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

[Paul Belanger]

On 11-07-23 01:18 PM, bilal ghayyad wrote:
> 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