[asterisk-users] * Queues with legacy pbx extensions ?

[David fire]

hi
we ALWAYS use
sip phone <IP>*<-E1->pstn
or
sip phone<-IP->*<-E1->legasypbx<-?-> pstn

we use the digiums cards whit echo canceller and we havent any echo problem.

more than 20 or 30 installations.
whit almost every provider in the country.

dont be SO scared.
David

2009/1/26 Sriram <d_r_sriram at hotmail.com>

>  Hello Everybody
>
> I am using Trixbox 2.4 (with TE420P & PRI lines) .. my setup is like Calls
> -->Asterisk-->legacy pbx--->analog extensions(agents). Whenver a call comes
> in , asterisk dials the ACD number of the legacy pbx which in turn decides
> to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the
> 4th span and 5432 is the ACD number of legacy pbx under which agents like
> 102,103,104 are present. Now my question is  (might be very silly) :
>
> 1. I dont have a queue stats software for my legacy pbx, the agents stil
> logon and logout on their legacy pbx..I was wondering if its possible that
> all these 102,103,104 etc login to a queue which resides on asterisk ? Can
> they login and logout of the asterisk queue..I am asking this since there
> are very nice asterisk queue reporting utilities out there for free..and i
> wud like to use them ...this will also help me using screen pop-ups for my
> agents...i know SIP extensions can simplify my setup but echo problems scare
> me also i dont want to throw my legacy pbx on which i invested heavily..
>
> Can anyone throw some pointers ?
>
> Thanks in advance
>
> Sriram
>
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