I have 3 GXP 2000's I would like to all ring at the same time,


Currently, this isn't available. Our view (perhaps wrongly) is that group-pickup and CFBS do this better than group ring, which can be distracting in an office environment.  Here's how it works;

Make sure all three phones are on the same pickup group (set in extensions panel). Put the phones into a CFBS loop (i.e. 1->2->3->1) using *22*. You shouldn't do this with most asterisk implementations because they've got no loop/spiral detection, however it's perfectly safe with SAIL.  Point your inbound trunk at any one of the 3 phones.  An inbound call will now always ring at least one phone (uness all three are busy, in which case, it will drop to voicemail) .  The call can be picked up from any phone using group pickup (*8#).  If this is not practical for you (perhaps the phones are not within earshot of one another), then we will put ring groups in for you.  It's not difficult to do, we just haven't done it yet.


Quote
Also is there a way to pass caller id to the incomming calls


Can we ask a few questions to understand this better?  
    Can you explain what you mean by "name of the pbx (account name)"?

    What exactly are you seeing on the GXP2K screen?

    Where are the calls coming from?  Are they VOIP or analogue, and if analogue, how are you interfacing to the analogue line (TDM board, Spa3K, whatever).


Thanks in advance.

Selintra

Offline Tib

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« Reply #8 on: March 26, 2006, 02:43:20 PM »
ok

I think I need a bit more help here ... I recorded a greeting but have no idea how to get the system to play my greeting and not the standard one.

I have my trunk lines setup ... Sipura 3000 and Astratel

I have Extentions all setup ... spa1 on 5000 and two X-Lite 5001 and 5002

I have 2 routes setup .... one to go through the sipura3000 and one to go through Astratel.

I have a heap of speed dial numbers

I have setup an Agent ... not being used atm

Also have one queue called general ... not being used atm

I have recorded a Greeting that I called ...Standard greeting

To get this greeting working do I setup IVR menus?

I have no idea what way to go atm.

Maybe a flow chart of what to setup for where and what ties in with what could be good for ppl like me :)

Regards,

Tib

dswillia

[Announce] SAIL-2.1.11-142 Beta
« Reply #9 on: March 26, 2006, 05:50:25 PM »
Thanks for the response, my problem is that I run this in my home, and the phones are not withing earshot, ring groups would be nice, however queues seem to work fine so don't spend any "development" time on my account.

Can you explain what you mean by "name of the pbx (account name)"?

It just says Incomming call, rings 2 lines and says "asterisk asterisk" on the CID.

I am accessing the TDM lines via a X100P "clone" card, as well as a Voip Circuit from MixNetworks.  Neither display CiD info.

Also, generaly when running amp/asterisk on the server I had to open ports 5060 and 10000 to (Whatever) UDP for my system to get incomming voip calls.  Does SAIL do this for me, or do I still need to open them?

Regards,
dswillia

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« Reply #10 on: March 26, 2006, 05:55:33 PM »
Hi Tib,

Sounds like you've been busy.  You don't have much more to do to activate your greeting.  Go to the IVR panel and create a new menu.  Name it and in the greeting drop down you will see all of your user-recorded greetings.  Choose the one you want to use for this menu.  We assume your greeting asks the caller to make a choice of some kind based around hitting keys on the phone.  In the menu you can specify the required  actions against the various keys.  The drop downs contain all of the extensions on your system together with all of the "action" entities you've defined.  An "action" entity can be a menu, a custom app or a queue.   Against the keys you've referenced in your greeting choose an action or an extension to call.  In the "Timeout" box you also choose an action or extension  (which will occur if the caller does nothing).  

To publish your greeting choose the trunk for which you want it to be active and press the change (CHG) button.  On the change panel, turn IVR on (it's a check box) and choose your greeting number from the "open Greeting number" drop down.

That's it you're done.  Call the trunk and test your greeting.

