Using PBX to just to route calls between systems

1. Brekeke Product Name and Version:
3.8.3
2. Java version:
8v161
3. OS type and the version:
W7
4. UA (phone), gateway or other hardware/software involved:
Jive PBX and BSS
5. Your problem:
I want to coordinate a Jive PBX with my BSS. It is a strange setup where I must use a client to communicate with Jive, I can't do it straight with the BSS because BSS can't register on a PBX.
So my idea is to use the Brekeke PBX to make this bridge between both systems.
All systems are in different networks/sites and therefore only accessible through internet:
Jive proxy/registrar, Jive outbound proxy, Jive terminals, Brekeke SIP Server and Brekeke PBX.
SIP signaling will happen between all elements involved, but RTP should flow directly between Brekeke SIP Server and the Jive terminals.
Brekeke PBX will register on Jive registrar.
Brekeke PBX will receive calls from Jive outbound proxy and relay them, without handling RTP, directly to the Brekeke SIP Server.
There will be no routing to internal network on Brekeke PBX for these calls, they should just happen on internet.
As an option, I may in the future relay also calls from BSS to Jive
What is the quickest and simplest approach?
ARS for Jive and in the OUT pattern use the Brekeke SIP Server IP address?

BR

Udo

Brekeke PBX bundles Brekeke SIP Server (BSS).
So you don't have to use an independent BSS in your scenario.

Unfortunately this is not an option. Different sites, different companies...

BR

Udo

Is Brekeke PBX located behind the Brekeke SIP Server?
Or is it located in the front of the Brekeke SIP Server?

BSS and Brekeke PBX are n different sites.
All elements are in different sites.

BR
Udo

My question is ...
Which is your deployment?

Jive -- Brekeke PBX -- Brekeke SIP Server
or
Jive -- Brekeke SIP Server -- Brekeke PBX

Sorry, I didn't realize it.
Jive receives registration requests from BPBX, and then sends calls to BPBX.
BPBX then sends the calls, which don't require further authentication (just trusted IP relationship), to BSS.

Therefore:
Jive <-> BPBX <-> BSS

> Jive <-> BPBX <-> BSS

It will work but my preference is like this.

Jive <-> BSS <-> BPBX

BSS's Upper Registration feature can forward BPBX's REGISTER to Jive.
With the above deployment, you don't have to write complicated DialPlan/ARS at BPBX.

Hello James,

No, this won't work because Jive has different servers, one for registrar and one for outbound proxy. To make BSS work with thru registration in this case it would require 2 levels of outbound proxy.
If this would work I actually wouldn't need BPBX, I could make the system that is behind BSS authenticate on jive.


BR

Udo

> ... because Jive has different servers, one for registrar and one for outbound proxy.

So the PBX needs to send REGISTER to Jive registrar, and send INVITE to outbound proxy?

It can be possible with BSS's DialPlan if a PBX (or any SIP client) behind the BSS.

Hello James,
You are actually making me think a bit different.
BSS is an entry point for a network of GSM gateways in VPN. So what would actually happen is that I need the outbound proxy mainly to receive the INVITEs.
And the registration is just to tell the Jive system to which IP address to send the calls.
Therefore, if I have ANY SIP client, even a phone, make the registration on the registrar, the authentication conditions will be met and I can use BSS's DialPlan to handle the calls just as if BSS did the authentication.
Just as you mentioned.
I will try this,
Thanks

Udo

It is what I suggested.

You can send REGISTER from any SIP client through the BSS (with Thru Registration function) to the Jive to inform the BSS's IP address.

You can catch Jive's INVITE with BSS's DialPlan. So you can route this INVITE to any destination.


Thru Registration is enabled in the default.
So let you configure a SIP client to send REGISTER to the Jive's registrar with the BSS as the outbound proxy.

And test you can receive a call sent from the Jive at the SIP client through the BSS.

If it works, add a DialPlan rule to catch Jive's INVITE.

Hello James,

Apparently it didn't work. Registration was denied because the realm was the realm of my BSS instead of their outbound proxy.
☹

Any other suggestion?
If not, I'll have to check BPBX...

BR

Udo

Which SIP UA client did you try?
Some SIP clients can handle this situation. I mean they can send re-REGISTER correctly even if both Outbound-proxy and Registrar send "407 Proxy Authentication Required" with unique realms.

To solve this issue quickly, disable REGISTER Authentication at Brekeke SIP Server. If you think this is a security risk, use $auth=off at DialPlan only for this SIP UA.

James,

I've tried a Snom IP phone, which works very nicely with other systems.

I will have to go back to BPBX, as it is the only way from their point of view to send the right REGISTER packet, as the realm must be their outbound proxy.

