asterisk users - Feb 2008 - play promt at the same time to calling and callee

Hello,

   I want  that,  when call is answered , callee and calling would hear
different prompts and after promts the calls would be bridged. I've tried
this situation:
exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world)
exten =>
s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno))

But these prompts play not in the same time: just after conf-enteringno
prompt asterisk plays hello world promt.
-- <SIP/trunk-out-08155880> Playing 'conf-enteringno' (language
'en')
 -- <SIP/sip3.call.lt-08151550> Playing 'hello-world' (language
'en')

So my question is , how to do this in the same time. Maybe somebody is using
Dial G(context^exten^pri) for this purpose?

Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/f2448c3f/attachment.htm

2008/2/1, Giedrius Augys <voipas at gmail.com>:
>
> Hello,
>
>    I want  that,  when call is answered , callee and calling would hear
> different prompts and after promts the calls would be bridged. I've
tried
> this situation:
> exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world)
> exten =>
> s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno))
>
> But these prompts play not in the same time: just after conf-enteringno
> prompt asterisk plays hello world promt.
> -- <SIP/trunk-out-08155880> Playing 'conf-enteringno'
(language 'en')
>  -- <SIP/sip3.call.lt-08151550> Playing 'hello-world'
(language 'en')
>
> So my question is , how to do this in the same time. Maybe somebody is
> using Dial G(context^exten^pri) for this purpose?
>
> Thanks
>

I have tried this :
exten => s,1,Dial(SIP/trunk-out/37052390920|60|rG(music-testinis^s^1))

[music-testinis]
exten => s,1,goto(1,1)
exten => s,2,goto(2,1)


exten => 1,1,Playback(lt/conf-enteringno)
exten => 2,1,Playback(lt/conf-enteringno)

but I get this:
god*CLI>
    -- Executing [37052031382 at test:1]
Goto("SIP/sip3.call.lt-08141e00",
"testuojame|s|1") in new stack
    -- Goto (testuojame,s,1)
    -- Executing [s at testuojame:1] Dial("SIP/sip3.call.lt-08141e00",
"SIP/trunk-out/37052390920|60|rG(music-testinis^s^1)") in new stack
    -- Called trunk-out/37052390920
    -- SIP/trunk-out-0818fb40 is ringing
    -- SIP/trunk-out-0818fb40 is making progress passing it to
SIP/sip3.call.lt-08141e00
    -- SIP/trunk-out-0818fb40 is making progress passing it to
SIP/sip3.call.lt-08141e00
    -- SIP/trunk-out-0818fb40 answered SIP/sip3.call.lt-08141e00
    -- Executing [s at music-testinis:1]
Goto("SIP/sip3.call.lt-08141e00",
"1|1") in new stack
    -- Goto (music-testinis,1,1)
    -- Executing [1 at music-testinis:1]
Playback("SIP/sip3.call.lt-08141e00",
"lt/conf-enteringno") in new stack
    -- <SIP/sip3.call.lt-08141e00> Playing 'lt/conf-enteringno'
(language
'en')
    -- Executing [s at music-testinis:2]
Goto("SIP/trunk-out-0818fb40", "2|1")
in new stack
    -- Goto (music-testinis,2,1)
    -- Executing [2 at music-testinis:1]
Playback("SIP/trunk-out-0818fb40",
"lt/conf-enteringno") in new stack
    -- <SIP/trunk-out-0818fb40> Playing 'lt/conf-enteringno'
(language 'en')
  == Auto fallthrough, channel 'SIP/sip3.call.lt-08141e00' status is
'UNKNOWN'
  == Auto fallthrough, channel 'SIP/trunk-out-0818fb40' status is
'UNKNOWN'


My question is , how to bridge these two calls. I'm using Asterisk 1.4.11,
Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/a1bc1c26/attachment.htm