please advise

51 posts · Brekeke PBX Forum

tissa 2011-01-19 15:15 UTC Original post

1. Brekeke Product Name and version: pbx version 2.4.7.3

2. Java version: jre 6

3. OS type and the version:xp

4. UA (phone), gateway or other hardware/software involved: gateway

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:I'm using BREKE PBX software and have installed the trial version 2.4.7.3.

My Quintum AFT400 gateway is located in Sri Lanka

Currently,my PAP2-PAP2 PBX server is running perfectly.

Also I have registered my Quintum AFT400 as PBX user.

However,when a call is orginated from PAP2, call is not being connected to Quintum and is giving a busy tone.

Can you please advise me on how to give ARS settings abd dial plan for quintum on this regard?

Thanks & Regards,
Tissa

hope 2011-01-19 18:04 UTC

can you make call from pap2 to brekeke pbx users like autoattendant or registered phones?
what ars have been created for pap2 and quintum?

tissa 2011-02-26 06:19 UTC

Hi,
Thanks for replying me.
Yes,it is possible make call from pap2 to

brekeke pbx users you mentioned(auto attendant

or registered phones)

Also regrading your second question,at the

moment my quintum is only registered as

registered client.I want to use the quintum as a

SIP terminiation gateway and terminate the call

of PAP2 users via PBX server.

I hope your clear on my requirement and can you

please urgently advice on what the required dial

plan, ARS settings and neccessary steps?

Thanks
Tissa

hope 2011-02-28 18:19 UTC

ARS
matching
sip:([0-9]{7,25})@
deploy:
sip:$1@quintum_IP

when caller dial number is 7-25 digits the call will be sent to quintum IP

tissa 2011-03-01 16:53 UTC

Hi,

Thanks for replying me.I have setup the ARS settings as you mentioned in the reply. But it seems my problem hasnt not been solved yet.

I would like to mention here configuration details further.

1. My X- Lite soft phone is a registered client which is not in the local network.
2. PAP2 is registered in local network with PBX server installed.
3. Quintum AFT 400 is registered in the local network with PBX server installed.

Furthermore, X-Lite softphone can call to PAP2
PAP2 can call to X-Lite
Quintum can call to PAP2 or X-Lite(PSTN to VoIP)

The problem now i'm having is when dialing wither from PAP2 or X-Lite to quintum it's not giving PSTN dial tone.Instead PBX's IVR is responding "Person who you calling is not available"

Much appreciate if you could please urgently reply.

Thanks
Tissa

hope 2011-03-01 18:17 UTC

what number is dialed from PAP2 or xlite?
with the ars in above, just dial the pstn number from caller, the call will be sent to gateway.

if there is any other ARS rule for inbound or outbound,
disable them just leave the one above when testing.

here is some same configuration for quintum
http://wiki.brekeke.com/wiki/Quintum-Tenor-ASM200
http://wiki.brekeke.com/wiki/Quintum-Tenor-ASM200
if your gateway is set up like above wiki, you need to change ARS to the one in wiki post,
and from caller dial prefix 9+pstn destination number

if still not work, please capture packets and check the response from quintum gateway.

tissa 2011-03-08 15:09 UTC

Hi,

I couldn't register my Quintum AFT 400.But I was able to register my AFT 400 unit earlier.However the Xlite and PAP 2 are successfully registered.I also tried registering another SIP account (for e.g Webcalldirect) and it was successfully registered.Only the Brekeke PBX is failed to registered . In my router PORT forwarding is set as follows:

PORT UDP:5060 10000 10999 11000 11999

Can anyone tell me what may be the issue in here.

