Creating Outbound and Inbound Dial Pattern to PSTN

1. Brekeke Product Name and version:

2. Java version:JDK6

3. OS type and the version:Windows Server 2003

4. UA (phone), gateway or other hardware/software involved:xlite and grandstream HT503

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :Pattern 2

6. Your problem:How do i configure my dial plan to allow outbound and inbound calls to PSTN number e.g 08052309023. Below are the dial plans i have currently working
Matching Pattern :
$request=^INVITE
To=sip:(.+)@

Deploy Patterns :
$auth=false
To=sip:%1@192.168.12.229
$continue=false

For your Outbound Dial Plan:

Matching Pattern :
$request=^INVITE
To=sip:0(.+)@

Deploy Pattern :
$auth=false
To=sip:%1@192.168.12.229


1) Currently the INBOUND Calls to our network works like this:
(a) When a mobile (PSTN) calls into the network our sip phone rings but at the PSTN callers end it does not indicating that it is ringing at our end, it just keeps showing calling. Also when the PSTN caller ends the call the sip phone keeps ringing for about 35sec then stops.

(b) When a sip phone answers an incoming call from a PSTN caller there is short silence at the sip users end then the call drops, while at the PSTN callers end the mobile phone still shows calling no indication that the sip phone rang and was answered.


2) The following was my observation for OUTBOUND calls:
(a)When a sip phone make a call out to a PSTN line it shows calling for some second the call fails. At the switch session it show inviting then the call ends.

3)Do i have to create IN and OUT patterns in the PBX ASR. If this is needed how do i create the i and out patterns in the ASR The sip server ip=10.1.13.1 and Gateways IP=192.168.12.229

which version PBX are you using?
what Interface IP address can you see from first page after log in pbx?
only 10.1.13.1 ?
brekeke should be in the same ip range as gateway?

when call from gateway to pbx, what dialing number is sent to pbx/sip server? is it 08052309023 or a registered user number at pbx/sip server?

Thanks for replying:

1)which version PBX are you using?

I am using PBX/Server: 2.3.6.0/286


2)what Interface IP address can you see from first page after log in pbx?
only 10.1.13.1 ?
brekeke should be in the same ip range as gateway?

I have 3 IPs on my NIC and they are all showing at the page after logging on to the PBX : 10.1.13.1, 10.1.13.5, 41.189.X.X Brekeke's IP is 10.1.13.1


3)when call from gateway to pbx, what dialing number is sent to pbx/sip server? is it 08052309023 or a registered user number at pbx/sip server?

On the Active Session page it shows:

FROM: 08052309023@129.168.12.229 (192.168.12.229:5060)

TO: 016264002@10.1.13.1 (127.0.0.1:5060)

STATUS: Closing

is number 016264002 a register user phone number at brekeke sip server?
and add an ip 192.168.12.* at brekeke server NIC

1) Is number 016264002 a register user phone number at brekeke sip server?

Yes 016264002 is a User phone number registered to the SIP Server

I have add 192.168.12.* to Brekeke NIC

after logging into pbx, there are four IP shown?
after logging on to the PBX :192.168.12.X, 10.1.13.1, 10.1.13.5, 41.189.X.X

Add the following dial plan, apply rule after add.
Matching Pattern :
$request=^INVITE
$addr = 192.168.12.229
To=sip:(.+)@

Deploy Patterns :
$auth=false
To=sip:%1@

and you can try call from 08052309023 to 016264002.
can the call be established?

Thanks for your reply sorry i wasnt able to reply for the past few days. I can now make inbound and outbound calls to and from within SIP Server and PSTN.

I took out the Public IP because i noticed using wireshark the Public IP was confusing the Private IP. Now my problem is how do i connect to my 2nd Exchange House now that i have taken out the Public IP. Just a reminder i 'm connected to 2 Exchange Houses:
a) the 1st by Private IP
b) the 2nd by Public IP

I need to have the 2nd Exchange some how connected to the Same SIP Server for failover calls is there a way to do this ?(any of the exchange can be the failover 1st or 2nd). Will be looking forward to your response.

Thank you

is your pbx server pc behind router?
how many NIC on the pbx server pc?

Hi Hope,

1) No i'm not behind a router.

2) The PBX Server has 2 NIC.

Would i be needing a router, please advice.

Thanks

MJ

what kind of problem do you have when public ip is used?
what calls go through sip server private ip? the calls from gateway?
what calls go through sip server public ip?

1)what kind of problem do you have when public ip is used?

The calls dont go through the session status just shows "inviting" status until i end the call.

2)what calls go through sip server private ip? the calls from gateway?
a)Currently only the private IP calls go out since i took out the public IP

b) Call to and from the gateway are working okay

3) The public ip and private ip trunks should act as failover for the each other. The public IP should be able to receive and send out calls just like the private IP

Thank you

1)what kind of problem do you have when public ip is used?

The calls dont go through the session status just shows "inviting" status until i end the call.

are both caller and callee registered at sip server?
what ip address of the caller and callee?
is there any dial plan used for the call?
and what dial plans currently set at sip server/dial plan?

Hello,

1)are both caller and callee registered at sip server?
ans: No just the caller is registered to the system

2)what ip address of the caller and callee?
ans: The callers ip add is a public ip callee's ip could be static or DHCP.

3)is there any dial plan used for the call? and what dial plans currently set at sip server/dial plan?

ans:
Matching Pattern

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

Deploy Patterns

$auth=true
$b2bua=false
To=sip:%1@62.173.x.x
$continue=false

I notice the above dial pattern works for inbound and outbound call to the server when the SIP Server has private ip but when the server is configured with a public ip i cannot make outbound calls but can receive calls with the following dial pattern:

Matching Pattern

$request=^INVITE
To=sip:(01.+)@

Deploy Patterns

To=sip:%1@
$continue=false

how about if set the private IP default gateway the same as the private IP?
Can the call be sent to 62.173.x.x when enable both private and public ip on pbx server?