Brekeke not acting correctly when getting a BYE

1. Brekeke Product Name and version: 2.2.7.8

2. Java version: 6.0.26

3. OS type and the version: CentOS 5.5

4. UA (phone), gateway or other hardware/software involved: Asterisk 1.4.25

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

6. Your problem: 1 ad 9 (Brekeke sends call to clients PBX with a public IP).

When the Brekeke gets the BYE from the customers PBX it starts sending BYE's so the PBX. It should send a 200 OK. THE PBX getting the BYE is not expecting it and does not know what to do with it since it already sent the BYE. to the Brekeke.

I only get the call hung up correctly once LEG B (from the carrier's end) hangs up the call.

Here is a copy of the trace from the Brekeke server:

http://pastebin.com/Lp0155EZ

The issues seems to be that the Brekeke is being thrown off by the contact field in the 200 OK. I was going to change it in a new rule but it seems (from what I understand) I can only change what I am sending the PBX and not what I get from it.

do you have any reason to use version 2.2.7.8 ??

if you don't have a reason, use the latest version.

you can download it from http://www.brekeke.com/download/download_sip_2_0.php

and update it at current SIP Server's GUI.

James,

I updated to 2.4.5.5 and I still have the same issue.

Regards,

Dovid

do you have the same issue if you use another client instead of Asterisk??
Try X-Lite as a client for testing.

also.. do you have any DialPlan rules at SIP Server?
Is the SIP Server located behind NAT?
Which edition of SIP Server are you using??

i've looked at the packet trace.

Who is the SIP Server?
Is it 19.10.150.44??

James,

1) The IP's there are fake (I took out my real IP's). I believe that 19.10.150.44 is the IP of the switch. I would need to run the trace again to be sure.
2) The issue is Brekeke -> OpenSipS -> Asterisk. The issue is that when Asterisk returns the contact it sends it's IP. For instance.

Brekeke: 192.192.192.1
OpenSipS: 192.192.192.2
Asterisk: 192.192.192.3

The initial invite goes from the Brekeke to OpenSipS and OpenSipS passes it to Asterisk. The initial invite is to NUMBER@192.192.192.2. When Asterisk responds to sends a contact of NUMBER@192.192.3 which gets passed to OpenSipS and then back to the Brekeke. None of these servers are behind NAT. With the set up above there is an issue.

If I send the calls from Brekeke -> Asterisk there is no issue. The only difference is the contact so it seems it is a bug in the Brekeke Sip Server.

I can not understand your situation..
Is it problem in 200 OK's Contact header?

If you remove "OpenSipS", there are no problem... right?
Why do you need OpenSipS?


You can ask brekeke's team to analyze packets.

James,

The issue is that in the 200 OK the contact is of the endpoint and not the box that the Brekeke is talking to.

I need OpenSipS for load balancing and to send options to the two servers to make sure that they are up running (which as far as I know Brekeke does not do).

How do I contact Brekeke to analyze the packets ?

http://www.brekeke.com/buy/buy_techsupport.php
email to support@brekeke.com

Dovid,

> I need OpenSipS for load balancing and to send options to the two
> servers to make sure that they are up running (which as far as I
> know Brekeke does not do).


Brekeke can do them...

Load balancing:
http://www.brekeke.com/sip/sip-server_l ... ancing.php

Send OPTIONS to make sure that they are up running:
http://wiki.brekeke.com/wiki/Brekeke-SI ... tification

You should have two Brekeke SIP Servers if you want to do.