Thanks in Advance
Tissa

hope 2011-03-08 19:20 UTC

have you set router global ip at Brekeke SIP Server Admintool > [Configuration] > [System] > [Network]>[Interface Address] = Your router's global IP Address?
is authentication for register/invite set ON?
if yes, create user authentication account for quintum with same user id as quintum register sip id.
dial plan can be used to accept calls from quintum even it doesnot register at brekeke.

it is better to capture packets for register and call about quintum to check the cause of problem

tissa 2011-03-11 16:10 UTC

Hi,

Thank you very much for the solution provided.It worked.Now i get the dail tone from

PSTN.This is how it dial;99+PSTN.Now i want to dial in international format like

00941+PSTN.Can you help me in giving me the dial plan and ARS setting?

The existing dial plan and ARS settings are as follows:
ARs- Matching Pattern: sip:([0-9]{7,25})@
Deploy Pattern : sip:$1@192.168.1.5

Dial Plan-Matching Pattern:$request=^INVITE
To=sip:9(.+)@
Deploy Pattern: To=sip:%1@192.168.1.5

(QUINTUM IP :192.168.1.5)

THANKS,
Tissa

hope 2011-03-11 20:45 UTC

glad to know it worked.
which one solved your problem? add global ip or add dail plan?
i would like to know if you can make pstn call successful with following settings in order to make ARS for international format calls:
1. disable the dial plan created for pstn call
2. change ARS like below
ARS settings
Matching Pattern:
To: sip:99([0-9]{7,25})@
Deploy Pattern :
To: sip:$1@192.168.1.5

if above one does not work, please try the following ARS
Matching Pattern:
To: sip:99([0-9]{7,25})@
Deploy Pattern :
To: sip:9$1@192.168.1.5

which ARS works when call to quintum gateway?

and for international format call, is 00941 a part of destination number for country or area code?

tissa 2011-04-07 19:48 UTC

We've got our two PBX Servers in two differnt countries( ex: A & x ).

1. VOIP Gateway (FXO) in A is connected to the PBX Server which
passes the call termination.
2. How I can connect the PBX Server in X (originating) to the PBX
Server in A.

3. When we use the PBX Server through the billing Radiuscat in X
(license version), the billing Radiuscat functions correctly
through the PBX Server but the call charge is not deducted from
the value of the created card(prepaid phone card).

Please give instructions.

hope 2011-04-08 00:49 UTC

How I can connect the PBX Server in X (originating) to the PBX
Server in A.
you can register pbx X to the other pbx A by using ARS on pbx X like the sample templates for itsp
when call from X to A just dial through registered ARS route pattern with a certain prefix

and do the same when call from pbx A to pbx X

for billing, maybe it is the setting problem on Radiuscat.
you can contact with svk

tissa 2011-04-08 16:07 UTC

Hi,
Thanks for your reply
BUt I'm not aware on how to configure both PBX

servers as ITSP?
Can you please explain on how to ?(including

relevant ARS and sample template for ITSP)

hope 2011-04-11 17:35 UTC

at pbx A, create ARS rule like
General:
register uri: sip:1234@pbx _B_IP
Proxy Address: pbx_B_IP
User: 1234
password: password

pattern-IN
matching patterns
To: sip:1234@

Deploy patterns
To: 1000

at pbx B, create "user Authentication" for pbx A user 1234 at pbx B sip server
and create a pbx user as 1234 pbx B pbx side

from pbx B call 1234 the call will be sent to pbx A and answer by pbx A user 1000

tissa 2011-04-17 23:14 UTC

Hi,
I have install pbx server in my local area network.with quintum voip gateway.It works smoothly and connects with PSTN when dial 99#from PAP2 user. Here are the ARS settings and Dial plan I have applied.
ARs- Matching Pattern: sip:([0-9]{7,25})@
Deploy Pattern : sip:$1@quintum_IP (Globel IP)

Dial Plan-Matching Pattern:$request=^INVITE
To=sip:9(.+)@
Deploy Pattern: To=sip:%1@quintum_IP (Globel IP)

now my new requirement was to install the pbx server in one country (which is in france) and Quintum voip gateway to be installed in another country.(INDIA) I was able to installed these in respective countries successfully.Quintum and Pap2 registered. But when a pap2 user dials 99# the call is not getting connected and gives busy tone.
Can you help me in finding in proper solution in configuring this.
Thans in Advance.

