On 28 August 2014 07:56, Leandro Dardini wrote:
> Can you post an example?
>
> Leandro
>
>
> 2014-08-28 0:47 GMT+02:00 Ishfaq Malik :
>
> Do the pause/unpause in a Macro or Gosub and reference that from the
>> features.conf
>>
>> Also, make sure you put the filename into a variable and give it fu
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik :
> Do the pause/unpause in a Macro or Gosub and reference that from the
> features.conf
>
> Also, make sure you put the filename into a variable and give it full
> inheritance so you can resume recording to the same file (
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini wrote:
> Hello,
> I ha
Hi guys,
I'm trying to get blind transfer to work and automatically transfer call
to another number on key sequence press.
Extensions.conf_snippet
[from-pstn]
exten => _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer)
exten => _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1)
exten =>
2010/6/21 Aksel Celasun
> Hello dear list.
>
>
>
>
>
> I am having issues on parkedcalls.
>
>
>
> I am using a Cisco SPA525G as a test phone, and I have the transfer button
> there when I am in a call,
>
> But when I want to transfer the current call I am in, I push the transfer
> button, and on
risk-users-boun...@lists.digium.com] På vegne av Ira
Sendt: 21. juni 2010 19:16
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] features.conf - parkedcalls - transfer
At 12:27 AM 6/21/2010, you wrote:
>Almost 10 seconds, before the transfer to sip
>200 is ma
>>And I can't see any button on the Cisco phone which will function like a
>>"direct transfer now", do I have to wait...?
Thank you for your reply.
In my Dialplan menu on the SPA525g, I have a field where the input are, and I
must say, I don't know if this is the right one, but the field conta
At 12:27 AM 6/21/2010, you wrote:
>Almost 10 seconds, before the transfer to sip
>200 is made, can I reduce that timer?
>
>And I cant see any button on the Cisco phone
>which will function like a direct transfer now, do I have to wait
?
On my Aastra phones, I press Transfer 101
Transfer. S
On Mon, Jun 21, 2010 at 2:27 AM, Aksel Celasun wrote:
> I am using a Cisco SPA525G as a test phone, and I have the transfer
> button there when I am in a call,
>
> But when I want to transfer the current call I am in, I push the transfer
> button, and onscreen I se “Enter Number”, and if I enter
When I enable the automon-feature (*1) the callee can start recording
the conversation. No problem there.
But I can't get my user-defined features to work.
I have setup the following test-feature in features.conf :
[applicationmap]
testfeat => *3,self/callee,Playback,tt-weasels
I have the follo
8 sep 2009 kl. 10.17 skrev jonas kellens:
> Erik,
>
> I have placed everything in features.conf in comment ( ; ). Still
> when I run show features, I get this :
>
>> clarkconnect*CLI> show features
>> Builtin Feature Default Current
>> --- --- ---
>> Pick
Erik,
I have placed everything in features.conf in comment ( ; ). Still when I
run show features, I get this :
> clarkconnect*CLI> show features
> Builtin Feature Default Current
> --- --- ---
> Pickup*8 *8
> Blind Transfer
just a hint. you might have # assigned the moh in feature.conf and #3
to starting the recording. check your feature.conf and makesure that #
isn't assigned to anything.
