[Freeswitch-users] Some question about mod_fifo ??

[Anthony Minessale]

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.


2010/4/4 Nguyễn Mạnh Hùng <hungngm at bkav.com.vn>

>  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> [
> 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.
>
>
> 2010/4/3 Nguyễn Mạnh Hùng <hungngm at bkav.com.vn>
>
>>  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> [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.
>>
>> 2010/3/31 Nguyễn Mạnh Hùng :
>>
>> > 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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>
>
> --
> Anthony Minessale II
>
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-- 
Anthony Minessale II

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