hope 2011-04-18 23:42 UTC

is pbx server behind router?
if yes, you need to set router global ip on brekeke server and set port forwarding as below
http://wiki.brekeke.com/wiki/Set-port-forwarding

it is better to capture packets and check if brekeke sent invite to quantum gateway new ip and who return busy, brekeke or gateway.

tissa 2011-05-06 16:37 UTC

My Quintum Gateway is configured as a PBX server and is country specific.It is working

properly.
My PBX's ARS got the ITSP registered.But failed to do international call termination

and is giving a busy tone.Hoever when i remove the PBX dail plan configured in

Quintum,ITSP can terminate the call.So i need to setup this correctly so that i can do

international call termination.
Can anybody tell me are there any settings to be applied to work both Quintum and

ITSP?

Following is the dial plan configured in the Quintum:
Matching Pattern: Deploy Pattern:
$request=^INVITE To=sip:%1@quintum_gw_ip_address
TO=sio:00(,+)@

Thanks in Advance.
Tissa

hope 2011-05-09 18:10 UTC

Following is the dial plan configured in the Quintum:
Matching Pattern: Deploy Pattern:
$request=^INVITE To=sip:%1@quintum_gw_ip_address
TO=sio:00(,+)@

where this dial plan is used? at quitum side or Brekeke pbx/sip server?
and it should be To=sip:00(.+)@

tissa 2011-05-09 23:24 UTC

it use Brekeke pbx/sip server

tissa 2011-05-09 23:34 UTC

yes, it use brekeke pbx/sip server

To= sip00(.+)@

hope 2011-05-10 17:46 UTC

if there is ARS rule at brekeke pbx to send call to quintum, the dial plan isnot needed for the call at sip server side.

or change dial plan as
matching pattern:
$request=^INVITE
To = sip:(00.+)@

deploy patterns:
To=sip:%1@quintum_gw_ip_address

the above dial plan will router call with prefix 00xxxx in dialing number to quintum ip without going though pbx, and keep prefix 00 in the dialing number.

do you need call to quintum bypass brekeke pbx?

tissa 2011-05-11 17:06 UTC

Yes it is ok to bypass the Brekeke PBX when calling to Quintum/

hope 2011-05-11 17:38 UTC

change dial plan as
matching pattern:
$request = ^INVITE
To = sip:(00.+)@

deploy patterns:
To = sip:%1@quintum_gw_ip_address

put above dial plan on top of other dial plans.
with this dial plan, when there is call to 001234567, brekeke will send call to quintum and keep the destination number as 001234567

does changed dial plan work?
if not, what response is sent from quintum?
and what prefix should be used for the call to quintum?

tissa 2011-05-17 13:15 UTC

Hi,

My requirement is not fullfilled yet.My Quintum Gateway is configured to a PBX Server

so that calls are terminated to one country(India).It works properly.However what i

would like to have is to get calls to other international countries via ITSP and do

the call terminations using the same PBX server.I have followed many instructions but

failed to achieve this requirement.Can anyone please help me.

This is my Dialing Plan: Maching Pattern
$request=^INVITE
To=sip:(.+)@

Deploy Pattern
To=sip:%1@quintum_ip_address

Thanks In Advanced
Tissa

james 2011-05-17 17:17 UTC

It seems you want to make a call to a destination user through the ITSP. Right?

but... Why do you point "quintum_ip_address" in the Deploy Pattern??

I suppose you should set the ITSP's IP address at the Deploy Pattern.

If the above is not correct, explain the purpose clearly.

tissa 2011-05-18 12:32 UTC

Hi,
Let me explain it again clearly.

My brekeke sip server is running in france.
Quintum gateway is located in india and it is registered with pbx server in france
The Quintum voip gateway is only using for call termination to india.