Now, I have already a BPBX set up on a Windows 2012 R2 machine, but for some reason it isn't sending out the INVITE when I program the ARS with user/password/registrar/outbound proxy.
Is it required to do the full BPBX configuration, including extensions, to make ARS start working? Or is just Windows Server tricky? Although I did shut down the firewall...
I have BSS running on W7 without any problem, so it would not be an issue for me to deploy the BPBX on another W7 machine.

What do you think?

BR

udo

> ... as it is the only way from their point of view to send the right REGISTER packet, as the realm must be their outbound proxy.

Does it send REGISTER to the Jive successfully?

> Now, I have already a BPBX set up on a Windows 2012 R2 machine, but for some reason it isn't sending out the INVITE

How did you try to send an INVITE from the Brekeek PBX?
Was it originated from an IP phone?

Yes, it sends the REGISTER correctly, then I receive two rejects. The second one shows authentication issues.

Correction in what I've written regarding the INVITE of BPBX:I meant REGISTER, not INVITE.
From the ARS screen I try to register, using the REGISTER button.
I keep wireshark running with the SIP filter on.
No SIP packet leaves the machine.

BR

Udo

> Yes, it sends the REGISTER correctly,

How did you observe the REGISTER? because you said "No SIP packet leaves the machine."...

Can you find any REGISTER attempt at [DialPlan]->[History] page?

The snom phone sends the REGISTER correctly, not the PBX.

Dial Plan history is empty

In Start/Stop page there is an event: registration on the Jive registrar failed due to Timeout.

This is the part I don't understand. How can there be a registration timeout if no registration package was sent?

BR

Udo

For sending REGISTER from Brekeke PBX to Jive registrar, you don't have to use another phone such as Snom because Brekeke PBX can send REGISTER by itself.

> In Start/Stop page there is an event: registration on the Jive registrar failed due to Timeout.

It seems you have a configuration issue at PBX's ARS settings.
Did you set the correct Jive registrar address there?

> For sending REGISTER from Brekeke PBX to Jive registrar, you don't have to use another phone such as Snom because Brekeke PBX can send REGISTER by itself.
Yes I know, but the snom phone sent the REGISTER when I was trying to make it work with BSS. This option doesn't work, I have to use BPBX.

I've double checked: I had copy/pasted the info to the ARS screen and also to the snom phone.
Besides, if it was bad registration info, I would have a packet leaving the machine. Yet I'm using Wireshark to control if there is any SIP packet leaving the machine. No SIP packet is laving the machine.

Could it be because it is a VM?

BR

Udo

Since DialPlan History shows nothing (Is it correct at the PBX's bundled SIP Server?), the PBX didn't send any REGISTER.

I suppose you missed some configuration at the PBX.

I've found something, which is possibly a bug.
I'm having to use an alternate SIP port for BPBX because BSS is already on port 5060 and I prefer to open the SIP port on the firewall instead of redirecting the port.
When I change the SIP server port to something different from 5060 (tried 5070 and 6060) and renew the registration of the UAs, the calls don't work any more. BPBX becomes a black hole. I see the INVITEs and the UA sits there, waiting for an answer that never comes.
Once I change the port back to 5060 the UAs start communicating between themselves.
And now I have a message on the ARS attempt to register:
02-13 15:41:01: Register failed (Timeout) - user01@reg.jiveip.net
02-13 15:41:07: Register failed (408 SIP/2.0 408 Request Timeout) - user01@reg.jiveip.net

This time it is ok, because the port 5060 is directed to BSS, so I will always get a Timeout

BR

Udo

How did you change the PBX's port??
Do you mean you changed the port number of PBX's bundled SIP Server?

Correct.
The SIP port must not be changed from 5060

Are they on the same Windows running on the VM?

Different Windows VMs, but same internet link.
That is why I need the different SIP ports.

Are there any SIP UA clients such as Snom connecting to this bundled SIP Server running under Brekeke PBX?
If so, did you change the proxy settings at UAs to point new SIP server's port?

Yes, the Snom and the Phoner Lite UAs are connecting to the proxy and whenever I changed the SIP port I had to change it in the configurations of the UAs to make them register.
Whatever the SIP port used (5050, 5060, 5070, 5080, 6060), the UAs register on the bundled SIP Server. However, I can only establish the calls between UAs when the port is 5060.

Hello James,
I have now the BPBX on its own link and the ARS is registering.
Now I'm back to the original topic:
Depending on the route and which originating number within the ARS I want to:
a) Modify the originating number to make it unique in the system (Patterns IN -> Deploy patterns -> From -> Modify number)
b) Pass this call to another ARS (it is an ARS where I want to combine several gateways with different sessions capabilities)

Is this possible?

Or should I just use for each of this routes just a gateway configuration in the Patterns - OUT?

BR

Udo

There are several ways. I recommend that you request some advice from Brekeke's support team.