[asterisk-users] The queue is not routing for the agent: returned -1:

[bilal ghayyad]

Dear Paul;

I got to know the reason for the problem, it is becaused I have to use SIP/bilal instead of SIP/599 as user configured as bilal in the sip.conf and not 599.

But I am still having some points that are really not clear and causing a problem:

1) If I need to login via the agent ID, so I am using the below line:

exten => 150,1,AddQueueMember(CustomerSupport,Agent/8001);
exten => 150,2,Playback(agent-loginok)

And when I dialed 150, I heared that Agent is logged in.
Now, when the call come to the queue, it is not sending for this Agent !! And it stay hearing the music, WHY?

Is it something related to the associatiation between the Agent ID and the extension? But as I know that the extension of the agent will be considered the same Phone that I logged from it (the same Phone that I dialed the 150 from it, correct)? Or I am wrong?

2) When I am login via the extension by using:

 exten => 150,1,AddQueueMember(CustomerSupport,SIP/bilal)

Then, if the user bilal already in call, another call is coming for it?! How to avoid this?

3) Where I have to use the Agent password? Until now did not see that I need to use the agent password, so why I am configuring it?

Regards
Bilal




--------------------
> > Hi All;
> >
> > Asterisk version is 1.8.4
> > The call is going to the queue but it is not routing
> to the agent which is logged in.
> >
> > I am afraid that I am missing a parameter to be set or
> enable or disable in the queues.conf so it is causing not to
> route for the interface.
> >
> > I am getting the following error messages:
> >
> > [Jul 24 18:45:02] WARNING[13177]: pbx.c:4893
> __ast_pbx_run: Don't know what to do with
> 'SIP/gwbilalhome-00000bfd'  == Using SIP RTP CoS mark
> 5
> >
> > [Jul 24 18:45:07] WARNING[13175]: acl.c:708
> ast_ouraddrfor: Cannot connect
> > [Jul 24 18:45:07] WARNING[13175]: chan_sip.c:3280
> __sip_xmit: sip_xmit of 0x7ff3 6c0d6380 (len 774) to :5060
> returned -1: Invalid argument
> >
> >
> > Below is my extensions.con
> >
> > [IncomingFromPSTN]
> >
> > exten =>  4,1,Goto(CustomerSupport,s,1)
> >
> > [QueueLogin]
> > exten => 
> 150,1,AddQueueMember(CustomerSupport,SIP/599);
> > exten =>  150,2,Playback(agent-loginok)
> > exten => 
> 151,1,RemoveQueueMember(CustomerSupport,SIP/599);
> > exten =>  151,2,Playback(agent-loggedoff)
> >
> >
> > [CustomerSupport]
> >
> > include =>  Internal
> >
> > exten =>  s,1,Queue(CustomerSupport,t,,,120)
> >
> > Now, there are also another two problems:
> >
> > 1) When I am dialing 150 to login, then after I hear
> the message that agent is logged in, then I hangup the SIP
> Phone, then I see the following message:
> >
> > [Jul 24 18:53:49] WARNING[13263]: pbx.c:4893
> __ast_pbx_run: Don't know what to do with
> 'SIP/gwbilalhome-00000c09'
> >
> > So, why this?
> >
> > 2) If I configured in the queues.conf the member to be
> in the queue (so no need to login, correct), if the memeber
> is Agent, then how the association will be between the Agent
> and the Extension? How it will be determined that the Agent
> 1003 will be existed in the extension 599 as I am not
> dialing anything from the Phone to login?
> >
> What does your sip.conf look like for [599]?  The
> problem is asterisk 
> (specifically app_dial, then netsock2.c) is converting
> 'SIP/599' to 
> 'SIP/0.0.2.87' and sending the INVITE to that address. This
> is a 
> regression introduced when IPv6 was added.
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org