Regards

Selintra

Offline xboxer21

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Working config for teliax/bv
« Reply #11 on: March 27, 2006, 06:05:02 AM »
I'm having trouble configuring SAIL to work with Teliax and broadvoice.
If anybody here has sucessfully configured sail please post the how to.

Thanks
......

Offline SARK devs

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[Announce] SAIL-2.1.11-142 Beta
« Reply #12 on: March 27, 2006, 08:37:22 AM »
Quote
Also, generaly when running amp/asterisk on the server I had to open ports 5060 and 10000 to (Whatever) UDP for my system to get incomming voip calls. Does SAIL do this for me, or do I still need to open them?



See

http://selintra.com/docs/cgi-bin/view/Main/AstUdpPorts#SME_Server_Version_7_0_and_UDP_P

for a discussion of UDP port issues.


Regards

Selintra

Offline Tib

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« Reply #13 on: March 27, 2006, 09:44:11 AM »
Thank you Selintra,

That worked nice ... one thing though once the extention rings out it goes back to the standard asterisk message ... that would be ok if it actually said something about leaving a message but it doesn't.

Is there a way to over write that message or put a different message in place of it.

Also how do I get the system to ring a few more times then what it does ... it's a bit too short if your not near the phone.

Regards,

Tib

Offline SARK devs

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« Reply #14 on: March 27, 2006, 10:18:53 AM »
Quote
how do I get the system to ring a few more times then what it does


Hi Tib,

It's set in Globals (Ringdelay) - see

http://selintra.com/docs/cgi-bin/view/Main/SysGlobals2

Quote
Is there a way to over write that (voicemail) message or put a different message in place of it.


Yes, it's actually pretty sophisticated.  You can read a full discussion here

http://www.asteriskguru.com/tutorials/asterisk_voicemail.html

To record your busy and unavailable greetings go into voicemail (*50*) then press 0 (mailbox options) followed by 1 to record your unavailable greeting or 2 to record your busy greeting.  


Kind Regards

Selintra

Offline Tib

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« Reply #15 on: March 27, 2006, 02:00:34 PM »
hello again

At the moment I have ...

Extention: 5000 ... spa1 = sipura3000
Extention: 5001 ... my computer = X-Lite
Extention: 5002 ... wife's computer = X-Lite
IVR Menu: Please pres 1 for so on and so forth

Call comes in ... goes to menu ... person presses 1 no one answers ... receptionist comes on and says ... sorry the person at extention 5001 is not available and there is silence ... then all of a sudden says thankyou.

The caller not knowing what to do just hangs up and wonders what that was all about.

Would make more sence if the receptionist said ... Sorry the person at extention 5001 is not available please leave a message ... then thankyou.

Now what I wouls like the system to do is ...

Call comes in ... menu (pre recorded message says what to do) .... this bit works.
Person decides to go for option 1 and call my computer .... I do not pickup but I am at home ... just not near computer ... this is where I would like the system to go back to a menu (new one maybe) and let the person select another extention or the system puts them in a queue and start going through the other extentions ...
If no one picks up then let the person have a choice to hang up or leave a message.

What would be the best way to execute this.

Regards,

Tib

Offline JonB

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« Reply #16 on: March 27, 2006, 02:43:58 PM »
Tib,

A couple of things.

The default voicemail has a tone at the end of the message however I suggest that you record your own greeting and say 'please leave a message after the tone'

You can create multi level IVR's but to do what you want you will have to create a custom app. Check out the asterisk documentation.

You can also change the SPA3k so that it answers the incoming call after so many sec's. This is how I have my SPA3k set up at home.

The home phones ring for 10 sec's so I can pick up the calls. If it is not answered after 10 sec then the SPA3k picks up the call and passes it to the 1st IVR message.

"we are unable to take your call......"
"press 1 to leave a message for Jon" (goes to my voicemail and email)
"press 2 to leave a message for Sue" (goes to Sues voicemail and email)
"press 3 to contact Computer Troubleshooters - Howick" (goes to a second IVR message)"

second message

"You have reached Computer Troubleshooters....."
"to contact the office press 1" (dials an office extension via a Sail2Sail trunk)
"to contact Jon by mobile press 2" (dials my mobile via a voip provider trunk)

Jon
...

