On 1/28/08, Thomas Kenner <[EMAIL PROTECTED]> wrote: > Hi, > > when I'm trying to call the following extension > > exten => 6002,1,Verbose(1|Extension 6002) > exten => 6002,n,Dial(Agent/6002) > exten => 6002,n,Hangup() > > the call is terminated and I get the following warning from asterisk: > > app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' > (cause 17 - User busy) > > When calling the agent with Dial(SIP/6002) no problem occurs. > > What could be wrong?
I never got this working, not sure why (wiki states that it should work). However Agent channel is considered obsolete - because of locking problems. You should consider using Local channels with GROUP_COUNT, and if you're using call queues, you would want to use this backported patch from 1.6. http://lists.digium.com/pipermail/asterisk-dev/2008-January/031545.html Regards, Atis > > > > Some additional information about the configuration: > > The asterisk version is 1.4.10 > > ----------------------------------------------------------------------------- > In users.conf I defined a user 6002: > > [6002] > fullname = Test Agent > email = [EMAIL PROTECTED] > secret = 1234 > zapchan = 1 > hasvoicemail = yes > vmsecret = 1234 > hassip = yes > hasiax = no > hash323 = no > hasmanager = no > callwaiting = no > context = international > host=dynamic > ----------------------------------------------------------------------------- > In agents.conf I added the agent > > agent => 6002,1234,Test Agent > ----------------------------------------------------------------------------- > and in queues.conf I added a queue testQueue2: > > [testQueue2] > music=default > strategy=ringall > timeout=15 > retry=5 > wrapuptime=0 > maxlen = 0 > announce-frequency = 0 > announce-holdtime = no > member => Agent/6002 > servicelevel = 60 > ----------------------------------------------------------------------------- > > > Thanks a lot, > Thomas > > -- > Thomas Kenner > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users