Displaying 20 results from an estimated 3000 matches similar to: "MixMonitor & RingBack Tone Issue"
2006 Feb 10
1
2wav2mp3, monitor, mixmonitor, mpg123, queues
Hello!
I'm using Asterisk for our office telephony, but we have some problems
that still we can't resolve about it. Here they are:
1) merge in/out call recording files
I also tried to use a script I found on the internet, called 2wav2mp3
In extensions.conf I added the following lines
; script to be executed when monitoring has been finished
MONITOR_EXEC=/usr/local/bin/2wav2mp3
exten
2015 Apr 22
1
MixMonitor Files Always Empty
Hi, sorry to bump this one but I still have this problem.
The file is always created but is always zero size. This is the dial plan that records the call:
exten = _0[1-8]X.,1,Set(CALLFILENAME=/var/spool/asterisk/callrecordings/its/${STRFTIME(${EPOCH},,%Y/%m/%d)}/Outbound-${UNIQUEID})
exten = _0[1-8]X.,2,MixMonitor(${CALLFILENAME}.gsm,b)
The dial plan then calls a macro that makes the call.
I?ve
2006 Feb 22
3
DTMF Mode supported by VoiceMail Application
Hi,
I would like to use Asterisk as VoiceMail system ...
the only issue I have is with DTMF recognition.
Which mode should I force into sip.conf ( general, only for peer ? )
so that the Voicemail application is understanding password from users ...
inband : works, but has some glitch ... not always good ... don't know why.
rfc2833 : doesn't seem to work ..
info : said to be not working
2006 Feb 17
3
MixMonitor and command
Has anyone had any success using the MixMonitor() plus "command" as
nothing I have tried works.
I am using 1.2.1 I did google the archive but couldn't see any mention
of anyone using this. What I am hoping to do is run a macro on hangup,
current method I am using seems to miss some calls 5% of calls fail to
mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.
2012 Jan 25
3
Executing Script after MixMonitor is called
Hello Guys,
I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan
exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"
2004 Dec 28
1
Asterisk / 183 message
Hello,
My company is doing some * testing with our Class 5 softswitch and had
some questions regarding ringback being provided to our PSTN users (off
--> on net calling)
Currently with MGCP subscribers, we know the PSTN ringing is provided by
a digital PBX for example, However, it looks like with SIP, our
softswitch is relying on MGCP signaling on our PSTN gateways to provide
ringback
2007 Feb 04
1
Help - Received response: "Forbidden" from '"Unknown"
I have a weird problem....
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext -> Panasonic Ext No Problems
Panasonic Ext -> SIP Ext No Problems
SIP Ext -> VOIP Provider No Problems
Panasonic Ext -> VOIP Provider Errors
---------- Working SIP -> VOIP
-- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",
2007 Jan 28
0
Trouble outgoing VOIP Provider Calls
I have a weird problem....
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext -> Panasonic Extensions No Problems
Panasonic Ext -> SIP Extensions No Problems
SIP Ext -> VOIP Provider No Problems
Panasonic Ext -> VOIP Provider Errors
---------- Working SIP -> VOIP
-- Executing [903........@from-sip:1]
2007 Feb 05
0
Help - Received response: "Forbidden" from'"Unknown"
I did a NoOp and see what the callerid was and when coming from the SIP Ext->Voip it is set to the Extension Number of the SIP Extension (as you would expect).
When coming from the Panasonic the CallerID is blank, I tried setting it to nothing again, and I tried setting it to the callerid of the voip provider, a sip extension id, the extension number on the Panasonic side, the zap channel
2005 Feb 20
0
Traditional Ringback Tone
I am trying to get Asterisk to emulate the sounds of the earlier
telephone systems, and the settings in [us-old] are pretty helpful. The
only thing lacking is ringback tone, which is not quite as complex as
the real phone systems of the day. For example, it is true that a
ringback tone commonly used is 420Hz modulated by 40Hz. This is what
shows up in [us-old]. But that modulated tone was
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323 connections
I am using Asterisk (Debian unstable packages) with an OH323 connection to my
provider. Everything is working except for the generation of ringback tones
when I receive inbound calls from the PSTN. My provider tells me that we're
sending call progress indications and that because of this they're expecting
us to generate the ringback tone. Does anybody know how to configure this in
2008 Jun 06
2
Bad ringback tone on zap channel
Hi,
I've noticed that sometimes instead of getting a regular ring tone
when calling out on a Zap channel, I get this obnoxious loud noise
which forces me to hang up.
Is this a problem in the Zaptel driver? I seem to recall that ringback
tones are generated by zaptel when dialing out from a SIP phone over a
Zap trunk.
Thanks.
2011 May 08
1
no ringback tone on outgoing call PRI line
Hi,
I have PRI configured and up but when i am dialing outside i am not getting any ringback tone but my call is connected. following is my example
SIP----------------->PRI ------------> mobile
I have set progress=yes in chan_dahdi.conf but still not working
if i call inbound from my mobile to internal extension ringing working
please help me
-S
-------------- next
2009 Jan 09
1
fake ringback tone
hi:
When iam sending calls through sip a fake ringback tone is generated and then call status can't be viewed (if call is ringing,busy,offline) it just rings and rings.
Can i disable this?
Thanks in advance.
_________________________________________________________________
Windows Live?: Keep your life in sync.
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323connections
are you giving answer()?
..o-------------------------------------------------------o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose
Comellas
Sent: Friday, September 30, 2005 10:32 AM
To: Asterisk Users
2004 Dec 21
0
No Ringback tone on Stable 1.0.2
I am noticing that calls that come from our IAX pstn gateway provider
and terminate to our Asterisk IVR do not receive ringing when an
extension is dialed. For example:
1. An inbound PSTN caller calls our number
2. Asterisk answers and provides greeting
3. PSTN user dials extension of internal SIP phone
4. No ringback is heard from PSTN callers perspective
5. SIP user picks up or the
2010 Jul 23
1
ringback tone after MOH, before queue member bridged
Good morning,
i've noticed many times that there are IVRs that play a ring tone just
before bridging me to an agent. My asterisk does not behave like this
but i've always wanted to.
I'm now playing with 1.6.2.9 and i've read in queue's doc:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
R ? stops moh and rings once an agent is ringing (Asterisk Trunk)
(in
2003 Nov 07
2
No ringing tone
I have the following setup:
AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2
When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well.
When making a call from Phone2, I get a dial tone but after dialing the number
I hear nothing (no ringing tone). On Asterisk console it says that a call is
coming in and that it is ringing Zap/2. I can also hear the
2006 Feb 13
1
How to Get SIP Header : To Field ?
Hi,
I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch.
When forwarding a call to Voicemail, here is somehow what the softswitch
sends to Asterisk :
In INVITE : Vm Phone Number ( to route the call )
In To : Person who has been called !
In From : Person who was calling !
Of course, I need to send the call into the "Called User" Mailbox (Thus To
SIP header) !
So
2016 May 03
2
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Whoops, email client auto-filled dev previously. Let's try this again.
Michael Maier wrote:
<snip>
> Ok - but this doesn't seem to answer my main question:
>
> Why must
>
> progressinband=never
>
> be applied especially if asterisk uses it by default? The big difference
> between w/ and w/o it is:
The default in 13 is "no" which still