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[nik600 at gmail.com]

Dear all

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:


1) receive a call and put it on-hold in a queue (OK)
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)
3) when the remote number answer, trigger an AGI/diaplan script that ask to accept the call pressing a specific key (OK)
4) if right key is pressed redirect the current call to the channel2Link, connecting the call in queue with the remote number (?)


Step 1,2,3 works properly but i'm not able to link the two channels, even using redirect,goto or pickupChan.


Any idea or help will be appreciated!


Thanks

--
/*************/
nik600
http://www.kumbe.it

[mad at madpilot.net]

On 06/30/16 15:08, nik600 wrote:

[nik600 at gmail.com]

oh, yes! Many thanks


2016-06-30 15:28 GMT+02:00 Guido Falsi <mad@madpilot.net (mad@madpilot.net)>:

[johnkiniston at gmail.com]

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.


You can use the M option in your dial to run a macro to prompt the user to accept the call.


Here's my connector macro, I call it with:

Dial(LOCAL/${CELLPHONE}@intern,60,M(connector))

[macro-connector]
exten => s,1,NoOP()
 same =>   n(TOP),Read(RESPONSE,beep&accept-reject,1,s,3,5); 3 tries with 5 seconds to respond each time
 same =>   n,GotoIf($[${LEN(${RESPONSE})} = 0]?REJECT,1)        ;If we didn't get a response try and fail gracefully
 same =>   n,GotoIf($["${RESPONSE}" = "1"]?ACCEPT,1)            ;Take the call
 same =>   n,GotoIf($["${RESPONSE}" = "2"]?REJECT,1)            ;Reject the Call
 same =>   n,Goto(s,TOP)

exten => ACCEPT,1,NoOP()                                        ;Just connect the caller and callee
 same =>   n,Playback(pls-wait-connect-call)
 same =>   n,MacroExit()                                        ;Return

exten => REJECT,1,NoOP()
 same =>   n,Playback(beep&cancelled&goodbye)
 same =>   n,Set(MACRO_RESULT=BUSY)                        ;Reject the call
 same =>   n,Hangup()
 same =>   n,MacroExit()                                        ;Return




On Thu, Jun 30, 2016 at 6:08 AM, nik600 <nik600@gmail.com (nik600@gmail.com)> wrote:

[nik600 at gmail.com]

thanks John

yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones.


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:


    -- Executing [s@macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new stack
    -- Executing [s@macro-myconnector:2] Read("SIP/pbx2-000004b2", "RESPONSE,beep,1,s,3,5") in new stack

    -- Accepting a maximum of 1 digits.
    -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en')

...
    -- User entered nothing, 2 chances left
    -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en')

...
    -- User entered nothing, 1 chance left
    -- <SIP/pbx2-000004b2> Playing 'beep.gsm' (language 'en')

...
    -- User entered nothing.
    -- Executing [s@macro-myconnector:3] GotoIf("SIP/pbx2-000004b2", "1?REJECT,1") in new stack



Any idea?












2016-06-30 21:50 GMT+02:00 John Kiniston <johnkiniston@gmail.com (johnkiniston@gmail.com)>:

[johnkiniston at gmail.com]

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.





On Thu, Jun 30, 2016 at 1:18 PM, nik600 <nik600@gmail.com (nik600@gmail.com)> wrote:

[nik600 at gmail.com]

this is the point, and the strange thing: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


2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston@gmail.com (johnkiniston@gmail.com)>:

[asterisk.org at sedwar...]

Please don't top post.

On Thu, 30 Jun 2016, nik600 wrote:

[nik600 at gmail.com]

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem.

btw,the 2 show channel are reported above:


the channel with DTMF working


kcenter*CLI> core show channel SIP/pbx2-000004b9 
 -- General --
           Name: SIP/pbx2-000004b9
           Type: SIP
       UniqueID: 1467323106.1275
      Caller ID: xxxx
 Caller ID Name: xxxx
    DNID Digits: yyyy
       Language: en
          State: Up (6)
          Rings: 0
  NativeFormats: 0x4 (ulaw)
    WriteFormat: 0x4 (ulaw)
     ReadFormat: 0x4 (ulaw)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: 29
      Frames in: 325
     Frames out: 44
 Time to Hangup: 0
   Elapsed Time: 0h0m6s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: c_Queues
      Extension: 01
       Priority: 1
     Call Group: 0
   Pickup Group: 0
    Application: Read
           Data: RESPONSE,beep,1,s,3,5
    Blocking in: ast_waitfor_nandfds





the channel with DTMF not working:


kcenter*CLI> core show channel Local/user1@c_Queues-5d47;1 
 -- General --
           Name: Local/user1@c_Queues-5d47;1 
           Type: Local
       UniqueID: 1467323176.1277
      Caller ID: zzz
 Caller ID Name: zzz
    DNID Digits: (N/A)
       Language: en
          State: Ringing (5)
          Rings: 0
  NativeFormats: 0x4 (ulaw)
    WriteFormat: 0x4 (ulaw)
     ReadFormat: 0x4 (ulaw)
 WriteTranscode: No
  ReadTranscode: No
1st File Descriptor: -1
      Frames in: 1
     Frames out: 0
 Time to Hangup: 0
   Elapsed Time: 0h0m13s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: c_Queues
      Extension: 01
       Priority: 1
     Call Group: 0
   Pickup Group: 0
    Application: AppQueue
           Data: (Outgoing Line)
    Blocking in: ast_waitfor_nandfds



the only difference i see is the "1st File Descriptor" pointing to -1


2016-06-30 23:29 GMT+02:00 Steve Edwards <asterisk.org@sedwards.com (asterisk.org@sedwards.com)>:

[nik600 at gmail.com]

i've tried rfc2833,inband and info having the same behaviour in all situation.

2016-06-30 23:53 GMT+02:00 nik600 <nik600@gmail.com (nik600@gmail.com)>:

[nik600 at gmail.com]

to simplify the scenario, i've changed some settings to create a more simple test-case:

i'm using this callfile:


Channel: DAHDI/g0/{mycellnumber}

Context:mytestdtmf
Extension:01
Priority:1


and this is my dialplan:


[mytestdtmf]


exten =>        01,1,Answer
exten =>        01,n,Read(digito,,1)
exten =>        01,n,SayDigits(${digito})



Any idea?





2016-07-01 0:13 GMT+02:00 nik600 <nik600@gmail.com (nik600@gmail.com)>:

[asterisk.org at sedwar...]

On Fri, 1 Jul 2016, nik600 wrote:

[nik600 at gmail.com]

finally i've found that the SIP gateway i'm using is based on a DAHDI channel and it seems that on outgoing calls, if the called leg sends some digit they are not forwarded toAsterisk.

i'm investigating on it....


2016-07-01 4:25 GMT+02:00 Steve Edwards <asterisk.org@sedwards.com (asterisk.org@sedwards.com)>: