302 moved temporarily
Moderator: Brekeke Support Team
302 moved temporarily
1. Brekeke Product Name and version: bss 2.2
2. Java version:1.5
3. OS type and the version:2003
4. UA (phone), gateway or other hardware/software involved:
grandstream gxe pbx, speech server 2007
5. Select your network pattern from : pattern 8
6. Your problem: getting a 302 moved temporarily
sip trunk to gxe works fine (in and out)
forwarded the sip trunk in bound to ext 2100
a sip peer on the gxe looks at prefix with "2" and forwards to BSS
BSS has a dial plan
$request=^INVITE
$registered=false
$outbound=false
To=sip:2(.+)@
$transport=TCP
$auth=false
&net.sip.transport.follow.request=true
To=sip:%1@MSSIPADDRESS
call gets to the MSS box, but after a trying from the MSS box back to BSS, it then sends a 302....which BSS sends to GXE, then the GXE sends back an Ack... and then nothing else happens (no call).
I previously had the Brekeke PBX/SIP proxy configred with my sip provider and calls went in and out from MSS with no issues. I now have another PBX that I need to use, so I just want to use the BSS for tcp to udp help with MSS. I'll have to assume that the dial plans for the pbx handled this 302??? what do I need to do in bss to handle this 302?
thoughts?
2. Java version:1.5
3. OS type and the version:2003
4. UA (phone), gateway or other hardware/software involved:
grandstream gxe pbx, speech server 2007
5. Select your network pattern from : pattern 8
6. Your problem: getting a 302 moved temporarily
sip trunk to gxe works fine (in and out)
forwarded the sip trunk in bound to ext 2100
a sip peer on the gxe looks at prefix with "2" and forwards to BSS
BSS has a dial plan
$request=^INVITE
$registered=false
$outbound=false
To=sip:2(.+)@
$transport=TCP
$auth=false
&net.sip.transport.follow.request=true
To=sip:%1@MSSIPADDRESS
call gets to the MSS box, but after a trying from the MSS box back to BSS, it then sends a 302....which BSS sends to GXE, then the GXE sends back an Ack... and then nothing else happens (no call).
I previously had the Brekeke PBX/SIP proxy configred with my sip provider and calls went in and out from MSS with no issues. I now have another PBX that I need to use, so I just want to use the BSS for tcp to udp help with MSS. I'll have to assume that the dial plans for the pbx handled this 302??? what do I need to do in bss to handle this 302?
thoughts?
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.
--------------------------
[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
--------------------------
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.
--------------------------
[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
--------------------------
Hi Andrey - sounds like you are familiar with the 302 returned by MSS.
I'm not sure but perhaps the 302 is an indicator that something is wrong with the configuration of my MSS. Is that possible? Even when I call it directly with a softphone, it returns to me a 302. In this case my softphone does not handle the message, aside from sending an ACK.
Can you elaborate further on specific details for creating a User in SIP Server?
thanks,
Karl
I'm not sure but perhaps the 302 is an indicator that something is wrong with the configuration of my MSS. Is that possible? Even when I call it directly with a softphone, it returns to me a 302. In this case my softphone does not handle the message, aside from sending an ACK.
Can you elaborate further on specific details for creating a User in SIP Server?
thanks,
Karl
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 details for creating a User in SIP Server?
You need to read "7. Basic Setup" in the document.
http://www.brekeke-sip.com/download/bss ... min_en.pdf
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 details for creating a User in SIP Server?
You need to read "7. Basic Setup" in the document.
http://www.brekeke-sip.com/download/bss ... min_en.pdf
Hi Andrey,
The 403 problem happens when the call is redirected to the PBX. In this case no communication occurs to the MSS server.
When I setup the BREKEKE SIP server to redirect the incoming calls directly to MSS, I get response code 302. This is not actually an error, it is just MSS instructing the peer to reconnect on a different port.
My config is as follows:
PSTN --> softswitch -> BREKEKE server -> MSS
(MSS on Brekeke currently on the same machine, using different ports of course)
When I test, I dial in directly via a phone number. This goes to the softswitch which only supports UDP based SIP. The purpose of the BREKEKE server was to server as the bridge between the softswitch and MSS. The softswitch is not registered with the BREKEKE PBX - they are just peers on the network.
My current dial plan rule does not work.
here's the rule I currently have in place that at least attempts talking to MSS:
inbound:
$port="5060";
$registered="false";
$outbound="false";
$request="^INVITE";
To="sip:9(.+)@";
outbound:
$transport="TCP";
$auth="false";
$b2bua="true";
&net.sip.transport.follow.request="true";To="sip:%1@192.168.9.194:7060",
Here is the data I get back from MSS with Message code 302:
CONTACT: <sip:494593206@192.168.9.194:1099;user=phone;transport=Tcp;maddr=127.0.0.1;x-mss-call-id=b1be0a8d-dda6457e-4ffff29-aac97bbf%40192.168.9.194>
By setting up users to authenticate in the Breke pbx or sip server, it just seems to me this is not going in the right direction. The softswitch doesn't register or authenticate with BREKEKE, which is why the "403 Forbidden" error occurs.
It seems to me that we must instruct BREKEKE SIP server to follow the 302 temporarily moved request, and we'd basically be up and running.
Thoughts on what to try next are appreciated...
-Karl
[/i]
The 403 problem happens when the call is redirected to the PBX. In this case no communication occurs to the MSS server.
When I setup the BREKEKE SIP server to redirect the incoming calls directly to MSS, I get response code 302. This is not actually an error, it is just MSS instructing the peer to reconnect on a different port.
My config is as follows:
PSTN --> softswitch -> BREKEKE server -> MSS
(MSS on Brekeke currently on the same machine, using different ports of course)
When I test, I dial in directly via a phone number. This goes to the softswitch which only supports UDP based SIP. The purpose of the BREKEKE server was to server as the bridge between the softswitch and MSS. The softswitch is not registered with the BREKEKE PBX - they are just peers on the network.
My current dial plan rule does not work.
here's the rule I currently have in place that at least attempts talking to MSS:
inbound:
$port="5060";
$registered="false";
$outbound="false";
$request="^INVITE";
To="sip:9(.+)@";
outbound:
$transport="TCP";
$auth="false";
$b2bua="true";
&net.sip.transport.follow.request="true";To="sip:%1@192.168.9.194:7060",
Here is the data I get back from MSS with Message code 302:
CONTACT: <sip:494593206@192.168.9.194:1099;user=phone;transport=Tcp;maddr=127.0.0.1;x-mss-call-id=b1be0a8d-dda6457e-4ffff29-aac97bbf%40192.168.9.194>
By setting up users to authenticate in the Breke pbx or sip server, it just seems to me this is not going in the right direction. The softswitch doesn't register or authenticate with BREKEKE, which is why the "403 Forbidden" error occurs.
It seems to me that we must instruct BREKEKE SIP server to follow the 302 temporarily moved request, and we'd basically be up and running.
Thoughts on what to try next are appreciated...
-Karl
[/i]