So, pap2 users registered in pbx server can get call to india perfectly.
pap2 users dialing format : 00 91 123456789 #
Pbx server Dial plan for quintum is as follows;

Maching Pattern
$request=^INVITE
To=sip:(.+)@

Deploy Pattern
To=sip:%1@quintum_Ip_Address

However, My problem is , since my Quintum gateway (registered with pbx server)is only configured to terminate

calls to india, pap2 users who need to dial to other countries like singapore(other than India) are unable to get

their calls.
Therefore,in order to call other countries an ITSP account must be registered with pbx server.
Please kindly advise me on required ARS settings and Dial Plans to register ITSP account with pbx server.

ITSP Account Details:
user id-tissa
password-123456
itsp proxr server address; sip.voicetrading.com

Thanks in Advance
Tissa

tissa 2011-05-19 23:48 UTC

Posted: Wed May 18, 2011 5:32 am Post subject: please advice

--------------------------------------------------------------------------------

Hi,
Let me explain it again clearly.

My brekeke sip server is running in france.
Quintum gateway is located in india and it is registered with pbx server in france
The Quintum voip gateway is only using for call termination to india.

So, pap2 users registered in pbx server can get call to india perfectly.
pap2 users dialing format : 00 91 123456789 #
Pbx server Dial plan for quintum is as follows;

Maching Pattern
$request=^INVITE
To=sip:(.+)@

Deploy Pattern
To=sip:%1@quintum_Ip_Address

However, My problem is , since my Quintum gateway (registered with pbx server)is only configured to terminate

calls to india, pap2 users who need to dial to other countries like singapore(other than India) are unable to get

their calls.
Therefore,in order to call other countries an ITSP account must be registered with pbx server.
Please kindly advise me on required ARS settings and Dial Plans to register ITSP account with pbx server.

ITSP Account Details:
user id-tissa
password-123456
itsp proxr server address; sip.voicetrading.com

Thanks in Advance
Tissa

hope 2011-05-20 00:10 UTC

check "Administrator's Guide (Basic)" at http://www.brekeke.com/download/download_pbx_doc_en.php
section 4.17 on page 22

tissa 2011-06-17 16:12 UTC

Posts: 1

My New Gateway - GRANDSTREAM 4108 GXW 4108

Hi.
When a sip user attempt to dial a number which is in international format (example: 00-94-11-2845300 ) call is getting connected to pstn but does'nt reach number. How ever,when it dials a number whit out international prefix (example: 2845300) call is being reached successfully.Please kindly advise me on this with neccerssary dial plan.
thanks.
tissa

ARS
matching
sip:([0-9]{7,25})@
deploy:
sip:$1@Grandstream 4108_IP

matching pattern:
$request=^INVITE
To = sip:(00.+)@

deploy patterns:
To=sip:%1@Grandstream 4108_gw_ip_address

hope 2011-06-21 17:08 UTC

matching pattern:
$request=^INVITE
To = sip:009411(.+)@

deploy patterns:
To=sip:%1@Grandstream 4108_gw_ip_address

tissa 2011-07-11 15:15 UTC

Greetings!

I must mention the following again for your information.

When a sip user attempt to dial a number which is in international format (example: 00-94-11-2845300 ), call is getting connected to pstn but it doesn't reach the number. How ever,when it dials a number without a international prefix (example: 2845300) call is being reached successfully.

Now my Questions are as follows
1. I want to know the settings for dialing the international call without PBX server dial plane.

Can please kindly advise on this?

2. Also, if Granstream 4180 gateway requires a dial plan to Granstream 4180 gateway, I would also like to have the dial plane for it.

Thanks,
Tissa

hope 2011-07-11 20:36 UTC

when making call with prefix 009411 check this call detail from pbx/call status if correct ARS rule is applied to the call

and at same time check if any changes for dialing number made from dial plan applied to call from sip server/active sessions about call.

if some unexpected changes made to call dialing number, change the ARS or dial plan.

tissa 2011-07-16 11:49 UTC

Hi,
Please find below for results I obtained.

