Dial Plan Setup Problems

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howard0320
Posts: 4
Joined: Thu Jul 03, 2008 6:47 pm

Dial Plan Setup Problems

Post by howard0320 »

1. Brekeke Product Name and version:
Version 2.1.6.6
2. Java version:

3. OS type and the version:
Window XP
4. UA (phone), gateway or other hardware/software involved:
X-lite
5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:
1. I can't unsderstand the Dial Plan Setup Document.Has any one can provide more information about this to me ? Thanks
2. Sip server IP: 192.168.1.1
Xlite IP : 192.168.1.2 (Ext 1002)
FXO Gateway IP: 192.168.1.3 (Ext 1003)

How can i setup PSTN to Ext and EXt to PSTN ?
We want to set up dail 9+ PSTN number.

PSTN---> FXO Gateway --->Sip Server----> Ext
Ext ----> Sip Server-----> FXO---> PSTN
Niggy
Posts: 86
Joined: Sun Oct 07, 2007 3:56 am
Location: Riyadh-ksa

hi

Post by Niggy »

Hi howard,

To do PSTN>>Ext call u have to configure ur FXO gateway baiscly u have to set up ur Brekeke server IP which is (192.168.1.1) & ur SIP user ID (ur FXO ID in brekeke)

To do Ext >> PSTN call u have to configure either ur PBX ARS or ur SIP server Dial Plan.

for the ARS:
add a new rule & jst configure the PATTERN - OUT

Matching Pattern:
TO sip:<ur out call prefix >(.*)@

Deploy Pattern:
TO sip:$1@192.168.1.3 >>(ur FXO ip)

try this out.. good luck.
Niggy
Posts: 86
Joined: Sun Oct 07, 2007 3:56 am
Location: Riyadh-ksa

hi

Post by Niggy »

if u want to configure ur Dial plan:

Matching Pattern:
$request=^INVITE
To = sip:9(.+)@

Deploy Pattern:
To= sip:%1@192.168.1.3

hope it'll work..
howard0320
Posts: 4
Joined: Thu Jul 03, 2008 6:47 pm

Re: hi

Post by howard0320 »

Niggy wrote:if u want to configure ur Dial plan:

Matching Pattern:
$request=^INVITE
To = sip:9(.+)@

Deploy Pattern:
To= sip:%1@192.168.1.3

hope it'll work..


Noted and OK now.
howard0320
Posts: 4
Joined: Thu Jul 03, 2008 6:47 pm

Re: hi

Post by howard0320 »

howard0320 wrote:
Niggy wrote:if u want to configure ur Dial plan:

Matching Pattern:
$request=^INVITE
To = sip:9(.+)@

Deploy Pattern:
To= sip:%1@192.168.1.3

hope it'll work..


Noted and OK now.

I have any others problems want to ask.
1. Do i need to set up Dial Plan When from Pstn to VOip ?
if yes, Also setup in ARS IN part ?

2. For Calling PSTN Problems.
UA-->ext FXO-->sound singal change-->PSTN No---Connected

Press ext FXO and wait 3-5 sec and Dail PSTN Number
Where can i set the waiting Time to 1-2 sec ?
Where can i set # for ending signal ?

3. I use xlite and hardware phone in offfice for press ext FXO and dial PSTN is ok, but when i use x-lite or hardware phone in non-office area, I just can dail ext Fxo and no any sound signal chnage. What is that problems ??

Thx

Howard Lam.
Mohney
Posts: 79
Joined: Tue Nov 20, 2007 12:05 pm

Post by Mohney »

Hi

>> 1. Do i need to set up Dial Plan When from Pstn to VOip ?

No. You don't have to.
As Niggy said, you just need to set the SIP Server's address as the proxy at the settings of the gateway for making a call from PSTN to VoIP.
See the gateway's document for more details.


>> Where can i set the waiting Time to 1-2 sec ?
>> Where can i set # for ending signal ?


It depends on gateway. See the gateway's document.


>> but when i use x-lite or hardware phone in non-office area, I just can dail ext Fxo and no any sound signal chnage. What is that problems ??

It seems NAT problem...
Set "rtp relay = on" at the SIP Server.
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