please advise

tissa · Post by **tissa** » Wed May 18, 2011 5:32 am

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 · Post by **tissa** » Thu May 19, 2011 4:48 pm

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 · Post by **hope** » Thu May 19, 2011 5:10 pm

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

tissa · Post by **tissa** » Fri Jun 17, 2011 9:12 am

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 · Post by **hope** » Tue Jun 21, 2011 10:08 am

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

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

tissa · Post by **tissa** » Mon Jul 11, 2011 8:15 am

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 · Post by **hope** » Mon Jul 11, 2011 1:36 pm

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 · Post by **tissa** » Sat Jul 16, 2011 4:49 am

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 · Post by **hope** » Mon Jul 18, 2011 10:57 am

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 · Post by **tissa** » Tue Jul 19, 2011 7:49 am

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 · Post by **hope** » Tue Jul 19, 2011 10:00 am

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 · Post by **tissa** » Mon Jul 25, 2011 9:01 am

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 · Post by **hope** » Mon Jul 25, 2011 10:54 am

http://rubular.com/

tissa · Post by **tissa** » Thu Apr 12, 2012 6:17 am

How to give DID Number to ARS?

hope · Post by **hope** » Thu Apr 12, 2012 11:02 am

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

tissa · Post by **tissa** » Fri Apr 13, 2012 6:09 am

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 · Post by **hope** » Fri Apr 13, 2012 12:29 pm

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 · Post by **tissa** » Mon Aug 27, 2012 5:33 am

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 · Post by **hope** » Mon Aug 27, 2012 11:12 am

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

tissa · Post by **tissa** » Mon Aug 27, 2012 5:22 pm

one number,yes i need register to provider

tissa · Post by **tissa** » Thu Aug 30, 2012 6:38 pm

one number,yes i need register to provider
thanks

tissa · Post by **tissa** » Mon Sep 10, 2012 9:40 am

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 · Post by **hope** » Mon Sep 10, 2012 10:37 am

- 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 · Post by **tissa** » Wed Oct 10, 2012 7:22 am

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 · Post by **tissa** » Wed Oct 17, 2012 10:07 am

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

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DID number

DID REGISTER

Accounting

RADIUSCAT ACCT