need UDP -> TCP translator for MSS 2007 Speech Server

[karl]

1. Brekeke Product Name and version:
PBX v2.3.6

2. Java version:
1.6

3. OS type and the version:
Windows 2003 Server

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

A SIP gateway exists which sends the PSTN call to the following address: 192.168.9.194:5060. this is however in UDP and must be in TCP, as the speech server only support TCP.

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

I believe pattern #2.

6. Your problem:

I read the posts to setup a Dialplan entry to forward the UDP calls via TCP. I have the MSS server listening on port 6060 on the same machine. So the objective is to take the call and relay it directly to the speech server via TCP.

I took a dialplan entry from an earlier post :

[Matching Patterns:]
$port = 15062
$localhost = true
$registered = false
$outbound = false
$request = ^INVITE
To = sip:2(.+)@

[Deploy Patterns:]
$transport = TCP
$auth = false
&net.sip.transport.follow.request = true
To = sip:%1@MSSIPADDRESS

This does not work. No calls are being routed out from the PBX.

Currently I get the following error:
403 Forbidden

Although I have monkeyed with the configuration settings and in the process have experienced many other error types, including 486 and 407.

One piece of information the other posts did not mention was what do do about the other pre-existing dialplan entries, (leave them or disable them), and what order my new dialplan entry should be specified as. (priority 1?)

Also, should a "user" be entered into the system for the incoming call (949XXXXXXX) in either the SIP proxy or the PBX or both? If this is required, is it possible to disable authentication (have no password or check for password?).

Regards,
Karl G.

[peng]

Did you check the wiki?

http://wiki.brekeke.com/wiki/Brekeke_PB ... rosoft_OCS

[Andrey]

>> 403 Forbidden

It will happen if a caller doesn't send valid credential.

Add a user at SIP Server's [User Authentication] page.

Otherwise, you may need to disable an authentication at [Configuration] -> [SIP] -> [Authentication]. (it will be a security risk..)

[karl]

Peng,

I checked the WIKI, but it did not really help.

My config may be different, because we are not using OCS, but only MSS 2007 (a subcomponent of OCS).
Our setup is as follows:

PSTN -> softswitch (UDP) -> BREKEKE Product (TCP) -> MSS 2007

The softswitch is setup as peer to the BREKEKE product - they don't register with one another at this point.

The 403 forbidden continues to happen when I enter the dialplan rules in the Wiki. I don't have any registered users in the Brekeke product, which is the cause of this problem. Really I don't want any registered users - all we need is to redirect calls to the MSS 2007 server via TCP, as well as handle outbound calls going back the other way.

Do you know of a way to enter a dialplan rule that will force the Brekeke SIP Proxy to follow a 302 message and reconnect? If so, I believe this problem would be solved.

Regards,
Karl

[Andrey]

Try following DialPlan rules.

Rule: From PSTN to MSS
----------------------------------------------------------
[Matching Pattern]
$request = ^INVITE
$addr = <softswitch's IP address>
To = sip:(.+)@

[Deploy Pattern]
To = sip:%1@<MSS 2007's IP address and port>
$transport = tcp
$auth = false
&net.sip.transport.follow.request = true
----------------------------------------------------------


Rule: From MSS to PSTN
----------------------------------------------------------
[Matching Pattern]
$request = ^INVITE
$addr = <MSS 2007's IP address>
To = sip:(.+)@

[Deploy Pattern]
To = sip:%1@<softswitch's IP address and port>
$transport = udp
$auth = false
&net.sip.transport.follow.request = true
---------------------------------------------------------

Note that you need to remove your previous DialPlan rules at first.
Good luck.