asterisk users - Jan 2008 - Default delay time for Attended call

A call comes in from the PSTN, Asterisk answers it, it goes to the directory,
and then to the extension the caller designates and the user at that extension
answers.  The user at the extension then wants to transfer the call to another
extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer?
soft key, then enter the extension number they want to transfer to, and hit the
?dial? soft key.  The user at the new extension answers and the talks to the
user doing the transfer.  They agree to transfer the call to the new extension
and the person who got the original call then hits the ?transfer? soft key and
hangs up.  6 seconds later the caller and the new extension can talk to each
other.  The line at the new extension is silent for those 6 seconds.

 

Thank you for help.

 

Here is the extension.cong file:

 

[general]

static = yes

writeprotect = no

clearglobalvars = no

 

 

[globals]

CONSOLE = Console/dsp

 

[trunkint]

exten => _9011.,1,Dial(Zap/g1/${EXTEN:1},70,Tt) 

 

[trunkld]

exten => _91NXXNXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

 

[trunklocal]

exten => _9NXXXXXX,1,Dial(Zap/g1/${EXTEN:1},70,Tt)

 

[trunktollfree]

exten => _91800NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

exten => _91888NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

exten => _91877NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

exten => _91866NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})

 

[international]

include => longdistance 

include => trunkint

 

[longdistance]

ignorepat => 9

include => local

include => trunkld

 

[local]

ignorepat => 9 

include => default 

include => trunklocal 

include => trunktollfree 

include => longdistance

 

include => parkedcalls

 

[macro-trunkdial]

exten => s,1,Dial(${ARG1})

exten => s,n,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Hangup

exten => s-BUSY,1,Hangup

exten => _s-.,1,NoOp

 

[macro-stdexten]

exten => s,1,Dial(${ARG2},20)             ; Ring the interface, 20 seconds
maximum

exten => s,2,Goto(s-${DIALSTATUS},1)      ; Jump based on status 

(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to
voicemail w/ unavail announce

exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/ busy
announce

exten => _s-.,1,Goto(s-NOANSWER,1)        ; Treat anything else as no answer 

exten => a,1,VoicemailMain(${ARG1})       ; If they press *, send the user
into VoicemailMain

 

[macro-stdPrivacyexten]

exten => s,1,Dial(${ARG2},20|p)  ; Ring the interface, 20 seconds maximum,
call screening ; option (or use

P for databased call screening) 

exten => s,2,Goto(s-${DIALSTATUS},1)  ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to
voicemail w/ unavail announce

exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/ busy
announce

exten => s-DONTCALL,1,Goto(${ARG3},s,1)  ; Callee chose to send this call to
a polite "Don't call again"

script.

exten => s-TORTURE,1,Goto(${ARG4},s,1)  ; Callee chose to send this call to a
telemarketer torture script.

exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as no answer 

exten => a,1,VoicemailMain(${ARG1})  ; If they press *, send the user into
VoicemailMain

 

[macro-page];

exten => s,1,ChanIsAvail(${ARG1}|js)  ; j is for Jump and s is for ANY call 

exten => s,n,GoToIf([${AVAILSTATUS} = "1"]?autoanswer:fail) 

exten => s,n(autoanswer),Set(_ALERT_INFO="RA")  ; This is for the
PolyComs

exten => s,n,SIPAddHeader(Call-Info: Answer-After=0)  ; This is for the
Grandstream, Snoms, and Others

exten => s,n,NoOp()  ; Add others here and Post on the Wiki!!!!

exten => s,n,Dial(${ARG1}||)

exten => s,n(fail),Hangup

 

[default]

exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

exten => _9011.,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

exten => _9911,1,Dial(${TRUNK}/${EXTEN:1},70,Tt)

 

exten => 6500,1,VoiceMailMain

exten => 6500,1,NoOp(${Exten:6:11})

exten => 6500,2,VoiceMailMain(s${CALLERID(all):1})

exten = o,1,Goto(default,6000,1) 

include => parkedcalls

 

;[asterisk_guitools]

;exten = executecommand,1,System(${command})

;exten = executecommand,n,Hangup()