Offline JonB

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« Reply #17 on: March 27, 2006, 03:42:50 PM »
Here are a couple of custom apps may you find useful in setting up an IVR.

First is to send an incoming caller directly to a mailbox.

Code: [Select]
[custom-vmxxxx]

exten => s,1,Voicemail,uxxxx
exten => s,2,Hangup()



where xxxx is the extension voicemail you want the call to go to

Second is giving an incoming caller access to DISA. DISA is internal dial tone. I make this a hidden option so that if I am away from the office at say a client site I can dial in on my PSTN (local calls in NZ are free), press the unannounced option for DISA, enter the password and get local dialtone from the office.

Code: [Select]
[custom-disa]

exten => s,1,Answer
exten => s,2,DigitTimeout(5)
exten => s,3,ResponseTimeout(10)
exten => s,4,Authenticate(****) ; replace **** with a numerical password
exten => s,5,DISA(no-password|from-internal)


The third is an example of dialling a number via a trunk

In this example I am calling my NZ mobile via Voipjet in the USA.

Code: [Select]
[custom-mobile]

exten => s,1,Dial(IAX2/user@voipjet/011642XXXXXXX,45,r)
exten => s,2,Hangup


011 is the number I use to access the Voipjet route, 64 is NZ country code, 2 is mobile code.

Jon
...

dswillia

Caller ID
« Reply #18 on: March 27, 2006, 06:04:47 PM »
Some more information about my caller ID issue, in the Asterisk CLI I get the following when doing a reload:

Mar 27 10:02:38 WARNING[9769]: chan_zap.c:10829 setup_zap: Ignoring caller_id
Mar 27 10:02:38 WARNING[9769]: chan_zap.c:10829 setup_zap: Ignoring signalling


Also, it may be interesting to note, when I reboot my server asterisk dies on the initial reboot, I have to do a selautosniff and then asterisk start to get the system back up and running

Offline SARK devs

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« Reply #19 on: March 27, 2006, 06:56:55 PM »
Hi DS

Thanks for this extra info, we're still looking at this.  It's an asterisk config problem of some description.  
Questions
1. Is CLID enabled on your analogue lines?  
2. Which country are you in and who is your telco carrier?

Thanks

dswillia

[Announce] SAIL-2.1.11-142 Beta
« Reply #20 on: March 27, 2006, 08:48:23 PM »
Yes, CID is enabled on those lines, as well as my VoIP circuit from MixNetworks, I am in the US and Southwestern Bell/ATT is my provider.

One other note pertaining to the GCP2000's:

The GXP2000 allows for BLF's (Busy Lamp Fields) so I can see if one of the extensions is currently on the phone.  Asterisk allows these to work via the "hint" command in the extensions.conf dialplan.  I was able to create a custom app named "Hint" for internal and put the following in:

exten => 5000,hint,SIP/5000
exten => 5001,hint,SIP/5001

This allows me to monitor the lines as well as one button speed dial the extension.  Just a FYI

Regards

Offline Tib

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« Reply #21 on: March 28, 2006, 02:34:23 AM »
Thanks JonB

I'll try this as soon as I get home.

I'm loving all this ... mind you I really don't need all this just for a home setup  but it's good to know just incase.

Looks like I need to do a bit more reading and searching ... asterisk is so powerfull.

Regards,

Tib

dswillia

[Announce] SAIL-2.1.11-142 Beta
« Reply #22 on: March 28, 2006, 07:25:05 PM »
Trying to setup a remote phone to my system, I have opened the correct IAX port and the softphone works fine internally.  I get the following error in the asterisk CLI:

AGI Script Executing Application: (Dial) Options: (IAX2/5004|30|twW)
Mar 28 11:21:18 NOTICE[3717]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)

Do I need to open a udp port on the other side?  Interesting to note, it shows the phone is registered correctly in both asterisk and on the remote phone.

