Configuration for voiceblue

FreePBX Tips and Tricks
1

hello @ all,

i am having no success to get a voiceblue GSM gateway to work. my main problem is how to adapt the instructions given by the manual to freepbx. here is what they say to do:

quote start
Outgoing Calls

The core of Asterisk connection lies in the /etc/asterisk/extensions.conf file.

Open this file in your favourite editor and add the following lines:
exten => _6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,r)
exten => _7XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,r)
exten => _8XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,r)

Once you have saved and closed the file, restart Asterisk and from now on all calls starting with 6,7,8 should be routed to the 2N VoiceBlue Lite GSM Gateway.
quote end

where would that go in freepbx ? in my case i need to route 086 prefix and 087 prefix via the voiceblue box.

quote start
Incoming Calls

It is recommended to make a little restriction for incoming calls to prevent unauthorised persons from calling over your system.
Since 2N VoiceBlue Lite works with the SIP, modify the /etc/asterisk/sip.conf file where the 2N VoiceBlue Lite section could look as follows, for example:

[voiceblue]
type=peer
insecure=very
disallow=all
allow=alaw
host=10.0.0.20
username=voiceblue
permit=10.0.0.20/255.255.255.255
qualify=yes
canreinvite=no
call-limit=4

Again, restart Asterisk after saving the file. After that, Asterisk will be ready to receive calls coming from the 2N VoiceBlue Lite GSM Gateway.
quote end

and where would that go in freepbx ?

kind regards
Jan

2

snoozer,

First off those are directions for a asterisk only configuration. So that you know you should not edit the extensions.conf or SIP.conf files. See: http://freepbx.org/configuration_files for more info.

As for the info on creating the trunk that all looks like the correct needed information. So you really jus need to take a look at the directions on how to configure a trunk using almost any one of the other providers documented here on this site and you’ll get the general feel for what needs to be done.

Generally you need to go into the trunk options and configure a sip trunk. It will place the information in the correct sip.conf sub files files for the sip configuration.

As for the editing the extensions.conf part once a trunk is created you can create a outbound route to use that trunk, you enter the dialing patterns you need in to a new route and tell it to use those patterns on this newly created trunk. It will place that information into the correct extension.conf sub files.

If you go to the Documentation section and then select the howto’s you’ll find many set’s of directions for “Setting up for a VoIP provider trunks” read a few and you’ll see the common thread in them all about how to do it also take a look at “HOWTO: SPA-3102 and FreePBX” between the both of those you should be able to adopt the directions provided to fit this particular gateway.

3

fskrotzki has given you very good advice here. You would be far better off to make this work within FreePBX if possible.

That said, if you want to temporarily insert these lines for testing purposes:

exten => _6XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,r)
exten => _7XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,r)
exten => _8XXXXXXXX,1,Dial(SIP/${EXTEN:0}@10.0.0.20,r)

… you could put them in /etc/asterisk/extensions_custom.conf, under the existing [from-internal-custom] context. But if you find they work as expected there, you really should create a proper outbound route and trunk. However, I know the desire to have a way to quickly test something to make sure the darn thing is working!

(The PROPER way is to create an outbound route, use those same patterns but without the leading underscore, and then send them to the trunk associated with the device).

4

hi,

i got it working. in outgoing peer details i have:

type=peer
insecure=very
disallow=all
allow=alaw
host=10.x.x.x
username=yyyy
secret=xxxx
qualify=yes
canreinvite=no

in incoming i have:

secret=xxxx
from-user=yyyy
type=user
context=from-trunk

given it a trunk name and a name for the user context and it works. outgoing routes where simple just as all other routes to trunks.

another thing which just came up is. the voiceblue is receiving a call and i can either dial a number of an extension or give the voiceblue and extension number to dial after it has taken an incoming call. but it only works with extensions. i can not dial a queue, ring group or misc application. i try to get into the IVR which i have made as a misc application to access it via 699. the freepbx system will not allow me to dial that number via the GSM gateway. any ideas what i can do to solve that problem ?

thanks
Jan

5

Problem solved !

i have now changed the incoming settings to:

type=friend
host=10.x.x.x (ip address of the GSM gateway)
context=from-internal

and have calls routed to the IVR by configuring the GSM gateway to call the misc application 699. everything else i can handle perfectly fine in the IVR.

regards
Jan

6

Hello snoozer, could you post a sample working Voiceblue .ini file? Thanks.

