It seems you solve the issue.

It will happen because the SIP client detects its global IP address for the firewall traversal. Tune the SIP client's settings to disable the public IP address detection. It will be STUN or ICE in the settings. If you capture packets at SIP Server's PC, are SIP packets exchanged with the public IP ...

If you push "#" after you dial a shorter string at a phone, does it solve the problem?

voicemail is one of PBX functions.
any reason you don't want to create a PBX user?

Mahima, INVITE SIP:6711@192.168.1.102 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.117:5060 Max-Forwards: 70 From: To: Call-ID: 14462@192.168.1.117 User-Agent: TotalView 1.0 CSeq: 3 INVITE Contact: Content-Type: application/sdp Content-Length: 0 Is it an actual packet? It seems there are no From/To/Contact ...

what is "softswitch "?
Does it support 302 response?

Do you have any ARS settings at PBX for connecting to the MSS?

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 ...

>> 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..)

Hi Karl, Did you solve the 403 problem? The response code 302 means "Moved Temporarily". (Refer RFC3261 http://www.ietf.org/rfc/rfc3261.txt) It seems the MSS requested a caller to make a call to new destination. What kind of SIP softphone are you using? >> Can you elaborate further on specific ...

Hi karl,

Brekeke SIP Server should work with MSS.

>> I am now getting a 403 Forbidden error. Any suggestions?

You need to create a user in Brekeke SIP Server's [User Authentication] page.

Colin
why do you want to return to a previous version?

Which Radius feature do you want to use??
Is it Authentication? or Accounting?

Can you paste an INVITE packet which the gateway sent?
We need to check what kind of INVITE the gateway sends.

If you dial an unregistered device from a registered device, does the SIP Server forward the call to the ISDN gateway by the DialPlan?

Try "$registered(To)=false" instead of "$registered=false" in the DialPlan.

I think some of your settings are wrong..

If you disable the above DialPlan rule, can you make calls between registered users without problems?

What kind of UA are you using?

>> The problem is that apparently the SIP server most of the times doesn't recognize when a voip device is registered ...

Can you see the user's record in the [Registered Clients] page at that time?

which part of dial plan you can not understand ?
Regular Expression ?
http://www.google.com/search?q=Regular+Expression

think it simple.. let you try the dialplan below.. you don't need any other dialplan rules.. --------------------------------- Matching Patterns $request = ^INVITE $To= sip:9(.+)@ $registered = false Deploy Patterns To = sip:9%1@xxx.xxx.xxx.xxx (Gateway's IP address ...

Hi

Use $registered in Matching Pattern.
If it returns "true", the UAS is registered.

Anyway.. OSS version 1.5 is super old version.
Try the latest Brekeke SIP Server.
http://www.brekeke.com/download/download_sip_2_0.php

Contact SVK.. they have a billing solution for Brekeke.
http://www.svksoftware.com/

Contact them..

Voxbone:
http://www.voxbone.com/

DIDX:
http://www.didx.net/

Right.

If you want to accept a speech server's 302, you need to use Brekeke PBX.

Brekeke SIP Server is a proxy. it just forwards the 302 response to the UAC. But Brekeke PBX is a PBX. it accepts and handles the 302 response. So please try Brekeke PBX. If you are already using Brekeke PBX in this case, you need to modify your dialplan rule as the following ...