please advise

tissa · Post by **tissa** » Wed Jan 19, 2011 8:15 am

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 · Post by **hope** » Wed Jan 19, 2011 11:04 am

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 · Post by **tissa** » Fri Feb 25, 2011 11:19 pm

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 · Post by **hope** » Mon Feb 28, 2011 11:19 am

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 · Post by **tissa** » Tue Mar 01, 2011 9:53 am

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 · Post by **hope** » Tue Mar 01, 2011 11:17 am

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 · Post by **tissa** » Tue Mar 08, 2011 8:09 am

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 · Post by **hope** » Tue Mar 08, 2011 12:20 pm

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 · Post by **tissa** » Fri Mar 11, 2011 9:10 am

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 · Post by **hope** » Fri Mar 11, 2011 1:45 pm

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 · Post by **tissa** » Thu Apr 07, 2011 12:48 pm

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 · Post by **hope** » Thu Apr 07, 2011 5:49 pm

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 · Post by **tissa** » Fri Apr 08, 2011 9:07 am

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 · Post by **hope** » Mon Apr 11, 2011 10:35 am

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 · Post by **tissa** » Sun Apr 17, 2011 4:14 pm

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 · Post by **hope** » Mon Apr 18, 2011 4:42 pm

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 · Post by **tissa** » Fri May 06, 2011 9:37 am

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 · Post by **hope** » Mon May 09, 2011 11:10 am

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 · Post by **tissa** » Mon May 09, 2011 4:24 pm

it use Brekeke pbx/sip server

tissa · Post by **tissa** » Mon May 09, 2011 4:34 pm

yes, it use brekeke pbx/sip server

To= sip00(.+)@

hope · Post by **hope** » Tue May 10, 2011 10:46 am

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 · Post by **tissa** » Wed May 11, 2011 10:06 am

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

hope · Post by **hope** » Wed May 11, 2011 10:38 am

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 · Post by **tissa** » Tue May 17, 2011 6:15 am

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 · Post by **james** » Tue May 17, 2011 10:17 am

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.

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