7

hello biccy,

my voiceblue ini is here:

[System parametres]
X30=#,#,#
X31=7*,9#
X32=*55,#55
X33=*33,#33
S70=10.27.99.12
S71=255.255.255.0
S72=10.27.99.253
S91=7,0
S98=1234
X20=00.00,00.00
[Ethernet parametres]
E01=1,8,0
E02=600,10,600
E03=8000,8998
E08=0
E09=1
E10=10.27.99.13:5060
E11=10.27.99.13:5060
E14=10.27.99.13:5060
E16=0.0.0.0
E17=0.0.0.0:3478
E20=4,0
E23=1,0
E29=2,0
E80=,
E81=
E82=gsm
Codec=1
[GSM parametres]
G09=0,1,1,0
G02=1,2,2
G04=0,3,3
G08=2,9,9,3
G103=0,2,2,2,2
G101=
G06=
[Groups assignment]
G00=0100
G90=0100
[Outgoing groups]
S85=
S86=
G11=,2,0,0,1,1,1,0
G19=1,0,0,1,0,0,
G21=,2,0,0,1,1,1,0
G29=1,0,0,1,0,0,
G31=,2,0,0,1,1,1,0
G39=1,0,0,1,0,0,
G41=,2,0,0,1,1,1,0
G49=1,0,0,1,0,0,
G109=
[Incoming groups]
G91=3,3,3,0,1,
G95=699
G191=0,
G99=0,0
G92=3,3,3,0,1,
G96=699
G192=0,
G93=0,3,3,10,1,
G97=
G193=0,
G94=0,3,3,10,1,
G98=
G194=0,
G199=
[Network list]
N10=86
N11=086,087,1741/4,171/3
N19=,9
N20=87
N21=087,086,1741/4,171/3
N29=,9
N30=/
N31=0,1,2,3,4,5,6,7,8,9
N39=,9
N40=/
N41=0,1,2,3,4,5,6,7,8,9
N49=,9
N50=/
N51=0,1,2,3,4,5,6,7,8,9
N59=,9
N60=/
N61=0,1,2,3,4,5,6,7,8,9
N69=,9
N70=/
N71=0,1,2,3,4,5,6,7,8,9
N79=,9
N80=/
N81=0,1,2,3,4,5,6,7,8,9
N89=,9
[Autorouting table]
A=
[Extension table]
M=
[Lcr table]
L=1,0:00/24:00,1,0,2,0:00/24:00,2,0

this works with 2 sim cards, bot sim cards accept the 2 major mobile prefixes in ireland which are 086 and 087. this is because with number porting you can not determine anymore by the prefix which network it is.

here the config in free pbx:

Trunk Name: GSM
type=peer
insecure=very
disallow=all
allow=alaw
host=10.27.99.12
username=gsm
secret=xxxxxxxxxx
qualify=yes
canreinvite=no

User Context: gsm_in
type=friend
host=10.27.99.12
context=from-internal

i am not sure if i have the simplest possible config but it works. all outgoing calls with 86 or 87 will be routed via the GSM trunk. in the gsm gateway that will be tripped off from the actual number. 86 or 87 determine if i use O2 or Vodafone. so if i dial 860862222555 the number will be called on the 086 network as 0862222555. all inbound calls go to 699 which is a Misc Application and points to our normal IVR which everybody get on the landline too. with DISA you can now even go further. if you have free calls mobile to mobil you can basically ring the office, access DISA with your password and ring any number just as if you would be in the office.

only REALLY bad thing about the voiceblue box is that i have not been able to send sms via a terminal and AT commands.

hope this helps and sorry for the late answer.

Jan