Asterisk and Alcatel OXO

[nunomartins80]

Hello all,
I am trying to connect alcatel omnipcx office (oxo) with SIP trunk ( sip licenses already activated) to an elastix server, we already can dial from elastix extensions to the alcatel OXO extensions number.
But we have two problems the first is the following, when I dial from the elastix client to any extension on the OXO's side the call is redirected to the operator's number.
The second problem is that I can’t dial from the OXO extensions to the elastix extensions.
I would like to know if , anyone has any experience with this type of configuaration between OXO and Elastix , and if so, I would like to know if its possible to help-me with the correct setting to configure the sip trunk (user context, user details and registration string) in the elastix and alcatel XO

Elastix Configuration:
Outgoing Settings
host=192.168.1.10
username=elastix
secret=teste
type=peer
qualify=yes
transfer=yes
dtmf=rfc2833
immediate=no
insecure=very
Incoming Settings

secret=teste
type=user
context=from-trunk

Regard’s
Nuno Martins

[tot3nkopf]

Hi,

A little search on the forum and:
viewtopic.php?f=227&t=18497

About your calls not hitting extensions but attendand ---> check DDI translation in public or private numbering plan.
Also take care of settings like installation number or other canonical related issues --> depending on SIP public numbering description (OMC-->Numbering-->ARS--->SIP public numbering) .

[nunomartins80]

Hello all,

First I would like to Thank you for the help, the problem was in fact canonical issues; we now can dial from, and to any internal extension in the elastix server and the alcatel oxo PBX.
But we still have some problems; I want to allow the extension in the elastix server to make external call using the alcatel PBX OXO. And also associate an OXO DDI to the elastix extension so I can receive external call directly in my internal elastix extension.
Is this possible, and if so can someone help-me with the configuration in both sides to allow this option
Regard's

Nuno Martins

[tot3nkopf]

This is basic transit scenario. Of course this is possible.
Keep in mind:
-Trunk joining
-trunk category (to be able to dial through TELCO trunk)
-send ARS prefix in front of the number if needed from asterisk to OXO

[nmanalich]

Hello, i am having some troubles to configure my Alcatel OXO with an Asterisk 1.8.
I just activated my SIP licenses in the OXO, can comunicate from asterisk to OXO, but i cant comunicate from OXO to Asterisk neither Asterisk to external lines.
can anybody help me? i am new in Asterisk, if you can send me some example just to see what i have to do in the sip.conf and extensions.conf

extensions.conf
[general]
language=es
static=yes
writeprotect=no
autofallthrough=yes

[internos]
include=>buzones
include=>alcatel
include=>hacia_alcatel

exten=>100,1,Dial(SIP/100,25)
exten=>100,n,Hangup()

exten=>200,1,Dial(SIP/200,25)
exten=>200,n,Hangup()

exten=>300,1,Dial(IAX2/300,25)
exten=>300,n,Hangup()

exten=>500,1,Dial(SIP/500,25)
exten=>500,n,Hangup()

exten=>789,1,GoTo(menu,s,1)

exten=>345,1,GoTo(bucle,s,1)

[externos]
exten=>_0XXXXXXXX,1,Dial(SIP/IP OXO/${exten:1})
exten=>_0XXXXXXXX,n,Hangup()

exten=>100,1,Dial(SIP/100,25)
exten=>100,n,Hangup()

exten=>200,1,Dial(SIP/200,25)
exten=>200,n,Hangup()

[grabacion]
exten=>999,1,Answer()
exten=>999,n,wait(1)
exten=>999,n,Record(menu-personal.gsm)
exten=>999,n,Hangup()

[buzones]
exten=>5100,1,Answer()
exten=>5100,n,VoiceMail(100@default)
exten=>5100,n,Hangup()

exten=>5200,1,Answer()
exten=>5200,n,VoiceMail(200@default)
exten=>5200,n,Hangup()

exten=>5300,1,Answer()
exten=>5300,n,VoiceMail(300@default)
exten=>5300,n,Hangup()

exten=>5000,1,Answer()
exten=>5000,n,VoiceMail()
exten=>5000,n,Hangup()

exten=>5900,1,Answer()
exten=>5900,n,VoiceMailMain(100@default,s)
exten=>5900,n,Hangup()

exten=>6000,1,Answer()
exten=>6000,n,VoiceMailMain(300@default,s)
exten=>6000,n,Hangup()

[alcatel]
include=>internos

exten=>_.,1,Dial(SIP/IP OXO/${EXTEN})
exten=>_.,n,Hangup()

[hacia_alcatel]
include=>externos

exten=>_0XXXXXXXX,1,Dial(SIP/IP OXO/${exten:1})
exten=>_0XXXXXXXX.,n,Hangup()

sip.conf

[general]
qualify=yes
context=default
allowguest=yes
context=internos
disallow=all
allow=alaw
allow=ulaw
allow=gsm
tcpenable=yes
transport=udp,tcp
bindaddr=0.0.0.0
bindport=5060

[alcatel]
canreinvite=no
context=alcatel
host=ip OXO
type=friend
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
nat=no
dtmfmode=rfc2833

[desde_alcatel]
canreinvite=no
host=0.0.0.0
insecure=invite,port
;port=5060
type=friend
;context=internos
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
nat=no


[100]
language=es
type=friend
secret=xxxx
host=dynamic
context=externos
nat=yes
dtmfmode=rfc2833
mailbox=100@default

[200]
type=friend
secret=xxxx
host=dynamic
context=internos
nat=yes
mailbox=200@default

Thanks and excuse me any heavy mistake in the code and the post

[yabapolido]

Olá Nuno.
Could you give me some hints about trunk between OXO and Asterisk?
I'm having troubles too.
I can reach the Asterisk extensions using *509 code, but can't dial 300 for example.
OXO has 200 to 299
Asterisk has 300 to 399
Also, same problem as you, from Asterisk do OXO, always end up in operator.
Thanks.

[samysaad]

dears,
please i need the trank configuration between oxo and elastix
cuz i have the same problem

[cavagnaro]

Search the forum. No magic tricks

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[adnaninline]

Hello;
I am having trouble configuring FreePBX and Alcatel OXO. Have some configuration working that I can call OXO extension from FreePBX extensions and vice versa. from FreePBX call connects to OXO extensions within nano second but from OXO extension to FreePBX it is taking around 30 seconds to reach to extension. Both outgoing external calls and incoming external calls are ending up to OXO attendant (Receptionist). Do not have enough knowledge of Dialing plans. If someone can plz help in resolving the issue and if some guide/link can be provided to understand Dialing plans and OXO routing, would be great help.

Thank you So much.

[cavagnaro]

So...if at least you show us what have you configured so one ends up guessing...would be great

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