Regards

Offline SARK devs

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« Reply #23 on: March 28, 2006, 07:49:16 PM »
HI

First of all, can we just say how astonished we were to release custom apps on friday evening and have Jon and DS put up working examples by monday!  You guys are amazing. :-)

Other stuff (in no particular order).

Bugs..  

Quote
Asterisk fails to start after reboot


and

Quote
FOP only displays one conference room


We've created a "known-issues" section on the wiki and you can get fixes for both these issues there.

UDP Ports; - we aren't opening 5060 and  4569 correctly (thanks JonB).  We'll put a fix up onto the known-issues pages shortly.  

UDP 10000:20000, you should ONLY need to open these if you are running server-only behind a firewall.   You shouldn't need to open them in server-gateway mode unless you want to support a remote IP phone.  If you want to know why see

 http://selintra.com/docs/cgi-bin/view/Main/AstUdpPorts

CID not detected (reported by DS).  

We've been playing with this today DS.  We detect VOIP CID just fine on our SME7rc1-143 system here and it displays properly on all our test phones (gs101/102, GXP2000, spa-841, SJPhone, X-lite et al) .  

We don't detect analogue CID but you shouldn't read too much into this, BT CID is transmitted quite differently to the Bellcore CID used in the US and we have a few challenges with it at the moment.  As far as we know, we aren't doing anything to stop Bellcore CID recognition in our setup but we can't be sure.  You might want to check our generated zapata code with that of A@H (which you said worked OK).   Other than that you need to examine ${CALLERID} on inbound to see what, if anything, is in there.  

Sorry we can't be more forthcoming at the moment.

Thanks for your continued interest and support.  We really do appreciate it.

dswillia

[Announce] SAIL-2.1.11-142 Beta
« Reply #24 on: March 29, 2006, 03:10:14 AM »
Caller ID Issue Solved:

Ok looks like there is a misconfiguration in the zapata.conf file for my installastion anyways.  You have the line:

usecallerid=us

I assume this is automagicly set by setting in globals.  Anyways I changed it to:

usecallerid=yes

and it all works.

Regards

Offline SARK devs

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« Reply #25 on: March 29, 2006, 02:22:28 PM »
Quote
Ok looks like there is a misconfiguration in the zapata.conf file


Hi DS,

Thanks for spotting this.  We've posted a fix on the wiki at

http://selintra.com/docs/cgi-bin/view/Main/SysV21-143-SysZtl


Thanks again for all your help.


Selintra

Offline ldkeen

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« Reply #26 on: April 01, 2006, 08:48:43 AM »
Hi dswillia,
Thanks for that awesome tip with the blf's on the GXP2000. Finally managed to get around to setting it up. I had to upgrade the firmware in the phone though - but now I can't seem to get account2 to register?? Prior to the firmware upgrade I had both account1 and account2 registered as 5000 and 5003 respectively but now 5003 wont register. I notice that account2 uses udp5062. have you come across this problem??
Tib
I saw in a previous post by yourself you explained how to setup a transform mask for astratel (<:07>[35]xxxxxxx). I have an astratel trunk setup and I would like all my local calls to go through this trunk. Astratel require users to prefix all numbers with the area code. What I would like is for all 8 digit numbers dialed to be prefixed with 02 and routed through the astratel trunk. Is this correct <:02>xxxxxxxx and do I put it in the route-create panel or the trunk-change-transform mask panel?
Thanks everyone for the help and thanks JonB for the great custom apps
Lloyd

Offline JonB

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« Reply #27 on: April 01, 2006, 02:23:57 PM »
Lloyd,

Add port=5062 to the configuration for extension 5003.

Jon
...

Offline SARK devs

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« Reply #28 on: April 01, 2006, 03:10:44 PM »
Quote
What I would like is for all 8 digit numbers dialed to be prefixed with 02 and routed through the astratel trunk.


