Good Morning everyone!

We had a client which "forced" us to use a GXW-4108 (normally we would not touch such equipment.... :-) ).

We had some trouble to get this working for us, and there is not much good help out there for this (since the manuals are not exactly of good quality).

Scenarios:

- Calls are coming and must be routed to Asterisk, where we hit the main IVR
- Phones must be able to call out by simply press 9 followed by the number

Sounds simple but not that straight forward with this peace of .... equipment haha.

First you need to find out the IP-Address your GXW-4108. our box was set to DHCP, since you have no display on the box (or any other way to figure the IP out) you need to download and install this tools:

WinPcap : WinPcap_4_1_2.exe
C++ redistributable: vcredist_x86
IP Query (Grandstream): IP Query

IP Query will scan your network and find the Grandstream GXW-4108 IP-Address.
Attention: You need to run the tool IP Query as administrator

Choose your network card on which you like to "listen". After a while you will get the IP-Address and MAC address of your GXW-4108.

Attention: If you don't get the address, reboot your GXW-4108.

Next step is to login into your GXW-4108 via browser.

Key in discovered IP into your browser and login with password: admin

Now you need to make sure you have the following settings:

Under FX0-Lines:

Still under FXO-lines:

Stage Method(1/2):

set to one, we use GXW-4108 like a trunk, all calls are routed to
one extension. Which we forward to IVR.

Unconditional Call Forward to VOIP:

USER ID ch1-8:1245;
Means all lines (1-8) are sent to extension 1245 in FreePBX

Sip Server
ch1-8:p1;
means all incoming calls (line 1-8) are send to sip server profile 1 (p1) which we configure later under "Profile 1"

Sip Destination Port
ch1-8:5060;
means all lines connect to SIP port 5060 (the port where FreePBX lsiten to)

Now you need to configure the Profile one like this:

Just use your asterisk server IP-Address.

Next we need to configure FreePBX.

Setup the trunk to GXW-4108 by creating a new SIP trunk with following settings:

key in Trunk Description and this settings

the "host" is the IP-Address of you GXW-4108.

That's it for the trunk.

Now make sure you have a extension matching the User ID from the GXW-4108. in our case this is 1245. (create a normal SIP extension). Since we want to route all calls from this "dummy" extension to our IVR we do not activate voice-mail on it.

In the Follow-Me settings of 1245 we set the Destination if now answer to our main IVR. Also set the the Ring strategy to" hunt

Since we never register the 1245 extension (no phone actually connected to asterisk with this extension) all incoming calls are immediate sent to IVR.

That's it for the incoming calls.

No we create a new outbound route for prefix 9.
If any user on a phone press 9 followed by any number we sent the call to our new
GXW trunk:

That's it you will be able to make and receive calls.

Hope this helps some of you guys out there.

Cheers
Daniel

Thanks a lot to all the great organizers!!!
Outstanding job.

I enjoyed the speech :-)

Facebook Event-Link: Conference

Cheers
Daniel

We have a client which requested to be able to lock/unlock phones by the user.

The scenario is that if the user is not at his desk, he/she want to be able to lock the phone with a personal password so no one else can make phone calls from this phone until the user unlocks it again.

We found some existing code from ch.jacobsen and GWalmsley on the FreePBX forum which we slightly modified.
As well we created some additional voice prompts (thanks to Shazmin from BFM www.bfm.com.my)

Usage of this function:

Lock phone:

- User keys in the feature code *888 (in our setup, you can change this)
- User keys in password followed by pound key
- User confirms password followed by pound key
- Phone is locked

Unlock phone:

- User dial any number
- System check if phone is locked
- If the phone is locked User need to key in password followed by the pound key
- Call is routed

How to install this feature:

Step 1:
Copy the following two context into extensions_custom.conf

[custom-checklock]
exten => _.,1,NoOp(AMPUSER/${CALLERID(num)}/locked)
exten => _.,n,Gotoif($[${DB_EXISTS(AMPUSER/${CALLERID(num)}/locked)}=0]?carryon)
exten => _.,n,Set(PINCOUNT=0)
exten => _.,n,Set(VALIDPIN=1234)
exten => _.,n(readpin),Read(PIN,lockunlockpassword,,,,)
exten => _.,n,Gotoif($[${PIN}=${VALIDPIN}]?carryon)
exten => _.,n,Gotoif($[${PIN}=${DB(AMPUSER/${CALLERID(num)}/locked)}]?carryon)
exten => _.,n,Set(PINCOUNT=$[${PINCOUNT}+1])
exten => _.,n,Playback(lockpasswordnotcorrect)
exten => _.,n,GotoIf($[${PINCOUNT}>3]?h)
exten => _.,n,Wait(1)
exten => _.,n,Goto(readpin)
exten => _.,n(carryon),NoOp(${DB_DELETE(AMPUSER/${CALLERID(num)}/locked)})
exten => _.,n,Goto(from-internal,${EXTEN},1)
exten => 999,1,Goto(from-internal,${EXTEN},1)
exten => h,1,Hangup

[custom-setlock]
exten => s,1,NoOp(In Set Lock)
exten => s,n,Read(PIN1,lockpassword,,,,)
exten => s,n,Read(PIN2,lockpasswordconfirm,,,,)
exten => s,n,Gotoif($[${PIN1}=${PIN2}]?okey)
exten => s,n,Playback(an-error-has-occured)
exten => s,n,Hangup
exten => s,n(okey),Set(DB(AMPUSER/${CALLERID(num)}/locked)=${PIN1})
exten => s,n,Playback(lockedsuccess)
exten => s,n,Hangup
exten => h,1,Hangup

Step 2:

Copy and unzip the following file(s) into /var/lib/asterisk/sounds

Download new voice prompts: lockprompts.zip

3. Change the context for the extension in FreePBX

Change the phone extension context to: custom-checklock

4. Create a new custom destination

Next you need to create a custom destination:

5. Create new Misc Application

Next you need to create a new Misc Application:

Click on the picture for larger size.

That's it! Enjoy.

Cheers
Daniel

Hi

Intuit is growing and need to "still it's hunger" for more qualified and enthusiastic
Intuiter's....

Here the job description:

We are looking for an enthusiastic techie to join our project team. We specialize and are very focussed on 2 open source products namely Nagios and Asterisk and solutions that compliment them. We also develop our own solutions based on them.

Our corporate heart is to the Open Source community (Does not need any further clarification for those familiar with Open Source). We're passionate about our solutions and we need fun, enthusiastic and brilliant minds to grow with us, you will be working in a dynamic team of people all with the same drive as you.

You'll be responsible for deployment of Nagios and Asterisk bases solutions and these can range from a single extension/node setup to multi clusters in a load balanced environment, you will work with the Project Manager but need to be very goal oriented thus take ownership of your projects and get it done (no micro management). Depending on your skill level you will also be asked to troubleshoot issues for existing as well as new clients (we offer global -remote- support) as well as rotate on our 24x7x365 support stand-by.

To achieve all off this efficiently we have developed a structure approach that is very process driven yet leave enough room for the creative technical mind. You also be involved in creating new and/or work to enhance these processes and flows.

Requirements;
* MUST have Open Source knowledge (LINUX various distro’s), does not matter if this is via education or self taught, as long as you have and can show the skills
* Nagios and/or Asterisk knowledge is a huge plus
* Debian, RRD, MRTG, SNMP, AsterNic, Redfone, Digium, FreePBX knowledge a plus
* Fresh grad can also apply, we will give on the job training
* Applicants should be Malaysian citizens or hold relevant residence status.
* Personality; Passionate about Open Source and can’t accept “not able to do”

Perks;
- Salary range between 2.000 for Fresh grad to 4.500 Very Senior.
- Entitled for project commission (usually a 2% bonus over the services sold payable to engineering team upon UAT)
- Entitled for Over time
- Entitled for Standby allowance
- Mileage, parking, toll etc claims
- Medical benefits
- 20 days leave

Duration
- 1 year contract subject to renewal with a 2 months probation period.

Contact us via:

www.intuit.my

or send your resume to

Frank (frank@intuitinnovations.com)

Cheers and im looking forward to work with you :-)
Daniel

We lately sold a couple of Yealink T18.
Good and affordable "entry level" IP phone.

In case you are looking for the manual and latest firmware (which somehow is not online on the Yealink website):

Firmware: 18.0.0.70.rom
Documents: T18 User Manual-V18.2.pdf

Cheers
Daniel