erik de wild
Tripple-o
Your Asterisk migration partner
the Netherlands
Verstuurd vanaf mijn iPhone
Op 7 sep 2009 om 20:40
On Monday 07 September 2009 13:40:16 jonas kellens wrote:
> [applicationmap]
>
> opnemencallee =>
> #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
=>
,[/],[,[,MOH_Class]]
it looks like "/var/samba/profiles/jonaskl/recording" is in the spot for
"[,MOH_Class]"
--
Anthony -
Also try putting Asterisk in the audiopath by setting "canreinvite=no" in
sip.conf
Regards
Ian
On Sat, Jun 7, 2008 at 4:07 PM, Michiel van Baak <[EMAIL PROTECTED]>
wrote:
> On 08:36, Sat 07 Jun 08, Russell Bryant wrote:
> >
> > On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote:
> > > i have this o
On 08:36, Sat 07 Jun 08, Russell Bryant wrote:
>
> On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote:
> > i have this on my features.conf:
> >
> > [applicationmap]
> > testfeature => *9,callee,Playback,tt-monkeys
> >
> > extensions.conf:
> >
> > [globals]
> > DYNAMIC_FEATURES=testfeature
> > trunk_1
On Jun 6, 2008, at 4:33 PM, Manolet Gmail wrote:
> i have this on my features.conf:
>
> [applicationmap]
> testfeature => *9,callee,Playback,tt-monkeys
>
> extensions.conf:
>
> [globals]
> DYNAMIC_FEATURES=testfeature
> trunk_1 = Zap/g1
> trunk_2 = Zap/g2
>
>
> what else i have to add in order to
Tilghman Lesher wrote:
> On Tuesday 22 April 2008 05:22, Sergey Shumeyko wrote:
>
>> I have following problem with my Asterisk installation (version 1.6.0. beta
>> 7.1). I want to assign start record conversation to #7 and stop record
>> conversation to #8, but it isn't working (on previous Aste
On Tuesday 22 April 2008 05:22, Sergey Shumeyko wrote:
> I have following problem with my Asterisk installation (version 1.6.0. beta
> 7.1). I want to assign start record conversation to #7 and stop record
> conversation to #8, but it isn't working (on previous Asterisk 1.2.17 it
> was working fine
I've ran into an issue (1.4.17) where anything in features.conf is being
totally ignored after the * or # for a particular feature.
Everything works just fine as long as I restrict the digit to only a * or #,
but apps that require #1 or *1 simply never get recognized.
No clue what's causing this,
Russell Bryant wrote:
> Here are some examples of setting up automon access when calling SIP/1234.
> Examples 2 and 3 use variable inheritance.
I think I made these examples *way* more complicated than they needed to be.
After going back and refreshing myself on configuring dynamic features in
Danny Brown wrote:
> I have been trying for a very long time to get asterisk to detect and
> utilize dtmf tones from my sip clients within my dial scripts. I have
> set automon=>#9 in my features.conf, I have Dial(,gWw) in my dial
> scripts. I have Set(DYNAMIC_FEATURES=automon) as the first scr
Danny Brown wrote:
> I have been trying for a very long time to get asterisk to detect and
> utilize dtmf tones from my sip clients within my dial scripts. I have
> set automon=>#9 in my features.conf, I have Dial(,gWw) in my dial
> scripts. I have Set(DYNAMIC_FEATURES=automon) as the first scr
John Novack wrote:
Can't this guy read?
I'll bet he runs Micro$oftware, and has fallen prey to one of its
thousands of exploits-du-jour.
Set a filter and move on. . .
B.
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Can't this guy read?
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From: Dave Morrow Sent: Friday, May
12, 2006 10:41 AMTo: '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
PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: 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') -
[EMAIL PROTECTED]
>
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Dave
MorrowSent: Friday, May 12, 2006 8:39 AMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users]
features.conf *1 Call Recording
Yes. I did.
David Morrow
Technical Syste
06 8:39 AMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users]
features.conf *1 Call Recording
Yes. I did.
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-3020
Fax: (519) 451-66
Friday, May 12, 2006 8:39 AMTo: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users]
features.conf *1 Call Recording
Yes. I did.
David Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodatasolutions.com
Tel: (519) 963-30
erisk Users Mailing List - Non-Commercial
DiscussionAsunto: RE: [Asterisk-Users] features.conf *1 Call
RecordingOK. You lost me.David MorrowTechnical
Systems LeadAutodata Solutions Company [EMAIL PROTECTED]http://www.autodatasolutions.comTel:
(519) 963-3020Fax: (519) 451-6615< Lead, follow
] [EMAIL PROTECTED]>-Original Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of FabioSent: Thursday, May 11, 2006 6:55 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] features.conf *1 Call Recordingif you ar using SIP clients, try c
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 li
ist - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] features.conf *1 Call Recording
2006/5/10, Dave Morrow <[EMAIL PROTECTED]>:
> 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.
Subject: Re: [Asterisk-Users] features.conf *1 Call Recording
2006/5/10, Dave Morrow <[EMAIL PROTECTED]>:
> 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?
