Asterisk 13 – Call Bridge Issue.
I have the following senerio.
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(${g_pmtPath}Intro)
same => n,WaitExten(60)
exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnuma68310000)
same => n,Set(_l_Carrier 01)
same => n,Set(l_DialToa67761066)
same => n,Set(l_Proxy 2)
same => n,Dial(PJSIP/${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
3 thoughts on - Asterisk 13 – Call Bridge Issue.
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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
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Bryant Zimmerman wrote:
What is the endpoint configuration and do you have direct media enabled?
Is NAT in use on both sides?