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

Fri May 12 19:20:37 MST 2006

CLI*> show application dial

 -----Original Message-----
From:   "Dave Morrow" <david.morrow at autodata.net>
To:     "Dave Morrow" <david.morrow at autodata.net>, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Cc:     
Sent:  Fri, 12 May 2006 14:12:21 -0400
Delivered:  Fri,  12 May 2006 15:16:15 
Subject:[Asterisk-Users] features.conf *1 Call Recording

I found the issue.

It was my Dial command!

In my dialplan I had Dial(SIP/100|20|Ttr,,wW) as this was something I
gleaned from a sample config for call forwarding.  I removed the |20|Ttr
and now the call recording works!  Anyone know what the |20|Ttr did
anyhow?

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

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________________________________

From: Dave Morrow 
Sent: Friday, May 12, 2006 10:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] features.conf *1 Call Recording

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

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________________________________

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! >

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Solutions. This email and any files transmitted with it are confidential
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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 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
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	are addressed. If you have received this email in error please
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	-----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]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] 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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 --Original Message Ends--

-- 
Melcon Moraes <melcon.moraes at terra.com.br>