[asterisk-users] Asterisk 13 - Call Bridge issue.

[Bryant Zimmerman]

Even when using the U option just issuing the Answer does not seem to 
always work. I end up having to play a prompt of some sort to force the 
answer.. There has to be some kind of bug going on here.
  
 Thanks

Bryant 
  

----------------------------------------
 From: "Bryant Zimmerman" <BryantZ at zktech.com>
Sent: Thursday, March 31, 2016 6:54 PM
To: bryantz at zktech.com, "Asterisk Users Mailing List - Non-Commercial 
Discussion" <asterisk-users at lists.digium.com>
Subject: re: [asterisk-users] Asterisk 13 - Call Bridge issue.   
  

----------------------------------------

From: "Bryant Zimmerman" <BryantZ at zktech.com>
Sent: Thursday, March 31, 2016 6:33 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Asterisk 13 - Call Bridge issue.   
 I have the following scenario.

    

   Call file calls 1st party.

   When connected give called party option to connect to second party.

    

   Issue Dial to second party. Caller answers and the two are bridged 
together.

   My issue is that 4 out of 5 calls fail to bridge the audio.

    

   Am I  missing something or is there some kind of bug? Here is my test 
dialplan 

    

    

;Dialer Base Code Files.
;Variables are sent in from .call file
[calluser-intake]
exten => s,1,NoOp(Start Call Intake)
exten => s,2,NoOp(Setup any vars)
exten => s,n,Set(_g_pmtPath=/vapp/dialerprompts/)
exten => s,n,NoOp(What is Path = ${g_pmtPath})
exten => s,n,NoOp(Read Call File Vars)
exten => s,n,NoOp(Dial To - ${l_DialTo})
exten => s,n,NoOp(Proxy - Proxy.${l_Proxy})
exten => s,n,NoOp(Carrier Trunk - ${l_Carrier})
exten => s,n,Set(_l_CallerIDnum=${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=${g_SIPUser})
exten => 
s,n,Dial(PJSIP/${l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer-
playmsg^s^1))  

[dialer-header]
exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier})
same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum})
same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)})
same => n,Set(CONNECTEDLINE(number,i)=vap_002)
same => n,DumpChan(1)
same => n,Return()  

[dialer-playmsg]
exten => s,1,Goto(hold,1)
same => n,NoOp(Enter Play Message)
same => n,NoOp(Path = ${g_pmtPath})
same => n,SayAlpha(${g_SIPUser})
same => n,BackGround(${g_pmtPath}Intro)
same => n,WaitExten(60)  

exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=6168310000)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${l_DialTo}@proxy_001,30,b(dialer-header^s^1))  

exten => _X,1,NoOp(Digit Entry)
exten => _X,n,NoOp(Log Response)
exten => _X,n,Playback(${g_pmtPath}YouPressed)
exten => _X,n,SayNumber(${EXTEN})  

exten => hold,1,NoOp(Park Called)
exten => hold,n,While($[1 < 5])
exten => hold,n,Wait(90)
exten => hold,n,EndWhile 

    

   Any ideas on why the media would not flowing after it sates they bridge 
has completed

    

   Another point. If I use a b option in the second dial. to call another 
context on connect of the second call. I get audio played on that both 
caller and callee channels.

    

   Thanks

Bryant 
  Ok it appears that the channel is not answering when it bridges the two 
calls together. 
 If I use the U option to gosub to a context to force an Answer() before 
the bridge then things seem to work. I also tried the lower case "a" option 
to force the answer and nothing happens with it appears to be ignored. .. 
So the U option with a gosub to an Answer seems to be the only way to get 
this to work...
  
 This seems like a bug. Should the called channel answer when a call is 
made with the Dial() function? Can anyone chime in on this one. 
  
 Note: Current systems are on Asterisk 13.5.0 (So if this was a bug has it 
been fixed in the latest release.) I did not see anything in the change 
logs that I would attribute to this.
  
 Thanks 
 Bryant


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