Trixbox and OSS

1. Brekeke Product Name and version: 1.5

2. Java version: 1.5

3. OS type and the version: Redhat Linux 9.0

4. UA (phone), gateway or other hardware/software involved:

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : 4

6. Your problem:

Dear Support Team,

We have one OSS and Trixbox Freepbx. OSS is in the public and Freepbx is in the private network. The Freepbx was registered to the OSS and we can call the destination number from PBX to OSS successfully. However, When we called the destination number from the OSS to the PBX, we received the error code 404 (Not Found) in the OSS. Is it any special dialplain we need to set in OSS? Please advice.

Matching Patterns
$request=^INVITE
To=sip:6(.+)@
$registeredaddr("8000")=(.*)

Deploy Patterns
To=%1@%2

Please noted: 8000 is our username in the pbx system.


Best Regards
Sin
03.04.2007

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 Pablo,

Thank you for your advice.

Could you tell me the meaning of %1, %2, %3 in the deploy pattern.

Thanks & Best Regards
Sin
15.04.2007

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 Pablo,

Thank you for your information again.

I changed to the %3, but the problem could not be solved (Error code=403). Do you have another suggestion for me?

Thanks & Best Regards
Sin
16.04.2007

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.

but your trixbox is behind a dyn. public ip address.

I would try to use this dialplan:

$request=^INVITE
To=sip:6(.+)@

Deploy Patterns
To=8000@

there is a dialplan tutorial available which can give you some ideas:

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

use ethereal to make some traces on your oss to see if sip packets appear as expected.

good luck,

Pablo.

Easy solution is:
- Create SIP User in OSS.
- Create SIP TRUNK from TRIXBOX.
- Create OUTBOUND ROUTES rule from TRIXBOX for outgoing call to OSS.
- Create INBOUND ROUTES rule from TRIXBOX for INCOMMING call from OSS.
- Create Dial plan for SIP user in OSS.

That's all.