Hello Everyone,
In an attempt to figure out the best way to build my OpenSIPS config with the B2BUA module included, I've started over with a very simple script implementing nothing but the B2BUA module (and usrloc). My goal is to allow the phones to place calls between them and transfer the calls to other local phones (similar to a PBX). I have the B2BUA module configured to allow the call to be transferred (REFER) up to 4 times. This scenario script and OpenSIPS config file work well depending on which leg of the call is being transferred. It appears that the originator of the call cannot be put back on the opposite leg of the call, as shown in the second example above. Another way to think of the problem is that the remote side cannot be transferred. For both of the scenarios below, I have 3 phones. One is for the Sales department, one is for Tech Support, and the third represents the Boss. Here is a scenario where I can transfer the call four times successfully: 1. Sales places a call to Tech Support. (Sales has the "local" side of the call, and Tech Support has the "remote" side of the call.) 2. Sales transfers his leg of the call to the Boss. (The Boss has the "local" side of the call, and Tech Support has the "remote" side of the call.) 3. The Boss transfers his leg of the call back to Sales. (Sales has the "local" side of the call, and Tech Support has the "remote" side of the call.) 4. Sales transfers his leg of the call back to the Boss. (The Boss has the "local" side of the call, and Tech Support has the "remote" side of the call.) Here is a scenario where the process is interrupted: 1. Sales places a call to Tech Support. (Sales has the "local" side of the call, and Tech Support has the "remote" side of the call.) 2. Sales transfers his leg of the call to the Boss. (The Boss has the "local" side of the call, and Tech Support has the "remote" side of the call.) 3. Tech Support transfers his leg of the call back to Sales. (The Boss has the "local" side of the call, and Sales has the "remote" side of the call.) **THE CALL FAILS AT THIS POINT** - The Boss's phone is left on the call with nobody on the other end. Tech Support's phone appears to transfer successfully. Nothing happens with the Sales phone. My OpenSIPS Server is at 1.2.3.4 in this example, and the phones are on the 10.20.1.x subnet. There is no NAT between the devices. This appears to be no correlation between which devices are doing the transferring (meaning the Boss and Tech Support could switch roles and it still fails). I am running OpenSIPS 1.6.3 on CentOS 5.5 x64. I'll send my log file, config file, and scenario file in a moment (they were too big to fit in one email). Brett Woollum br...@woollum.com
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