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(${g_pmtPath}Intro)
same => n,WaitExten(60)
exten => 2,1,NoOp(Dial Through)
same => n,Set(_l_CallerIDnum=6168310000)
same => n,Set(_l_Carrier=0001)
same => n,Set(l_DialTo=6167761066)
same => n,Set(l_Proxy=002)
same => n,Dial(PJSIP/${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
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----------------------------------------
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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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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