;exten = record_vmenu,1,Answer

;exten = record_vmenu,n,Playback(vm-intro)

;exten = record_vmenu,n,Record(${var1})

;exten = record_vmenu,n,Playback(vm-saved)

;exten = record_vmenu,n,Playback(vm-goodbye)

;exten = record_vmenu,n,Hangup

;exten = play_file,1,Answer

;exten = play_file,n,Playback(${var1})

;exten = play_file,n,Hangup

;hasbeensetup = Y

 

[DID_trunk_1]

include = default

exten = s,1,Answer()

exten = s,n,NoOp(${CALLERID(num)})

exten = s,n,Directory(default||f)

 

 

[support]

include = default

exten = _X.,1,Goto(default|6009|1)

exten = s,1,Goto(default|6009|1)

 

[numberplan-custom-1]

plancomment = DialPlan1

include = default

exten = _9XXXXXXX,1,Macro(trunkdial,${TRUNK}/${EXTEN:1})

comment = _9XXXXXXX,1,Local,standard

exten = ,1,Macro(trunkdial,${TRUNK}/${EXTEN:0})

comment = ,1,Emergency911,custom

exten = _9XXXXXXXXXXX!,1,Macro(trunkdial,${TRUNK}/${EXTEN:1})

comment = _9XXXXXXXXXXX!,1,Longdistance,standard

 

 

Date: Thu, 31 Jan 2008 09:04:03 -0500

From: "C F" <shmaltz at gmail.com>

Subject: Re: [asterisk-users] Default delay time for Attended call

      transfer

To: "Asterisk Users Mailing List - Non-Commercial Discussion"

      <asterisk-users at lists.digium.com>

Message-ID:

      <81000b5a0801310604r27004306q6e6ed333acab24ba at mail.gmail.com>

Content-Type: text/plain; charset=ISO-8859-1

 

First time or second time they hit transfer?

Dial plan config?

 

2008/1/30 Don Smith <HYPERLINK "mailto:Don_Smith at
catalysthealthcare.ca"Don_Smith at catalysthealthcare.ca>:
> 
> 
> 
> 
> Greetings,
> 
> I have an issue with the length of time that passes from when someone hits
> the transfer soft key on a Cisco 7940, after doing an attended transfer,
and
> when the recipient's connects with the transferred call.  It appears to
be
> around 6 seconds.  Is there a .conf in Asterisk where this time can be
> reduced?
> 
> 
> 
> Thank you for your help
> 
> Don
> 
 


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> ----- Original Message ----
> From: Don Smith <Don_Smith at catalysthealthcare.ca>
> To: asterisk-users at lists.digium.com
> Sent: Thursday, 31 January, 2008 4:46:27 PM
> Subject: Re: [asterisk-users] Default delay time for Attended call
> 
> A call comes in from the PSTN, Asterisk answers it, it goes to the
directory, and then to the extension the caller designates and the user at that
extension answers.  The user at the extension then wants to transfer the call to
another extension; on the Cisco 7940 they push the ?more? soft key, then the
?Transfer? soft key, then enter the extension number they want to transfer to,
and hit the ?dial? soft key.  The user at the new extension answers and the
talks to the user doing the transfer.  They agree to transfer the call to the
new extension and the person who got the original call then hits the ?transfer?
soft key and hangs up.  6 seconds later the caller and the new extension can
talk to each other.  The line at the new extension is silent for those 6
seconds.
     

Hi Don,

There is no setting you can adjust on Asterisk. For an attended transfer there
is no further interaction with the Asterisk dialplan once the transfer button on
your Cisco is pressed it's all handled automagically in the SIP channel and
channel.c.

When I do an attended transfer through Asterisk the transfer takes about 1
second to complete, 6 seconds seems like a very large amount of time for the
operation. With an attended transfer there are no new audio streams to set up
for Asterisk and therefore no NAT that could be slowing things down as the audio
trys to get through. The only thing that springs to mind would be if the REFER
request from your Cisco was getting dropped along the way and taking a few
re-transmits to get through. That wouldn't be likely to happen everytime
though so a consistent 6 second delay is puzzling.

Regards,

Greyman.












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