Also I retreieved the following for ARS pattern

ARS matching
sip:([0-9]{7,25})@

deploy:
sip:$1@Grandstream_ 4108_IP

- active session/pbx call status when dialed with 009411

X-SID 28
From-uri sip:1599@192.168.1.15:5060
From-ip 127.0.0.1:15062 (UDP)
From-if 127.0.0.1:5060
To-uri sip:0094112790340@192.168.1.3
To-ip 192.168.1.3 (UDP)
To-if 192.168.1.15:5060
Call-ID ad8d8be5-7f93e65e-5767b6cc-ed52b298@192.168.1.15
rule From PBX 1
plug-in InviteSession
sip-packet-total 6
listen-port 5060
session-status Talking
time-inviting Fri Jul 15 16:33:26 IST 2011
time-talking Fri Jul 15 16:33:34 IST 2011
length-talking 00:00:03
rtp-relay off

Status
ID 113000000006
Status CALLING
Call Park
Conference
Start Fri, Jul.15, 2011 04:35:29 PM

UAs
User ARS URI Connected
1599 101 <sip:1599@192.168.1.15> Disconnect
0094112790340 101 <sip:0094112790340@192.168.1.3> Disconnect

-active session/pbx call status when dialed without 009411

Status
ID 113000000007
Status TALKING
Call Park
Conference
Start Fri, Jul.15, 2011 04:37:08 PM

UAs
User ARS URI Connected
1599 101 <sip:1599@192.168.1.15> 04:37:11 PM Disconnect
2790340 101 <sip:2790340@192.168.1.3> 04:37:11 PM Disconnect

EX-SID 54
From-uri sip:1599@192.168.1.15:5060
From-ip 127.0.0.1:15062 (UDP)
From-if 127.0.0.1:5060
To-uri sip:2790340@192.168.1.3
To-ip 192.168.1.3 (UDP)
To-if 192.168.1.15:5060
Call-ID 257e3f68-1f785b7-c54ec713-6fb651f1@192.168.1.15
rule From PBX 1
plug-in InviteSession
sip-packet-total 6
listen-port 5060
session-status Talking
time-inviting Fri Jul 15 16:38:29 IST 2011
time-talking Fri Jul 15 16:38:31 IST 2011
length-talking 00:00:01
rtp-relay off

Please advise?

Thanks,
Tissa

hope 2011-07-18 17:57 UTC

are the same ARS rule name shown in pbx side / call status page details about call when dial with/without 009411?

if yes, then create another ARS pattern-OUT with smaller number in priority field than the current one and set as

matching:
To: sip:009411([0-9]{7,})@

deploy:
To: $1@Grandstream_ 4108_IP

tissa 2011-07-19 14:49 UTC

hi,
thank for your repply me. Call reached successfully for your settings. Another problem having is call is not reached when dialed differant area code (00 94 31 2256789) or a mobile number (00 94 77 1234567)

Kindly Advise?
thanks

hope 2011-07-19 17:00 UTC

change the ars rule as below
matching:
To: sip:0094(11|31|77)([0-9]{7,})@

deploy:
To: sip:$2@Grandstream_ 4108_IP

sip:0094(11|31|77) means 0094 followed by 11 or 31 or 77
$2 in deploy means the content in second parenthesis in matching To filed, which is ([0-9]{7,}) part

You'd better learn how to write Regular expression.
http://wiki.brekeke.com/wiki/buffer-in-the-dial-plan
it is similar in ARS rule, but use $ instead of %
http://en.wikipedia.org/wiki/Regular_expression

tissa 2011-07-25 16:01 UTC

hi,
can you send me ARS settings for 10 digit subscriber numbers.
my current ARS settings

ARS
matching
sip:([0-9]{7,25})@
deploy:
sip:$1@Grandstream 4108_IP

thanks

hope 2011-07-25 17:54 UTC

http://rubular.com/

tissa 2012-04-12 13:17 UTC

How to give DID Number to ARS?

hope 2012-04-12 18:02 UTC

do you need on inbound calls or outbound calls?
give some example about your question is better.

tissa 2012-04-13 13:09 UTC

What I meant to say is
- I have 2 Quintum gateways configured to Brekeke PBX server for call originate and termination.
- So, I'm using DID number for calling card user to access.

