<div dir="ltr">this is the point, and the strange thing:<div>DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested</div></div><div class="gmail_extra"><br><div class="gmail_quote">2016-06-30 22:53 GMT+02:00 John Kiniston <span dir="ltr"><<a href="mailto:johnkiniston@gmail.com" target="_blank">johnkiniston@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div></div>Looking at your logs it looks like you may need to modify your sip.conf, Check with your provider as to what kind of DTMF they support and configure sip.conf to use that type of signalling.<br><br><br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jun 30, 2016 at 1:18 PM, nik600 <span dir="ltr"><<a href="mailto:nik600@gmail.com" target="_blank">nik600@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">thanks John<div><br></div><div>yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones.</div><div><br></div><div>it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected:</div><div><br></div><div><div> -- Executing [s@macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new stack</div><div> -- Executing [s@macro-myconnector:2] Read("SIP/pbx2-000004b2", "RESPONSE,beep,1,s,3,5") in new stack</div></div><div> -- Accepting a maximum of 1 digits.</div><div><div> -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en')</div></div><div>...</div><div><div> -- User entered nothing, 2 chances left</div><div> -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en')</div></div><div>...</div><div><div> -- User entered nothing, 1 chance left</div><div> -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en')</div></div><div>...</div><div><div> -- User entered nothing.</div><div> -- Executing [s@macro-myconnector:3] GotoIf("SIP/pbx2-000004b2", "1?REJECT,1") in new stack</div></div><div><br></div><div>Any idea?</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><div><div><br><div class="gmail_quote">2016-06-30 21:50 GMT+02:00 John Kiniston <span dir="ltr"><<a href="mailto:johnkiniston@gmail.com" target="_blank">johnkiniston@gmail.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div>I think a simpler way to do this would be to define an member in your queues.conf that points to a local channel that calls the remote users cell phone.<br><br></div>You can use the M option in your dial to run a macro to prompt the user to accept the call.<br><br></div>Here's my connector macro, I call it with:<br><br>Dial(LOCAL/${CELLPHONE}@intern,60,M(connector))<br><div><br>[macro-connector]<br>exten => s,1,NoOP()<br> same => n(TOP),Read(RESPONSE,beep&accept-reject,1,s,3,5); 3 tries with 5 seconds to respond each time<br> same => n,GotoIf($[${LEN(${RESPONSE})} = 0]?REJECT,1) ;If we didn't get a response try and fail gracefully<br> same => n,GotoIf($["${RESPONSE}" = "1"]?ACCEPT,1) ;Take the call<br> same => n,GotoIf($["${RESPONSE}" = "2"]?REJECT,1) ;Reject the Call<br> same => n,Goto(s,TOP)<br><br>exten => ACCEPT,1,NoOP() ;Just connect the caller and callee<br> same => n,Playback(pls-wait-connect-call)<br> same => n,MacroExit() ;Return<br><br>exten => REJECT,1,NoOP()<br> same => n,Playback(beep&cancelled&goodbye)<br> same => n,Set(MACRO_RESULT=BUSY) ;Reject the call<br> same => n,Hangup()<br> same => n,MacroExit() ;Return<br><br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Thu, Jun 30, 2016 at 6:08 AM, nik600 <span dir="ltr"><<a href="mailto:nik600@gmail.com" target="_blank">nik600@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr">Dear all<div><br></div><div>i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is possible to configure a scenario like this:</div><div><br></div><div>1) receive a call and put it on-hold in a queue (OK)</div><div>2) monitor the queue and trigger an outbound call to a remote number using AMI, setting the channel of the on-hold on a specific var named channel2Link (OK)</div><div>3) when the remote number answer, trigger an AGI/diaplan script that ask to accept the call pressing a specific key (OK)</div><div>4) if right key is pressed redirect the current call to the channel2Link, connecting the call in queue with the remote number (?)</div><div><br></div><div>Step 1,2,3 works properly but i'm not able to link the two channels, even using redirect,goto or pickupChan.</div><div><br></div><div>Any idea or help will be appreciated!</div><div><br></div><div>Thanks<span><font color="#888888"><br clear="all"><div><br></div>-- <br><div data-smartmail="gmail_signature">/*************/<br>nik600<br><a href="http://www.kumbe.it" target="_blank">http://www.kumbe.it</a></div>
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