[Freeswitch-users] Some question about mod_fifo ??

[Nguyễn Mạnh Hùng]

Hi Anthony.

What is version of FS has this feature of mod_fifo. I have update the latest
version in svn: FreeSWITCH Version 1.0.trunk (16972M).

I capture packets in an agent of mod_fifo with Wireshark, but i can't see any
SIP UPDATE or SIP INFO packet.

Best Regards.

 

Anthony Minessale [anthony.minessale at gmail.com]

its done by SIP UPDATE on polycom/aastra or sip INFO packets on snom when the
call is bridged.
X-lite does not update anything when it receives them.

That's about it.

 


 

     

    Hi Anthony,

    

    Can you discuss some details in how polycom or snom can do this and x-lite
    not.

    

    If can, I want to edit some open source soffphone like officeSIP to do this.
   

    

    Best Regards.

    

     

    Anthony Minessale [ anthony.minessale at gmail.com ]

    

    We already do it.
    X-Lite does not support it.
    If you try it with a phone like snom or polycom you will see it works just
    like that.


     


     

         

        Hi Seven Du.

        

        Thanks to yours suggetion.

        

        I have an ideal, it is: when the call between caller and agent is set,
        the caller_id is determined. So, i want to edit code to sent the agent
        information (the call_id and call_id_number) which will be displayed
        againt in the agent's softphone (as Xlite..) when the call is happening.
       

        

        I read some documents but i still can't determine: It's maybe yes or
        maybe to do this and where to do this.

        

        Can you give me some comments.

        

        Best Regard.

        

         

        Seven Du [ dujinfang at gmail.com ]

        

        

        »¿As discussed in the list, it's not a freeswitch problem but a
        reality of life. 



        Think about customer A and B calls in one after another, then if
        FreeSWITCH call agent X with caller id A and Y with caller id B, and
        angent Y answers before X, then

        1) if bridge Y with A with the FIFO rule, then the caller id is wrong
        2) if bridge Y with B, the caller id is right but it breaks the rule
        of FIFO - A should be served before B!! And what even worse is that
        if X never answer A then A never can be served which is really
        unfair!!

        Of course you don't want 1), and you don't need mod_fifo if you want
        behavior 2), you just need some dialplan trick or some simple Lua
        script I think. Also FreeSWITCH is designed to be easily extended with
        almost any languages so feel free to implement anything.


        > Hi Mike and Seven Du.
        > Thanks to yours help.
        > I known the mechanism of mod_fifo.
        >>>
       
       
       
       
       
       
       
       
        http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html
        .
        > What a pity, It can't solve this problem. I can't use freeswitch for
        my call
        > center.
        > Hope new version can solve this !!!


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