Basic routing through a gateway from hardphones

1. Brekeke Product Name and version:
SIP server 2.1.6.6 std

2. Java version:
JRE 1.6.0_13-b03

3. OS type and the version:
Win svr 2003

4. UA (phone), gateway or other hardware/software involved:
Thomson ST2030 hard phones
Dialogic media gateway

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


6. Your problem:
I need help developing some basic dial plans - I want to dialout through my media gateway that is connected to my E1, from my SIP hard phones. I have set a basic dial plan in Brekeke as per the dial plan tutorial - but when I dial a number from the hard phone it does not appear in the brekeke log - the only numbers I can dial from the hard phone are the other numbers that are registered on the sip server - how can I ensure that the hard phone is going to brekeke and then processing the dial plan

$request=^INVITE
To=SIP:(8.+)@

To=sip:8%1@10.9.16.4

10.9.16.4 being my media gateway, and 8 being the prifix i want to use

do you want the prefix 8 sent to gateway?
with your dialplan, when dialing number 812345, number 8812345 will be sent to gateway.

Matching Patterns:
To=sip:([0-9]{7,25})@

Deploy Patterns:
To=sip:$1@10.9.16.4

with above dialplan, when dialing number has 7 to 25 digits, the call will be sent to gateway

many thanks for you response.

However I still have not solved my issue - I have implemented the dialplan you suggested with a minor change {4,25} as I want to route 4 digit extention through the gateway as well.

What I find strange is that when I make a call from one sip station to another say 102 -> 107 a line appears in the brekeke log indicating the call as a success.
However when I dial 8391 for example (an extention on the pbx that the gateway is connected to) there is nothing in the log - as if it is not attmpeting to use brekeke ???

I know the connectivity is available as when I program phone book entries into the Thomson sip phone for example Xavier = 8391@10.9.16.4 it will dial through correctly (this bypasses brekeke) directly through the gateway and I see the call in the dialogic gateway log.

But when I dial 8391 from the keypad on the thomson there is no entry in the brekeke log or the dialogic log.

If my phone is set to use brekeke as its registrar and proxy with authentication I was assuming that all the call handling / routing is done by brekeke ? but the lack of activity in the log suggests it is not getting as far as breke ????

sorry,
Deploy Patterns should be %1 in dial plan, not $1, which is for ARS
To=sip:%1@10.9.16.4

many thanks that is working great.

How do we set number of call to particular destination IP/Gateway?

what number would you like to set for the call to particular destination IP/Gateway? caller ID number or dailing number?
could you give more details, an example is better

Hi hope..it is number of calls limitation.


e.g. I just want to make sure that at any time the number of calls send to one gateway IP address is 8 calls.

I think i got the answer after browsing the forum...At one thread someone mentioned that there is no session limit on a particular UA. which mean 1 concurrent call per 1 UA

Let say I have this dial plan

Matching Patterns

$request = ^INVITE
To = sip:(65.+)@

Deploy Patterns

To = sip:%1@UA001


So if the call is to 65xxxxxx, the call will be sent to UA001 SIP UA..Assuming UA001 is 4 ports FXO SIP Gateway. How do I make sure that this UA001 will receive 4 concurrent SIP to PSTN calls ?

you can try to develop your own session plug-in to check the number of sessions used for a particular dial plan.
details at http://www.brekeke.com/download/download_sip_doc_en.php
Developer Guide- Accounting plug-in

otherwise, you need to use Brekeke PBX ARS, there is field of "Max sessions" to set the number of sessions for the rule.