Displaying 20 results from an estimated 1000 matches similar to: "Newbie One-touch Recording: Does not work"
2008 Mar 13
3
Newbie One-touch Recording: Does not work (more info)
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav file appear in /var/spool/asterisk/monitor or elsewhere.
Test A: Outside line calling in
2006 Apr 05
2
What causes deadlock?
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10 retries!
Here is the portion of the log:
Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)...
Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing
2009 Jan 22
1
Zap connection problem
Greetings all,
I'm trying to connect to an AT&T teleconference, but the
call is never marked as ANSWERED by asterisk and therefore won't bridge and
continue. The only work-around I've come up with so far is to dial like
this:
Exten => 744,1,Dial(Zap/g1,,p)
The "private" mode keeps the line open without trying to do a bridge, but
requires the
2006 Jan 16
2
automon - one touch record
Actually the docs for the Queue application say:
'w' -- allow the called user to write the conversation to disk via Monitor
'W' -- allow the calling user to write the conversation to disk via Monitor
couldn't get these to work tho. Does this mean I can do one touch recording with agents, or does it mean I can use the monitor() command? Very confusing...
Doug.
2007 Jun 29
2
features.conf / DTMF / automon hell
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 script in
my extension. I can see the dtmf tones on the wire as SIP INFO
packets. Using the Read() app I have verified that * is
2006 May 10
2
REPOST: features.conf *1 Call Recording
Hi all. I posted this earlier but never got any advice that helped. If
anyone knows how to get this going, I'd appreciate some advice.
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?
In extensions.conf
[globals]
DYNAMIC_FEATURES=>automon
[default]
exten =>
2008 May 08
3
Looking for a Snom expert
I would like to hire someone to help us tweak our asterisk system for Snom
phones.
We would like to enable things like:
One touch recording
One touch park orbits
Presence
Please contact off-list if you will be able to help.
Thermal
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2010 Sep 02
2
Call Recording Questions
Hi,
1) I want to create add *1 call recording and wanted to know whether the file is created during recording or only after? I want to syncronise the recorded files with my web server (on a different machine (Windows)) so I need a way of telling when the recorded call has ended before copying it over.
2) I tried setting up *1 in features.conf but when I press *1, all that happens is
2006 Apr 05
0
What does this error mean "app.c: Huh....? no dial for indications?"
Hi,
What does the following error mean:
Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications?
Here is the 'full' log around the error:
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to
agent '3002', on 'Local/510@default-6b6c,1'
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002
Apr 5 12:38:24 VERBOSE[22755]
2007 Oct 14
1
Problem: features (from features.conf) not available if call was originated by manager API or call file
Hello asterisk-users,
I setup my asterisk to support several features like
automon,blindxfer,atxfer,parkcall etc. by using features.conf and the
global variable
DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in
extension.conf. Every Dial() command in my diaplan has the appropriate
parameters out of {tTkWwW}.
For calls from my SIP phones everything works fine. Pressing #1 will
2008 Jan 02
3
1.4.?? or ZapTel 1.4.X DIGITAL Calls are Broken
Don't you just hate it when something was working and when you come to
use it in anger it's broken :-(
Something in the, fairly, recent series of Asterisk updates has broken
DIGITAL call passthrough.
I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a
Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover
cable).
This PBX used to be able to
2006 Mar 21
2
Voice mail not working with Asteriks 1.2.5
Hi,
I have upgraded my PBX to Asterisk 1.2.5 , previously I was using
Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not
working. The error I am receiving in log files is like following,
WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12'
I have searched for solution a lot can Any one of you let me know how can I
solve this issue
2008 Apr 03
1
Hearing "transfer" during call
Hi list,
I enabled the transfer function in my dialplan, but I may configure it
wrong because sometime when I call a SIP extension number from one FXS
port, the SIP side will hear word "transfer", I hear nothing, after
that, the call conversation is fine.I'v had this problem for a long
time, could not get clue where I configure it wrong. here is my
related config part:
sip.conf:
2006 Mar 06
0
No ring when doing blind transfer.
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I briefly hear music on hold and then it stops until the timeout for no
answer
2013 Apr 10
5
Setting a CDR field from using feature codes...
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I am trying to set the CDR(userfield) to a certain vaule using the
application map of features.conf but I am not able to do it. When I
receive a call I would like to tag it with a client code (3 digit
numeric) so I can referenci it later from the CDR. I have edited
features.conf with something like:
code => #111,self,SET(CDR(userfield(111))
or
2005 Jan 25
2
New native assisted transfer (atxfer) usage info required
Hi, I would like to use the new atxfer (native assisted transfer, see
mantis item #3241) , but I've partially been able to
make it work.
I can receive a call and then having the caller hear MOH while talking
with another extension (the one I want to transfer to), but then I can't
make the caller and the trasferred talk hanging up or pressing any key
combination I'm aware of.
My
2004 Jan 07
1
Call Rollover
Have a question about implementing Call Rollover with my current
extensions.conf configuration.
[macro-stdexten]
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail
w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they press #, return to start
exten =>
2012 Oct 31
1
Asterisk 11 and stdexten written in AEL invoked by pbx_config
Almost two years ago, a change between how AEL code is built into
Asterisk dialplan between minor versions made clear the need to
provide a sane entry point into AEL subroutines and that's how
AELSub() born.
With Asterisk 11 release, they way [stdexten] at extensions.conf is
invoked changed from Macro to Gosub using the 'missing context
feature' and this caused that any stdexten
2006 Jan 26
4
extension to extension dialing
Sorry for all the newbie questions. I really appreciate everyone's help
today.
Okay I've got outgoing and incoming calls working with no echo. yay! Now
I'm having an issue with SIP extension to extension calling. Any time I
dial another extension it goes right into voice mail. My
extensions.conf is pretty small and rough but, here's what I have right
now. Most of it was taken
2009 Mar 03
2
macro-stdexten question
I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489.
When one phone calls another, I see the following on the console
(here, 6223 dials 6123)
-- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489",
"stdexten|6123|SIP/6123&IAX2/6123") in new stack
-- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",