2006/5/10, Dave Morrow <[EMAIL PROTECTED]>:
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
In extensions.conf file, in that context, you must have:
include => featuremap
Thats lets you transfer calls.
Regards.
--
José Luis Gómez
Qualis Information Technology
Av. Rivadavia 2553, PB Of. 43 EP
+54-342-4565684 int 102
www.qualis.com.ar
Soporte 0810-8880022
Santa Fe - Argentina
El vie, 28
i used to have this problem,
with me, it appeared that i had to press the feature keys very quickly.
my solution was to set featuredigittimeout higher than the default 500.
also, when i use IAX phones, i had to set dtmf to ouband audio for asterisk to recognize the keys pressed.On 4/28/06, Ronald W
On 13:57, Tue 14 Jun 05, Alexis FECOURT wrote:
Hi,
I am trying to use the attended transfer. So I put this in my feature.conf:
[general]
[featuremap]
atxfer => *0
blindxfer => #0
exten => _10X,2,Dial(SIP/${EXTEN},20,htT)
exten => _10X,3,Hangup
Hi,
The problem is in that h param.
On 13:57, Tue 14 Jun 05, Alexis FECOURT wrote:
> Hi,
> I am trying to use the attended transfer. So I put this in my feature.conf:
>
> [general]
> [featuremap]
> atxfer => *0
> blindxfer => #0
>
> exten => _10X,2,Dial(SIP/${EXTEN},20,htT)
> exten => _10X,3,Hangup
>
Hi,
The problem is in that
PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, June 07, 2005 7:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] Features.conf - atxfer
Reading through the code, I don't see a way of exitin
You can use super-valet-parking
On Tue, 2005-06-07 at 06:18 -0500, Mike Holloway wrote:
> Reading through the code, I don't see a way of exiting the transfer and
> regaining the call with the customer, unless the third party hangs up or
> maybe doesn't answer and the dialplan doesn't do anything
Reading through the code, I don't see a way of exiting the transfer and
regaining the call with the customer, unless the third party hangs up or
maybe doesn't answer and the dialplan doesn't do anything else with the
call (send the call into voicemail).
I suggest you request this feature (ht
Kevin Walsh wrote:
Josh Roberson [EMAIL PROTECTED] wrote:
no.. WRONG. rename parking.conf, as parking.conf is what features.conf
Oops. I knew it was one of them. At least I didn't say sip.conf :-)
True that. This is another reminder that everyone needs to make sure
that when they
Josh Roberson [EMAIL PROTECTED] wrote:
> Kevin Walsh wrote:
> > Or simply rename musiconhold.conf as features.com and restart Asterisk.
> >
> no.. WRONG. rename parking.conf, as parking.conf is what features.conf
>
Oops. I knew it was one of them. At least I didn't say sip.conf :-)
--
_/
Kevin Walsh wrote:
Chris Shaw [EMAIL PROTECTED] wrote:
Not in configs or /etc/asterisk/. Asterisk is still running, just
curious why I am not seeing that file.
Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's
been in there for over a week now, I just checked out a n
Chris Shaw [EMAIL PROTECTED] wrote:
> > Not in configs or /etc/asterisk/. Asterisk is still running, just
> > curious why I am not seeing that file.
> >
> Hmmm.. Maybe try checking out a fresh new copy in a different dir? It's
> been in there for over a week now, I just checked out a new copy and i
- Original Message -
From: "AJ Grinnell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 03, 2004 8:02 AM
Subject: RE: [Asterisk-Users] features.conf
> Not in configs or /etc/asterisk/. Asterisk is still running, just curious
> why I am
Not in configs or /etc/asterisk/. Asterisk is still running, just curious
why I am not seeing that file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Robert
Jackson
Sent: Tuesday, August 03, 2004 10:36 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk
> -Original Message-
> From: AJ Grinnell [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 03, 2004 10:28 AM
> To: Asterisk
> Subject: [Asterisk-Users] features.conf
>
>
> Is features.conf included in the cvs as of 8-1-04? I have
> updated, but am not seeing it?
>
I think that it sho
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