HI Lloyd,

In theory you should just be able to code a transform of

Code: [Select]

:02


However, running through the code in my mind (I don't have it here) I've got a horrible feeling it won't work.:-x  

Of course if you also have other number sequences which need to go through the same trunk, then this won't work anyway.   One trick is to declare the same trunk twice with different names so you can route different number sequences through different transforms yet still punch them out through the same carrier.

Give it a try and let us know how you get on.  If it's broke, we'll fix it.  

Kind Regards

Selintra

Offline ldkeen

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« Reply #29 on: April 01, 2006, 10:14:03 PM »
Selintra,
<In theory you should just be able to code a transform>
I tried them all. I tried <:02>, :02, space:02 I tried putting the transform into the trunk-change panel as well as the route-create panel (by itself and combined with dial plan). I thought maybe Tib might have had a trick to get it to work.
<Of course if you also have other number sequences>
These 02 numbers probably account for well over 90% of the calls. I'm not to worried about others at the moment.
Thanks Lloyd

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« Reply #30 on: April 01, 2006, 10:32:28 PM »
Quote
These 02 numbers probably account for well over 90% of the calls.


No problem Lloyd, we'll have a look at it for you.  Sorry you've had to mess about with this mate. We know from the work we did with Stephen and Chris that these two-digit short codes are important in Aus.   We've got a maintenance/bug-fix minor release due for next week.  We'll try to get it into that for you.  :-)

Kind Regards

Selintra

Offline Tib

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« Reply #31 on: April 01, 2006, 11:46:42 PM »
ldkeen

That setup only works for the SPA3000 ... I haven't got it to work for the asterisk setup ... I've had a look at a few different plans for asterisk but I can't get any thing like that to work.

At the moment I have this setup ... prob can be simplified but it works ...
Astratel Line  Rout = "Primary Voip Out"
_0[23478]XXXXXXXX _8888XXXX _0011+ZXX.

PSTN Line ... Rout = "PSTN Rout"
_000 _13XXXX _1300XXXXXX _1800XXXXXX _190XXXXXXX _1[12]XX

There is supposed to be something similar but it doesn't work either ... would be nice though it's a pain to dial 07 or in your case 02 for local calls all the time when your used to just dialing the number without the area code.

I will keep searching though ... there has to be a way.

Regards,

Tib.

Offline Tib

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« Reply #32 on: April 01, 2006, 11:55:15 PM »
Hell again,

Here are some helpfull guides for plans

The various patterns you can enter are similar to Asterisk's definition of them:

·         X — Refers to any digit between 0 and 9

·         N — Refers to any digit between 2 and 9

·         Z — Any digit that is not zero. (E.g. 1 to 9)

·         [Various] — Match only one character that matches any of the one in the square brackets. (E.g. [02-68*#] would match 0, any number between 2 and 6 inclusive, 8, * and #. Or, another way of saying this would be 'Match * or #, or a number that isn't 1,7 or 9') –Avoid using this notation in your Trunk Dial Rule as it does not seem to work. Use this only in your Outgoing Route Dial Pattern.

·         . — Wildcard. Match any number of anything. Must match *something*.

·         | — This lets you use a '0 to dial out' (or '9' in the US) by matching anything before the line, but not sending it to the trunk.

Go here and check out some explanations etc ...

http://members.optusnet.com.au/~bsharif/asterisk/AsteriskDumbMeGuide.htm#_Trunks_and_Outbound

Regards,

Tib

Offline Tib

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« Reply #33 on: April 02, 2006, 12:46:25 AM »
ok

I tried trunk :07 as well as just 07 and route _3XXXXXXX then dial a local number ... no go

I do 0:07 and in route _0XXXXXXXX then dial local number with a 0 infront of it eg:  0337XXXXX .... XXXXX being numbers ... it works

I see salintra is looking into this ....

