Don't have tittle

My new tittle: FATHER :D

2010-02-23T15:56:00.000+01:00

On 21 Feb 2010 01:41 my baby is born and weights 3270 grams and OMG how much i love him. This is really something amazing. Happy happy happy...

Restricting outbound calls in Asterisk/FreePBX

2009-03-21T00:16:00.001+01:00

One of the most required features of any PBX in any company is call restrictions. Now i know there are many written articles about this and many different ways to do it from easy to complex way but i find my method quite easy to implement and involves no manually editing any file.

Now all you need is Custom Context module (click for more information on how to download and install it in FreePBX)

Now lets start making some people angry with restrictions.

For the rest of the guide i will use my own setup as How-to and example.

First thing that you need is identify how many levels/groups of restriction do you need:

I have 4 groups of restriction:
1) Only extensions – a group which will not be able to make any outbound calls only internal ones
2) National/local calls – a group which will be able to use all outbound routes that can call any number within a country
3) Any call with PINs – a group which will be able to make any call anywhere but each user has its own PIN code (PIN code will only be used when making international calls)
4) No restrictions – a group without any kind of restriction, that includes no PINs as well (bosses really hate having restrictions pins, etc and it makes sense since they pay the bills).

Now we go and apply these restriction accordingly.

Lets click Custom Contexts after it appeared when you installed the module which we will see two textboxes which we will fill right away with our first context:

1) The first one is Only local extensions so i will put in Context:
local-extensions
and Description:
Only local extensions
(I have to admit I'm very bad at naming so you are free to use your own naming convention)
Now we click submit and make the proper restrictions

First we Set All To: Deny (we don't want to change all those list boxes one by one)
Then we allow only ones that we want to allow so this groups is able to make local/internal calls.
Basically what we need to allow are:
Call Parking, ext-group, ext-local and ext-queues, there are some others that we could allow like app-speakextennum, app-speakingclock, app-userlogonoff but that is up to you and it depends what other app/modules you are using and have setup, and you have to make sure that all outbound routes are set to DENY or you shouldn’t read this guide at all.

Submit

Now before i continue with the rest of the guide I ASSUME that you allready have setup trunks and outbound routes (we’ll cover these in future… maybe…)

2) Create context: national-calls with a description Allow national calls.
Or better way you can go to your created 1st local-extensions context and duplicate it since we will use the previous setup and just add more allows.

Here i have allowed everything in the two first sections (Default internal context and Internal Dialplan) except ENTIRE Basic Internal Dialplan and ALL OUTBOUND ROUTES (these two should never be allowed)

And in the Outbound Routes sections i have allowed only the routes that are able to make local and national calls (i don't know what kind of hardware you do have but for example i have FXO adapters, Sipuras, GSM gateways and local ISP SIP accounts)

Click submit

3) Duplicate the context national-calls and name it lets say:
all-calls-wpin (All calls with pin restriction)

Now i believe you can guess what's next is that you just have to add those outbound routes that are left which can make international calls (in my case i have many different SIP providers that can make international landline/mobile calls for example, internetcalls, voipbuster, etc etc etc… there are really many of them)

4) Duplicate the previous one (all-calls-wpin) to no-restriction (Desc: Not any kind of restriction in the whole world ……… of this box :P)

So now just allow hmmm… what is more than calling internationally? Calling SPACE? Well there is a little trick that makes those international outbound routes work without PINs and as i mentioned outbound routes the trick is in there. Now all you have to do is duplicate/recreate those outbound routes that are with PIN sets :D, rename them to xxxxx-nopin and remove any PIN sets from these newly created outbound routes.

Okay now we have created duplicates of some outbound routes, removed the pin sets and now its time to go to Custom Contexts again. First select context no-restriction and make some restrictions (its now or never to make restriction for our bosses, grin…) and in Outbound routes deny access to Outbound routes that have PIN sets and allow to routes that are without pin.
Submit
Then
Go to context all-calls-wpin and deny outbound calls to routes that are without pins, of course allow the ones with pins.

You may ask why we are having double outbound routes with and without pins, well the thing is that when calling internationally you usually have your dial patter like “00.” and avoiding complications like having users call with 00 and bosses call with 99 or whatever is why we make double outbound routes but each group/context have access to its route respectively.

I hope you enjoy this guide.

