[Asterisk-Users] SIP transfers of queued calls doesn't make agentavailable

[Mark Roeten]

I'm using the asterisk transfer facility (#). The agent dials #00612345678.
The extensions.conf looks like this: 

 

exten => 00612345678,1,Dial(SIP/0612345678 at host.com)

exten => 00612345678,2,Congestion

 

Any ideas? 

 

  _____  

Van: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] Namens picciuX
Verzonden: woensdag 19 april 2006 11:40
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] SIP transfers of queued calls doesn't make
agentavailable

 

...mhm... are you sure the agent transfers the call using "asterisk
transfer" facility (ie, not using the "transfer" softkey of the phone)?

If this is the case, the agent should become free after the transfer...

 

2006/4/19, Mark Roeten <mroeten at tintel.nl>: 

Hello,

 

I have the following situation: 

 

-          Someone dials in and enters a queue;

-          Agent 1000 answers the call using a cisco 7912 phone;

-          Agent 1000 transfers the call using # to a external number (e.g.
mobile phone);

-          The caller is now talking directly to the mobile phone, agent
1000 is no longer needed;

 

The problem now is that Agent 1000 remains busy until the transferred call
ends. The situation I want to create is that the agent is available directly
after the transfer. Any thoughts how to solve this problem? 

 

Regards,

 

Mark


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