So close but not quite there yet ... is this a bug then?

Regards,

Tib

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« Reply #34 on: April 02, 2006, 03:44:58 AM »
Tib,

The transform mask is working exactly as it should.

0:07 replaces the first 0 with 07.

I believe the object of the exercise is to dial an 8 digit local number without having to dial any extra digits and have the area code prepended to the number.


There are a couple of ways to do it.

Jeff needs to change the selintra agi so that if you use 02+ as the mask it prepends 02 to the dialled digits.

The other option is to create a custom app.

First you need to create a route that will use the provider you are using for your local service. In the example below I have named this route local. Make it internal and you do not need to create a dial plan.

Next create a custom app [custom-local]

Code: [Select]
exten => _NXXXXXXX,1,Goto(02${EXTEN},1)
exten => _02NXXXXXXX,1,agi(selintra,OutRoute,local)


where _NXXXXXXX is the number of digits to dial a local number,02 is the area code and local is the name of the route you set up.

Save it and it should all work. Just dial your local number.

Jon
...

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« Reply #35 on: April 02, 2006, 05:09:19 AM »
JonB

That worked perfectly .... thanks.

When I was trying all the different ways I was just trying it with 0 infront for testing purposes ... that wasn't that way I wanted to call local numbers.

I was still using the full 10 digit to call even local till now.

Thanks again.

Tib.

Offline ldkeen

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« Reply #36 on: April 02, 2006, 05:16:27 AM »
<That worked perfectly .... thanks.>

Sounds good. I'll try it when I get home from work tonight. Thanks for the tip JonB
Lloyd

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« Reply #37 on: April 02, 2006, 08:40:02 AM »
Quote
Next create a custom app [custom-local]



Respect Jon.  We'll try to fix the transform this week, but in the meantime, this is a very nice solution.  It's also a pretty cool example of how to use Routes from a custom app.  We'll also try to find time to properly document the agi.  Sam (Mr agi), is snowboarding in Switzerland at the moment, so it will have to wait until he gets back.

Kind Regards

Jeff

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« Reply #38 on: April 02, 2006, 01:25:34 PM »
Fantastic package!
Experiencing a small bit of bother with Engin - unmonitored status in peers:
Name/username              Host            Dyn Nat ACL Port     Status
astratel1/88884121         210.8.40.188                5060     OK (61 ms)
koala1/71783               203.122.248.173             5060     OK (88 ms)
engin1/0282124358          202.61.12.166               5060     Unmonitored <<<----HERE
5000/5000                  192.168.0.21     D          5060     OK (151 ms)

What have I done wrong?  And does it matter?  Any assistance would be appreciated.
I am attaching below relevant portions of sip.conf

[engin1]
type=peer
host=syd.byo.engin.com.au
insecure=very
auth=md5    
qualify=no
canreinvite=no
username=0282124358
fromuser=0282124358
secret=XXXXX
disallow=all
allow=g729
allow=gsm
allow=alaw
allow=ulaw

Edited:  Found the problem,
change qualify=no to qualify=3000

Name/username              Host            Dyn Nat ACL Port     Status
engin1/0282124358          202.61.12.166               5060     OK (158 ms)

OK now.  Jeff, you may wish to pick this up in next maintenance upgrade.
chris
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« Reply #39 on: April 03, 2006, 05:37:57 PM »
Hi

Regarding Astratel and Sydney area prefixes .  This transform in the astratel local trunk will handle...

 In the Local Route include a dialplan of

 _[98]XXXXXXX

 In the Astratel trunk(s)

 set the transform field to

 9:029 8:028

 That's it,  now you need only dial the 7 figure number.  

 Thanks to Chris Burnat for staying up into the wee small hours testing this with us.


Regards

Selintra

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« Reply #40 on: April 04, 2006, 01:57:38 AM »
Code: [Select]
That's it, now you need only dial the 7 figure number.