Updated trunk configuration Asterisk, freepbx and Portech MV-3xx

2009-02-28T20:32:00.003+01:00

This is my new updated functional configuration of Portech.

This guide will help you settings up Trunk in Asterisk (freebox, trixbox, PBIF, etc.) for Portech GSM Gateway.

The new configuration will pass Caller ID.

First we will configure the Portech MV-372 i believe this configuration will also work with Portech MV-370 and other Portech MV-3xx like MV-374.

Login to your portech

Route

Mobile To Lan Settings:

Item CID URL

0 * 192.168.x.x (your asterisk ip)
Lan To Mobile Settings:

Item URL Call num

0 * #
Mobile
- Settings:
  Mobile 1:
  Sip From: Tel/Tel (No reg)
  CLID Presentation: Invocation
  LAN Answer Mode: Income
  
  Do the same for Mobile 2
SIP Settings
- Service Domain
  You only fill Domain Server and Proxy server with your asterisk IP address:
  Domain Server: 192.168.x.x
  Proxy Server: 192.168.x.x
  
  Again do the same for Mobile
- Port Settings
  Make sure SIP Port for Mobile 1 is 5060 and port 5062 for Mobile 2

Other settings are fine you may leave them as they are, only check Network (WAN) settings if you don't have DHCP or you need static IP for Portech gsm gateway.

Don't forget to save changes (should reboot after saving)

Asterisk/Freepbx

Login to your FreePBX and add SIP Trunk

Outbound Called ID: xxxxxxx (put you number here)
Maximum Channels: 1

Outgoing settings
Trunk Name: SIM1 (you may put anything you like)
PEER Details:
host=192.168.x.x (your Portech IP address)
type=peer
port=5060

Incoming Settings:
USER Context: xxxxxxx (put you mobile number)
Leave Incoming settings blank.

Click submit (don't forget the Orange bar on top after you make changes in your server)

Add another SIP Trunk for SIM2

Outbound Called ID: yyyyyyyy (put you phone number here)
Maximum Channels: 1
go to Outgoing settings

Trunk Name: SIM2
PEER Details:
host=192.168.x.x (your Portech IP address)
type=peer
port=5062 (important)

Incoming Settings:
USER Context: yyyyyyyy (put you second phone number)

Again apply changes.

We're almost done. Now to make this work we have to create Outbound Route, so click Outbound Routes
Put Route name as you wish, i have called it Portech_1 (since i will add another and will make it Portech_2)
Dial Patterns: i have put 049XXXXXX because i want only mobile numbers from the same provider to go through this trunk (through Portech) i mean i want to cut the costs right?

Trunk Sequence: i added SIP/SIM1 and SIP/SIM2
You can separate Trunks from OutRoutes if you have SIM cards from two different providers, just create another Outbound Route remove one Trunk from trunk sequence of the first route that we created and add it to this new one. Submit.

Also don’t forget in order to receive calls you need to have Inbound Route setup on Asterisk/freepbx. To get you started just create new Incoming route set you destination to an extension or ring group or any other destionation you would like to transfer calls to.

Only thing you left to do now is click Submit Changes then Apply Configuration Changes and pray for this to work.

Hope this new configuration will work better.

Portech MV-372

2008-06-24T17:36:00.005+02:00

UPDATE: This is old configuration you can try the new configuration in the post above

The Portech MV-372 gave me a lot of headache configuring it right. The configuration for Asterisk specified in documentations wasn't an option for me, working as extension, two dial stages while it works but come on, dial the extension, wait for signal, dial the number...
On the net i found some non-working configs so i had to spend 4 days trying to make it work as a Trunk and not as an extension.

Ok lets move on with configuration ;)

First we will configure the Portech MV-372 i believe this configuration will also work with Portech MV-370.

