Displaying 20 results from an estimated 4000 matches similar to: "canreinvite, bandwidth, dial option"
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]
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
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 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.
2008 Mar 13
5
Newbie One-touch Recording: Does not work
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.
Any suggestions?
Here is the console log:
2007 Feb 24
0
Call was hangup when LIMIT_WARNING_FILE was playing
Dear All,
I tried to use 'L' option on my dial command.
I set the x to 65000(65 seconds), y to 60000(60 seconds), z to
30000(30 seconds).
The max calltime should be 65 seconds, and it will play "beep.gsm" at
60 seconds left. And repeat the beep at 30 seconds left.
But the call will be hangup by system at 60 seconds left.
In another word, when it plays warning file, the call
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
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
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 Apr 30
0
automon *1 not working; asterisk-1.4.22.1
automon is not working for me with asterisk 1.4.22.1
in extension.conf
[globals]
DYNAMIC_FEATURES=>automon
dial is with "w"
feature.conf
automon => *1
-- Executing [11 at internal:1] Playback("SIP/218-007556b0", "transfer") in new stack
-- <SIP/218-007556b0> Playing 'transfer' (language 'en')
-- Executing [11 at internal:2]
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
2009 Dec 07
1
automon => *1 "one touch recording"
I'm using Asterisk 1.4 but my "one touch recording" is not working:
feature.conf
automon => *1
extension.conf
[globals]
DYNAMIC_FEATURES=>automon
exten => 117,1,Dial(SIP/117,30,jrwW)
When I press "*1" on incoming call asterisk is not recording anything.
Did I miss any setting?
--
Joseph
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 =>
2007 Apr 02
0
automonitor and CDR(userfiled)
Hi all !
I'm trying to make a automonitor generated filename to "make its way"
into CRD(usrefiled), so I can keep track of recorded conversations in
CDR logs. Looking how to do that, I have found cool (but almost
undocumented) option of res_monitor: if you set monitor format in form
of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the
2006 Mar 30
0
BUG: FOP reports incorrect (duplicate) IP address until restarted
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
phone. Okay so far. Call is hung up and the same extension is used to
call another agent okay again, no
2006 Jan 20
1
applicationmap
Hi -
I'm trying to implement the applicationmap stuff in features.conf, and I
can't seem to get it to work. I'm testing it out on 1.2.2 with Polycom
IP500s and Snom190s.
My features.conf looks like this:
[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
parkingtime => 240
transferdigittimeout => 2
;courtesytone = beep
2007 Oct 03
4
IAXy and hook flash transfer
In features.conf, I have uncommented the transfer features under feature
map, but I still cannot transfer using a POTS phone on an IAXy adapter.
I think I am missing something here.... Any help is appreciated.
Here is features.conf:
;
; Sample Parking configuration
;
[general]
parkext => 700 ; What extension to dial to park
parkpos => 701-720 ;
2010 Jun 16
1
Blind transfer feature
Hi,
Am running 1.4.18 at the moment, and am trying to implement inline blind
transfer.
I have :
[featuremap]
blindxfer => *6 ; Blind transfer
in features.conf
And in extensions .conf under [globals] :
DYNAMIC_FEATURES=automon#blindxfr
So what am I missing ??
Have read through
http://www.voip-info.org/wiki/view/Asterisk+config+features.conf
Thanks,
2008 Nov 01
0
asterisk 1.2 and Dial with LIMIT_WARNING_FILE
Hi fellows..
I have 2 asterisk servers in which the following line
exten => _09049.,111,SetVar(LIMIT_PLAYAUDIO_CALLER=YES)
exten => _09049.,112,SetVar(LIMIT_WARNING_FILE=beep)
exten => _09049.,113,Dial(${TYPE}${DESTINO}|30|L(30000:10000))
works OK on my Asterisk 1.2.9, it plays the beep 10 seconds before the
end of the call.
doesn't work on my Asterisk 1.2.13, it hungs 10
2009 Jun 26
0
Problem with RetryDial
I issue this command:
RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ
ueue^SIP/GXP280_18))
Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds.
Asterisk rings again for 10 seconds. I would expect this to happen a total
of 4 times.
The problem is that after the second ring for 10 seconds Asterisk exits the
RetryDial step with HANGUPCAUSE=0 and