The 8 figure number?
- chris
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« Reply #41 on: April 04, 2006, 04:09:51 AM »
Got myself into a couple of hitches, so I'll flag them:

1) Had allocated Trunk Preselects in the range 90 to 95, avoiding 91.  Then modified the system to dial AUS number using only the 8 digit method.  This morning, I tried to dial number 921844nn and ended up in all sorts of strife - Asterisk was dialing to my trunk #92 (Koala)...  Moved all Trunk Preselects in range 50-55, and now all is well.  

2) Just installed an SPA-841 on my SAIL box.  This phone has its own internal dialplan tailored for the US.  Resulted in some interesting moments. I could modify this dialplan for AUS conditions, but for the time being, given that Asterisk is dealing with dialplans, I have entered [*0-9]. in both extensions, and things are much better now (thank you Will!).

Question: Is there any benefit using dialplan in the Sipura when used as an extension on SAIL/Asterisk?
Thanks.
chris
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Offline chris burnat

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« Reply #42 on: April 04, 2006, 05:56:30 AM »
Code: [Select]
Regarding Astratel and Sydney area prefixes . This transform in the astratel local trunk will handle... In the Local Route include a dialplan of _[98]XXXXXXX
In the Astratel trunk(s) set the transform field to 9:029 8:028


Hmmm. Works like a charm except that after putting the transform into the Astratel Trunk, I cannot dial direct to another Astratel number.  Here how it goes;

dialing 88884070
-- Executing AGI("SIP/5001-ccbb", "selintra|OutRoute|ASTRATEL2") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (SetCallerID) Options: (88884121)
    -- AGI Script Executing Application: (Dial) Options: (SIP/0288884070@astratel1)
    -- Called 0288884070@astratel1

Note: ASTRATEL2 is a route for astratel to astratel: _8888XXXX
 
dialing 5088884070
-- Executing AGI("SIP/5001-997a", "selintra|OutTrunk|88884121") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (SetCallerID) Options: (88884121)
    -- AGI Script Executing Application: (Dial) Options: (SIP/0288884070@astratel1)
    -- Called 0288884070@astratel1

Note: Prefix 50 is the trunk preselect for Astratel
- chris
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« Reply #43 on: April 04, 2006, 07:01:00 AM »
burnat

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.

PSTN Line ... Route = "PSTN Rout"
_000 _13XXXX _1300XXXXXX _1800XXXXXX _190XXXXXXX _1[12]XX

Custom App=
[custom-local]
exten => _NXXXXXXX,1,Goto(07${EXTEN},1)
exten => _07NXXXXXXX,1,agi(selintra,OutRoute,local)

Now I can call numbers with 8 digits or 10 digits they all go no probs.

All my other settings are on default as in extentions .... 5000, 5001 etc

[edit]
I'm in QLD of course so you will have to change your to suit eg:
exten => _NXXXXXXX,1,Goto(08${EXTEN},1)
exten => _08NXXXXXXX,1,agi(selintra,OutRoute,local)
For WA, SA or NT
[Edit]

Regards,

Tib

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« Reply #44 on: April 04, 2006, 11:12:19 AM »
Quote
Hmmm. Works like a charm except that after putting the transform into the Astratel Trunk, I cannot dial direct to another Astratel number.


Hi Chris,

We've been sitting in the office looking at this mate and we've come to the conclusion that we don't understand  :-?

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.

Quote
Question: Is there any benefit using dialplan in the Sipura when used as an extension on SAIL/Asterisk?


Wow! That's a big question Chris.  Here's what we think...
Unless/untill we put automatic provisioning into SAIL (and we probably will at some point), we would recommend leaving the Sipura dial plans (on all their terminal devices; phones and ATA'a), set to "(*|*xx*.|x.)" which effectively means they allow anything through.   Having said that, SAIL is a very young platform and there may be cases where  the Sipura plan can compensate for a shortcoming within SAIL so it's nice to have (you could have used it to solve your Sydney "02" code problem).  However, most folk have mixed phone types (sipura and non-sipura) so it's better to have SAIL/Asterisk do the work if at all possible.  