Login to your portech

Route
- Mobile To Lan Settings:
  Item | CID | URL
  0 | * | 100
  notes: in URL you put your extension of your asterisk you want call from mobile to go. It can be extension, ringgroup, ...
- Lan To Mobile Settings:
  Item | URL | Call Num
  0 | * | #
  notes: URL will match the IP address that will allow to dial through portech, since my server is behind NAT i allowed all IP, haven't try to specify IP. Call num # will receive the number dialed from ip(soft)phone.
SIP Settings
- Service Domain
  Mobile 1 (Realm 1)
  Display Name: Sim1
  User Name: 1001
  Register Name: 1001
  Register Password: xxxxxx (choose a password)
  Domain Server: 192.168.x.x (you asterisk IP)
  Proxy Server: 192.168.x.x
  Mobile 2 (Realm 1)
  Display Name: Sim2
  User Name: 1002
  Register Name: 1002
  Register Password: xxxxxx (choose a password)
  Domain Server: 192.168.x.x (you asterisk IP)
  Proxy Server: 192.168.x.x
  
  notes: Username and registername you can change it to your needs just have a note of them since you will be entering those in asterisk trunks.
- Port Settings
  Just make sure SIP Port for Mobile 1 is 5060 and
  SIP Port for Mobile 2 is 5062

Other settings are fine you may leave them as they are, only check Network (WAN) settings if you don't have DHCP or you need static ip for portech gsm gateway.

Dont forget to save changes (should reboot after saving)

Now lets move to Asterisk.

Login to your FreePBX/Trixbox and add SIP Trunk

Outbound Called ID: xxxxxxx (put you number here)
Maximum Channels: 1
go to Outgoing settings

Trunk Name: SIM1 (i have called it that way)
PEER Details:
host=192.168.x.x (your Portech IP address)
type=peer

Incoming Settings:
USER Context: 1001 (important must match username/registername at Sip settings of Portech)
USER Details:
type=friend
secret=xxxxxx (match SIP Settings password from Portech)
username=1001 (match SIP Settings from Portech)
qualify=yes
nat=yes
canreinvite=no
context=from-internal
host=192.168.x.x (Portech IP)

And then just click Submit Changes (don't forget the Orange bar on top after you make changes in your server)

Add another SIP Trunk for SIM2

Outbound Called ID: xxxxxxx (put you number here)
Maximum Channels: 1
go to Outgoing settings

Trunk Name: SIM2
PEER Details:
host=192.168.x.x (your Portech IP address)
type=peer
port=5062 (important - this is for Mobile 2, remmber Port Settings on Portech?)

Incoming Settings:
USER Context: 1002 (important must match username/registername at Sip settings of Portech)
USER Details:
type=friend
secret=xxxxxx (match SIP Settings password from Portech)
username=1002 (match SIP Settings from Portech)
qualify=yes
nat=yes
canreinvite=no
context=from-internal
host=192.168.x.x (Portech IP)
port=5062

Again apply changes.

We're almost done. Now to make this work we have to create Outboud Route, so click Outbound Routes
Put Route name as you wish, i have called it Portech_1 (since i will add another and will make it Portech_2)
Dial Patterns: i have put 049XXXXXX because i want only mobile numbers from the same provider to go through this trunk (through Portech) i mean i want to cut the costs right? But because you don't have my gun on your head go ahead and do whatever it suits you.

Trunk Sequence: i added SIP/SIM1 and SIP/SIM2
You can sepparate Trunks from OutRoutes if you have sim cards from two different providers, just create another Outbound Route remove one Trunk from trunk sequence of the first route that we created and add it to this new one.

Only thing you left to do now is click Submit Changes then Apply Configuration Changes and pray for this to work.

Hope it will work for you.

Me, company, voip...

2008-06-24T16:01:00.004+02:00

Okay my first blog post, i am starting this Blog to post everything that involves my current job as an IT, so you can imagine that this will be only Tech Blog.

Anyway what i first wanted to post is about VOIP specifically FreePBX/Asterisk/Trixbox

My company is soon to switch to VOIP from traditional PBX and i am assinged to deal with it, make all configuration, preparation, choosing products, etc...

I have decided i will go with Trixbox (asterisk) just that it is Open "source" not that i really have a clue on code that i could intervene but you can very easy find help around the net.

I installed Trixbox on a VMWare ESX 3.5 server and it works prefectly, we have ordered 20 Linksys SPA942 ip phones, 4 SPA 3102 for connecting PSTN lines and 3 x Linksys 24 port gigabit POE switches (just for ip phones) ... (kidding), 2 x GSM gateways Portech MV-372 and cant remmember at the moment what else.

Enough of this, i'll soon start to post my configuration.

Item	CID	URL
0	*	192.168.x.x (your asterisk ip)

Item	URL	Call num
0	*	#