Leonardo Gomes Figueira
2006-Jun-21 12:16 UTC
[Asterisk-Users] Agent channel X SIP Transfer on 1.2.9.1
Hi, I wonder if on Asterisk 1.2.X calls from queue answered by Agent channel still must be transfered only by Asterisk internal transfer (features) like on 1.0.X ? The wiki says on http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Queue "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being offered another call." But I did some tests and on 1.2.7.1 calls can be transfered via SIP transfers and the agents don't get locked. It's working fine with SIP but with IAX2 the agent channel gets locked until the transfered call is dropped. On 1.2.9.1 the call is dropped after the transfer. If the call is transfered to a Polycom IP300 the caller is dropped too but the called (Polycom) is not dropped (but there is no audio of course) and if I run a "show channels" on cli Asterisk segfaults. I'm using 1.2.7.1 on production PBXs now cause 1.2.9.1 has bugs in queues/agents (I will post them soon) but what I really wanna know is what will be the standard to Agents transfer ? SIP transfers can be used or I must use attended transfer from features ? What about IAX2 transfers ? Leonardo