[Asterisk-Users] features.conf *1 Call Recording

Dave Morrow david.morrow at autodata.net
Fri May 12 10:21:34 MST 2006


I have one Sipura SPA-841 which is configured to use dtmfmode=info and
one Cisco 7905 which is using the default signalling (I believe this is
rfc2833) 
I have also set relaxdtmf=yes in sip.conf
 
I've tried pressing *1 on both phones (they are both on my desk) and
both behave the same.
 
;
; Sample Parking configuration
;
 
[general]
parkext => 700                  ; What ext. to dial to park
parkpos => 701-720              ; What extensions to park calls on
context => parkedcalls          ; Which context parked calls are in
;parkingtime => 45              ; Number of seconds a call can be parked
for
                                ; (default is 45 seconds)
;transferdigittimeout => 3      ; Number of seconds to wait between
digits when transfering a call
;courtesytone = beep            ; Sound file to play to the parked
caller
                                ; when someone dials a parked call
;xfersound = beep               ; to indicate an attended transfer is
complete
;xferfailsound = beeperr        ; to indicate a failed transfer
;adsipark = yes                 ; if you want ADSI parking announcements
;findslot => next               ; Continue to the 'next' parking space.
Defaults to 'first' available
;pickupexten = *8               ; Configure the pickup extension.
Default is *8
featuredigittimeout = 2000      ; Max time (ms) between digits for
                                ; feature activation.  Default is 500
 

[featuremap]
blindxfer => #1         ; Blind transfer
disconnect => *0                ; Disconnect
automon => *1                   ; One Touch Record
atxfer => *2                    ; Attended transfer
 
[applicationmap]
;testfeature => #9,callee,Playback,tt-monkeys   ;Play tt-monkes to
                                                ;callee if #9 was
pressed
 
~
~
~

 
David Morrow
Technical Systems Lead
Autodata Solutions Company
David.Morrow at Autodata.net
http://www.autodatasolutions.com <http://www.autodatasolutions.com/> 
 
Tel: (519) 963-3020
Fax: (519) 451-6615
 
< Lead, follow or get out of the way! >
 
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________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Giridhar
Reddy Bandi
Sent: Friday, May 12, 2006 12:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] features.conf *1 Call Recording


hi Dave 

i get the following log on *CLI> 

   -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4
    -- Playing 'beep' (language 'en')
    -- User hit '*1' to record call. filename:
wav|auto-1147452537-200-204|m 
    -- Playing 'beep' (language 'en')
    -- User hit '*1' to stop recording call.
    -- Attempting native bridge of SIP/200-39f4 and SIP/204-2ce4

what are you using as SIP client ( imean softphone/ analog phone + ATA /
IPphone ) ? 

if you are using a softphone and that doesnot have a dtmf signaling then
asterisk will not 
be able to recognize that you are pressing.

--Giridhar Bandi 


On 5/12/06, Dave Morrow <david.morrow at autodata.net> wrote: 

	It's quite strange. When I press *1 I do not hear a tone
indicated that it's even trying to record.
	
	 
	David Morrow
	Technical Systems Lead
	Autodata Solutions Company
	David.Morrow at Autodata.net
	http://www.autodatasolutions.com
<http://www.autodatasolutions.com/> 
	 
	Tel: (519) 963-3020
	Fax: (519) 451-6615
	 
	< Lead, follow or get out of the way! >
	 
	This message has originated from Autodata Solutions. The
attached material is the Confidential and Proprietary Information of
Autodata Solutions. This email and any files transmitted with it are
confidential and intended solely for the use of the individual or entity
to whom they are addressed. If you have received this email in error
please delete this message and notify the Autodata system administrator
at 

	Administrator at autodata.net <mailto:Administrator at autodata.net
<mailto:Administrator at autodata.net>  >
	 

________________________________

	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dave
Morrow
	Sent: Friday, May 12, 2006 8:39 AM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: RE: [Asterisk-Users] features.conf *1 Call Recording
	
	
	Yes. I did.
	 
	David Morrow
	Technical Systems Lead
	Autodata Solutions Company
	David.Morrow at Autodata.net
	http://www.autodatasolutions.com
<http://www.autodatasolutions.com/> 
	 
	Tel: (519) 963-3020
	Fax: (519) 451-6615
	 
	< Lead, follow or get out of the way! >
	 
	This message has originated from Autodata Solutions. The
attached material is the Confidential and Proprietary Information of
Autodata Solutions. This email and any files transmitted with it are
confidential and intended solely for the use of the individual or entity
to whom they are addressed. If you have received this email in error
please delete this message and notify the Autodata system administrator
at 

	Administrator at autodata.net <mailto:Administrator at autodata.net
<mailto:Administrator at autodata.net>  >
	 

________________________________

	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Giridhar
Reddy Bandi
	Sent: Friday, May 12, 2006 3:41 AM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: Re: [Asterisk-Users] features.conf *1 Call Recording
	
	
	did you include automon => *1 in your features.conf ?? 
	it should be somthing like this 
	
	[featuremap]
	automon => *1           
	
	--Giridhar Bandi 
	
	
	On 5/12/06, Dave Morrow <david.morrow at autodata.net> wrote: 

		Thanks for the response.  How would I change the DTMF
transfer mode?
		
