<br><br><div><span class="gmail_quote">2008/2/1, Giedrius Augys <<a href="mailto:voipas@gmail.com">voipas@gmail.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<br><br> 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:<br>exten => s,1,Set(LIMIT_CONNECT_FILE=hello-world)<br>
exten => s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno))<br><br>But these prompts play not in the same time: just after conf-enteringno prompt asterisk plays hello world promt.<br>-- <SIP/trunk-out-08155880> Playing 'conf-enteringno' (language 'en')<br>
-- <SIP/sip3.call.lt-08151550> Playing 'hello-world' (language 'en')<br><br>So my question is , how to do this in the same time. Maybe somebody is using Dial G(context^exten^pri) for this purpose?<br>
<br>Thanks<br>
</blockquote></div><br><br clear="all">I have tried this :<br>exten => s,1,Dial(SIP/trunk-out/37052390920|60|rG(music-testinis^s^1))<br><br>[music-testinis]<br>exten => s,1,goto(1,1)<br>exten => s,2,goto(2,1)<br>
<br><br>exten => 1,1,Playback(lt/conf-enteringno)<br>exten => 2,1,Playback(lt/conf-enteringno)<br><br>but I get this:<br>god*CLI><br> -- Executing [37052031382@test:1] Goto("SIP/sip3.call.lt-08141e00", "testuojame|s|1") in new stack<br>
-- Goto (testuojame,s,1)<br> -- Executing [s@testuojame:1] Dial("SIP/sip3.call.lt-08141e00", "SIP/trunk-out/37052390920|60|rG(music-testinis^s^1)") in new stack<br> -- Called trunk-out/37052390920<br>
-- SIP/trunk-out-0818fb40 is ringing<br> -- SIP/trunk-out-0818fb40 is making progress passing it to SIP/sip3.call.lt-08141e00<br> -- SIP/trunk-out-0818fb40 is making progress passing it to SIP/sip3.call.lt-08141e00<br>
-- SIP/trunk-out-0818fb40 answered SIP/sip3.call.lt-08141e00<br> -- Executing [s@music-testinis:1] Goto("SIP/sip3.call.lt-08141e00", "1|1") in new stack<br> -- Goto (music-testinis,1,1)<br> -- Executing [1@music-testinis:1] Playback("SIP/sip3.call.lt-08141e00", "lt/conf-enteringno") in new stack<br>
-- <SIP/sip3.call.lt-08141e00> Playing 'lt/conf-enteringno' (language 'en')<br> -- Executing [s@music-testinis:2] Goto("SIP/trunk-out-0818fb40", "2|1") in new stack<br> -- Goto (music-testinis,2,1)<br>
-- Executing [2@music-testinis:1] Playback("SIP/trunk-out-0818fb40", "lt/conf-enteringno") in new stack<br> -- <SIP/trunk-out-0818fb40> Playing 'lt/conf-enteringno' (language 'en')<br>
== Auto fallthrough, channel 'SIP/sip3.call.lt-08141e00' status is 'UNKNOWN'<br> == Auto fallthrough, channel 'SIP/trunk-out-0818fb40' status is 'UNKNOWN'<br><br><br>My question is , how to bridge these two calls. I'm using Asterisk 1.4.11,<br>
Thanks<br>