voipas at gmail.com
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 Posted: Fri Feb 01, 2008 9:44 am Post subject: [asterisk-users] play promt at the same time to calling and |
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2008/2/1, Giedrius Augys <voipas at gmail.com>:
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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
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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
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