		
		David Morrow
		Technical Systems Lead
		Autodata Solutions Company
		David.Morrow at Autodata.net 
		http://www.autodatasolutions.com
		
		Tel: (519) 963-3020
		Fax: (519) 451-6615
		
		< Lead, follow or get out of the way! >
		
		This message has originated from Autodata Solutions. The
attached 
		material is the Confidential and Proprietary Information
of Autodata
		Solutions. This email and any files transmitted with it
are confidential
		and intended solely for the use of the individual or
entity to whom they 
		are addressed. If you have received this email in error
please delete
		this message and notify the Autodata system
administrator at
		Administrator at autodata.net <mailto:
Administrator at autodata.net>
		
		
		-----Original Message-----
		From: asterisk-users-bounces at lists.digium.com 
		[mailto:asterisk-users-bounces at lists.digium.com] On
Behalf Of Fabio
		Sent: Thursday, May 11, 2006 6:55 PM
		To: Asterisk Users Mailing List - Non-Commercial
Discussion 
		Subject: RE: [Asterisk-Users] features.conf *1 Call
Recording
		
		if you ar using SIP clients, try changing DTMF transfer
mode.
		For test use
		> sip debug
		on your * console, then place a call and watch the
output. In INFO or 
		rfc2833 mode you must see the codes like SIP messages.
If you are using
		inband transfer mode (DTMF codes are  transferred like
sounds) you don't
		see the codes.
		
		Also, try adjusting featuredigittimeout in features.conf
:
		
		[general]
		featuredigittimeout = 2000 ; 2 seconds
		
		because the default 500ms is a very short time.
		
		Fabio Olaechea
		
		3Tech SRL
		Calle 48 Nro 632, Of. 67.
		La Plata, CP B1900AMZ
		Buenos Aires, Argentina. 
		Tel. +54 221 445 0244 Ext. 301
		Fax. +54 221 445 0245
		www.trestech.com.ar
		
		
		-----Mensaje original-----
		De: asterisk-users-bounces at lists.digium.com
		[mailto: asterisk-users-bounces at lists.digium.com
<mailto:asterisk-users-bounces at lists.digium.com> ]En nombre de Dave
Morrow
		Enviado el: Miercoles, 10 de Mayo de 2006 02:48 p.m.
		Para: Asterisk Users Mailing List - Non-Commercial
Discussion
		Asunto: RE: [Asterisk-Users] features.conf *1 Call
Recording
		
		
		OK. You lost me.
		
		
		David Morrow
		Technical Systems Lead
		Autodata Solutions Company 
		David.Morrow at Autodata.net
		http://www.autodatasolutions.com
		
		Tel: (519) 963-3020
		Fax: (519) 451-6615
		
		< Lead, follow or get out of the way! > 
		
		This message has originated from Autodata Solutions. The
attached
		material is the Confidential and Proprietary Information
of Autodata
		Solutions. This email and any files transmitted with it
are confidential 
		and intended solely for the use of the individual or
entity to whom they
		are addressed. If you have received this email in error
please delete
		this message and notify the Autodata system
administrator at
		Administrator at autodata.net
<mailto:Administrator at autodata.net>
		
		
		-----Original Message-----
		From: asterisk-users-bounces at lists.digium.com
		[mailto: asterisk-users-bounces at lists.digium.com
<mailto:asterisk-users-bounces at lists.digium.com> ] On Behalf Of
Alejandro
		Vargas
		Sent: Wednesday, May 10, 2006 10:29 AM 
		To: Asterisk Users Mailing List - Non-Commercial
Discussion
		Subject: Re: [Asterisk-Users] features.conf *1 Call
Recording
		
		2006/5/10, Dave Morrow <david.morrow at autodata.net >:
		> I am attempting to setup Asterisk to allow me to press
*1 while in a
		> call to use automon to record the call but have had
absolutely no
		> success.  Is there a trick to this?
		
		May be a problem with the way you are sending the
dialtones. Try sending 
		as data.
		
		--
		Alejandro Vargas
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