Best

Selintra

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« Reply #45 on: April 04, 2006, 12:09:35 PM »
Hello Jeff (and crew)

Quote
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:
Quote

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:

Quote
-- 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
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« Reply #46 on: April 04, 2006, 12:35:14 PM »
Heoo Tib, thanks for the info, good stuff.  Just one point:

Quote
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
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« Reply #47 on: April 04, 2006, 12:39:01 PM »
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

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« Reply #48 on: April 04, 2006, 12:50:35 PM »
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.

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« Reply #49 on: April 04, 2006, 01:51:45 PM »
Chris has another option. Move out of Sydney  :lol:



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
...

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« Reply #50 on: April 04, 2006, 11:27:55 PM »
"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
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« Reply #51 on: April 05, 2006, 12:08:05 AM »
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

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« Reply #52 on: April 05, 2006, 04:13:43 PM »
Quote
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

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« Reply #53 on: April 05, 2006, 06:32:48 PM »
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

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« Reply #54 on: April 05, 2006, 07:09:59 PM »
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

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« Reply #55 on: April 05, 2006, 08:53:27 PM »
Quote from: "selintra"
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


 :oops:  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 :lol:

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

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Can't get SPA2000 to call out
« Reply #56 on: April 05, 2006, 09:20:41 PM »
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

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« Reply #57 on: April 05, 2006, 09:57:15 PM »
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

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« Reply #58 on: April 05, 2006, 11:04:37 PM »
Quote from: "selintra"
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

Offline psoren

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At last.....
« Reply #59 on: April 05, 2006, 11:38:22 PM »
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

kangkc

[Announce] SAIL-2.1.11-142 Beta
« Reply #60 on: April 06, 2006, 03:21:10 AM »
Nice piece of work there. Will consider getting commercial support if this proved to work for production.

However, coming from a AMP based Asterisk implementation, a partuclar feature which is the 'Ring Group' seems to be missing. Is there a way I can define ring group (or multiple extensions) such that I can get any incoming call to ring? i can do so by coding my own routing in config files but will prefer to do that via the interfaces.

Offline del

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[Announce] SAIL-2.1.11-142 Beta
« Reply #61 on: April 06, 2006, 04:44:27 AM »
Hi All,

I find this thread interesting, however this maybe a silly question but where can I find out what hardware is required to get this up and running?
Thanks in advance.
Del
If at first you don't succeed, then sky-diving is not for you!
"Life is like a coin. You can spend it anyway you wish, but you can only spend it once." --Author Unknown

Offline SARK devs

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[Announce] SAIL-2.1.11-142 Beta
« Reply #62 on: April 06, 2006, 05:56:00 AM »
Quote
a partuclar feature which is the 'Ring Group' seems to be missing.


Hi

Ring group is in the -161 release (on the ftp site now) - we had this feature very low on our priority list as a business feature.  - Shows how much we know, lots of people have asked for it!  

It's easy to use, you just create a speed-dial number but instead of pointing it at one extension you can code as many as you like separated by whiespace - like this ....  5003 5002 5004 5006

Kind Regards

Selintra

Offline SARK devs

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[Announce] SAIL-2.1.11-142 Beta
« Reply #63 on: April 06, 2006, 06:14:07 AM »
Quote
where can I find out what hardware is required to get this up and running?


Any spare i686 PC with 256Mb or more will do to get you started. If you just want to place VOIP calls, then the only other thing you will need is a softphone (X-lite and SJPhone are two of the best).

To interface to your landline you will need a special PCI Board (either an X100P - 1 line, or a TDM400P - up to 4 lines), available from Digium (www.digium.com) or you can use an FXO Analogue Gateway device like the Sipura 3000 (available from Linksys).  Finally, we are currently testing our own range of TDM boards which will hopefully be available in a month or so.


Kind Regards

Selintra