Hi all.<br><br>I'm trying to upgrade my production gateways with asterisk 1.4.17 and i'm facing some problems with attended transfer. Actually this happens since asterisk 1.4.13. Today i'm using asterisk 1.4.5 that showed to be the most stable release in our environment because all other versions locks about 30 minutes/1 hour of asterisk running time. But with version 1.4.5 it locks with channels reserved and everyday we have 2 or 3 lockups. There is a bug already open for that but the last update was made day 9. <br>
<br>Let me try to explain the problem. We are using asterisk as gateways for genesys predictive dialer.<br><br>A - Customer on Asterisk Gateway with TE412p<br>B - Operator on Genesys Sip Server<br>C - Closer on Avaya connected by E1 Tieline in Asterisk with TE412p in another facility. <br>
<br>This is the step-by-step<br>1 - Genesys dial a number through Asterisk<br>2 - A picks upGenesys delivers the call to an available agent<br>3 - B receives the call with A that has been routed by Genesys Sip Server<br>4 - A is put on hold by asterisk when B opens a second line with C to transfer the customer<br>
5 - B talks with C and transfer A to C<br>6 - Transfer is completed, B exit the call, but A keeps on music on hold and C listens to A music on hold<br><br>All signalling is made by Genesys Sip Server. This transfer works fine until asterisk 1.4.12 from 1.4.13 to 1.4.17 it behaves like described.<br>
<br>All files are attached in the open bug <a href="http://bugs.digium.com/view.php?id=11716">http://bugs.digium.com/view.php?id=11716</a><br><br>I trying to debug this issue but i can't manage to see where in the code is the problem, if any one could help me i would appreciate.<br>
<br><br>Regards<br>Matheus Rossato<br>