Hi, On Mon, 2005-01-03 at 16:36, [EMAIL PROTECTED] wrote: > with * through ISDN PRI. Because of that transfer function is handled on > legacy PBX (Alcatel) and Asterisk does not 'know' if agent talks to callee > or if I transfer incoming call. Do you using some other PBX connected to > Asterisk PBX? That may be the case.
Thanks for the suggestion, but no, that is not the case. As you can see below, the SIP phone is not shown in the agents list, but the channel that a call was transferred to is. Confusing... BTW: This is how it looks in idle state: 101 (TIC 1) available at '[EMAIL PROTECTED]' (musiconhold is 'default') > > I have a queue with agents that log in using agentcallbacklogin. The > > extension that is logged in with is a Local channel. Now, if a call > > comes in to the queue and is handled by an agent (in our case using > > Cisco 7960 SIP phones) and transferred (attended) to another extension, > > the agent remains unavailable during the remains of the call. Using show > > agents gives this: > > > > 103 (TIC 3) logged in on MGCP/aaln/[EMAIL PROTECTED] talking to > > Zap/20-1 (musiconhold is 'default') Best regards, Florian _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