What I want to know is how to connect to originate gateway without IP PBX.

I hope this will elaborate you more on my inquiry.

Thanks in Advance,
Tissa

hope 2012-04-13 19:29 UTC

who issue the DID number?
is there any setting at DID provider side to point each DID number to gateway
and set gateway to send calls to Brekeke PBX.

tissa 2012-08-27 12:33 UTC

how to give DID Number to ARS. as Inbound calls.My DID Provider is DIDWW.Provider Settings are

[46.19.209.10]
host=46.19.209.10
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.11]
host=46.19.209.11
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.12]
host=46.19.209.12
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.13]
host=46.19.209.13
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.14]
host=46.19.209.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.15]
host=46.19.209.15
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.75]
host=46.19.209.75
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.76]
host=46.19.209.76
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.77]
host=46.19.209.77
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.78]
host=46.19.209.78
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.79]
host=46.19.209.79
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.209.80]
host=46.19.209.80
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.10]
host=46.19.210.10
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.11]
host=46.19.210.11
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.12]
host=46.19.210.12
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.13]
host=46.19.210.13
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.14]
host=46.19.210.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.15]
host=46.19.210.15
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.75]
host=46.19.210.75
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.76]
host=46.19.210.76
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.77]
host=46.19.210.77
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.78]
host=46.19.210.78
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.79]
host=46.19.210.79
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

[46.19.210.80]
host=46.19.210.80
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-didww
insecure=very
nat=never
allow=all

please advise me.
thank you

hope 2012-08-27 18:12 UTC

How many DID numbers do you need to set in ARS?
do you need to register to provider?

tissa 2012-08-28 00:22 UTC

one number,yes i need register to provider

tissa 2012-08-31 01:38 UTC

one number,yes i need register to provider
thanks

tissa 2012-09-10 16:40 UTC

Hi,

Appreciate your quick reply on following;

When a call is made and hung up ,it takes few mins (3-4 mins) for session to get time out.So during that period no calls can be made.Since the time out period is bit long pls let me know sort out this problem?

Thanks

hope 2012-09-10 17:37 UTC

- what is call status shown from sip server side / Active sessions?
- is the call going through ARS or just call between two pbx users?
- no call can be made to/from the same pbx user or no call can be made from/to any user?

tissa 2012-10-10 14:22 UTC

Hi All,

I have installed PBX version 3 . Can anyone pls let me know what are the accounting plug-ins required? I'm using Radius cat standard version,have used PBX Version 2 and configured accounting settings.However, it doesn't seems same settings working in PBX version 3.Please let me know if anything new plug-ins needs to be installed?

Thanks in Advance

tissa 2012-10-17 17:07 UTC

Hi All,

I have installed PBX trial version 3 . Can anyone pls let me know what are the accounting

plug-ins required? I'm using Radius cat standard version,have used PBX trial Version 2 and

configured accounting settings.The plugin- I installed are :

radius.acctport = 1813
radius.radiushost = 192.168.2.7
radius.sharedsecret = thiline
net.usrdir.plugins=com.sample.radius.proxy.RadiusAuth

radius.authport = 1812
radius.acctport = 1813
radius.radiushost = 192.168.2.7
radius.sharedsecret = thiline

dial Plan
Matching Pattern:
$request = ^INVITE

Deploy Pattern:
$session=com.sample.radius.proxy.RadiusAcct
$continue = true

However, it doesn't seems same settings working in PBX version 3.Please let me know if

anything new plug-ins needs to be installed?

Thanks in Advance

Mcgee 2012-10-17 19:22 UTC

Use plugin.radius.RadiusAcct instead of com.sample.radius.proxy.RadiusAcct with "$session"