you can configure authentication at registration time in your sip server, this prevents unwanted users using your sip server. go to sip server -> configuration -> sip(general) -> authentication register: on. this means you need to create users with passwords for the users you allow to register with ...

1. Brekeke Product Name and version: 2.0 sip server 2. Java version: 1.5 3. OS type and the version: win 2003 4. UA (phone), gateway or other hardware/software involved: 5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html : 6. Your problem: I need to send ...

If you have authentication enabled on your sipserver, your UAs will not be able to register at sipserver, if you didn't create your users at sipserver.

If you don't use authentication, creation of user on pbx is enough.

Pablo.

Hello Nick,


What you are saying is that when sip user calls other sip user, they can't use pbx funcionality, like transfer call to other user?

Or this is the case when the corresponding pbx user doesn't exist?

thanks, Pablo.

Hello Manik Gupta,

The newest beta version of sip server seems to handle loops better.

Good luck,

Pablo.

Hello Michelle, Did you create the ext under pbx user? Did you create the ext under authentication of sip-server? Hello Nick, default dialplan on sipserver makes sure all calls are going thru pbx. unless a dialplan rule was added before these default diaplan rules and there was a succesfull match ...

Hello Augusto,

pls try this dialplan on sipserver:

first rule:
matching pattern:
request=^INVITE
from=sip:8400@
to=sip:8401@

deploy pattern
empty

sec rule:
matching pattern:
request=^INVITE
to=sip:8401@

deploy pattern
$action=603

good luck,

Pablo.

Hello Paul, Maybe you can archive your goal by using the java plugin, to make your own call fork program. When an invite is sent: lookup in the user database the second entry of the number you want to call and send an invite to this one as well, but make sure you don't get into a loop. Maybe Brekeke ...

Shakir,

with brekeke pbx you should be able to do your tests:

phone1 -> pbx1 -> other proxy -> pbx1 -> phone2.

Regards,

Pablo.

why not configure like this:

phone1 -> sipserver -> phone2 ?

Pablo.

You can run softphone and sipserver on the same computer.

there is a setting on the sipserver - configuration - rtp relay (UA on this machine) which you can turn off or on as you wish.

good luck,

Pablo.

Hello shakir,

I'm not sure if I understand your problem:

phone1 -> brekeke sip server1 -> other proxy -> brekeke sip server1 -> phone2?

Is this what you want?

Regards,

Pablo.

Hello Paul,

What you want is really a pbx feature which is support by brekeke pbx: ring groups.

the way sip works and how brekeke handles the registration it will never work as you suggested. 2 AU's with same userid/pwd.

Brekeke will always use the last registered one.

Pablo.

As I remember well (according posts in the old forums), Brekeke doesn't support forking.

Pablo.

posting you dialplans/ARS here could help to solve your problem.

Pablo.

There is an old forum which contains several posts about this subject.

Pablo.

hello

The dialplan document provided by Brekeke gives also some examples how to change expire time:

http://www.brekeke-sip.com/download/bss ... lan_en.pdf

or

http://www.brekeke-sip.com/download/oss ... plan_e.pdf
for version 1.5

good luck,

Pablo.

Hello Sin, let me see if I understand your problem: you hace some sipdevices registered with OSS one of them is the trixbox. from trixbox you can call to a sipphone registered on oss. using a sipphone calling 6xxxx gives you a 403 or 404 error in oss log, while this call should be routed to trixbox ...

Hello Sin,

the value between first () that appears in the matching pattern is %1
the value between the sec () that appears in the matching pattern is %2
ect...

Regards,

Pablo.

Hello job,

There aren't 1.x upgrade versions available on brekeke website any more.


Pablo.

pls check this post on the forum:

http://www.brekeke-sip.com/bbs/viewtopic.php?t=1309

Pablo.

Hello,

why not keep it simple:

Matching patterns:
$request=^INVITE
$registered=true


Deploy Patterns:
$auth=false
$continue=on (if necesary)

hope it works,

Pablo.

Hello,

Did you configure port 5069 on your remote device?
you can install other sip product on the same machine.

which network pattern are you using?
which gateways/uas are involved?

these questions helps to understand your problem a lillte better.

regards,

Pablo.

Hello,

Deploy Patterns
To=%1@%2

%1=(.+)
%2 = "8000"

I guess you want to have:
depoly pattern
To=%1@%3

maybe this helps.

Pablo.

Hello,

Normally you change this parameter at the user agent side.
What user agent do you use?

Pablo.