Displaying 20 results from an estimated 20000 matches similar to: "Double DTMF digits"
2007 May 12
2
zonedata.c
Hi,
Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly.
Thank you.
Jad Wauthier
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2018 May 01
2
DTMF tones in MixMonitor recording
Hello list,
Hope you are all doing fine!
I have stumbled over some piece of dialplan code in which apparently they
were trying to avoid recording the DTMF tones in the wav file. It is really
messy and I am not sure if this really works. So after a bit of research I
found this comment (
https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is
said:
*"Asterisk strips the
2012 Sep 28
1
ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Hi list!
ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF
gets transmitted throughout the conference. I've tried Asterisk 10.7.1
from the official RPMs and 10.8.0 compiled from source.
I've confirmed that it's disabled via the CLI "confbridge show profile
user <profilename>".
It's an all-SIP scenario with RFC2833 as the DTMF protocol.
2005 Aug 19
3
Sending digits from SIP to Asterisk's VoiceMailMain
Hi,
I am using Asterisk cmd VoiceMailMain to manage voice mail.
Problem is, voice mail box can't read password sent from SIP phone, but I
don't have any problem to read password from the handset attached to my
asterisk box.
Your help will be greatly appreciated.
Thanks,
2004 Jun 11
11
Broadvoice and DTMF
I understand there has been some issues sending DTMF tone through
Broadvoice. Can some provide me with symptoms?
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2005 Jun 01
3
DTMF not working
Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes.
I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I
configure dtmfmode=rfc2833 (I've tryied inband and info).
Asterisk seems not to "see" the tones. Could somebody help me? Thanks
2005 Aug 08
4
DTMF issues with SIPPhone?
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc. Has anyone else been having these issues? I'm
only accepting ulaw and alaw, and my relevant sip.conf information
follows:
[sipphone]
type=peer
2004 Dec 10
2
dtmfmode: inband question
Hello folks. I'm not sure if this is the right list for this
question, but I'll start here.
If I'm using a SIP provider and I have an entry in sip.conf that looks
like:
[8315551212]
type => friend
...
dtmfmode => inband
...
When I pick up the phone, call someone through this provider, and press
numeric digits to generate dtmf tones, who is actually generating the tones
at the
2004 Sep 01
5
dtmf problem
Hello!
I have asterisk updated from CVS on 31/8/2004 with
sample configuration. I have just changed the
sip.conf to register asterisk with sip proxy in out
intranet.
Then I can successfully make call to asterisk and go
to demo IVR, but no response to dtmfs.
I try to make call from several sip phones: Cisco7960,
Ata186, Snom200. All of them send telephone-event in
INVITE, but asterisk answers
2005 Jun 06
5
Polycom 500...
I am having a strange problem with a couple of Polycom IP 500 phones. I
know this is not related to Asterisk, but maybe someone here had the same problem.
I configured my phones following the documentation at voip-info.org and
they are working very well. The only problem I have is that when I dial an
extension like 1100 the phone changes that to 0110 and obviously the call
fails. I have
2009 Apr 09
2
DTMF
[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted:
Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but
URGENT[image:
Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>
2018 Sep 26
2
chan_pjsip: DTMF mode "auto_info" on endpoints
Hey all!
I recently tried the dtmf_mode "auto_info" on my setup to support endpoints that only understand SIP INFO as a fallback.
My setup is the following:
Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO)
Both are configured with "auto_info" dtmf_mode in pjsip.conf.
What I ran into is, that DTMF sent from endpoint A to endpoint B is additionally sent via
2006 Jun 16
2
DTMF in the middle of a call
I found this old post using a Google search for DTMF tones heard during
an Asterisk call. The calling party does not hear them, only the called
party.
We are experiencing this sporadic DTMF "beep" followed by a momentary
silence on a random number of our calls. This happens on trunk to SIP
and SIP to SIP calls, so it's not PRI related.
Since this thread is dated 2003, and the
2010 Nov 05
3
Elementary question - accessing feature codes from cell phone
Hi, please forgive me for this (hopefully) simple question. I cannot seem to
find an answer or solution while searching around.
I want to be able to call in to my server using my cell phone and be able to
set call forwarding for my extension and enter a phone number and also be
able to call in to that extension and disable the call forwarding. I see I
can do this through the ARI web interface
2005 Mar 15
1
(Yet another) Music on hold problem and another...
Hi,
I've recently installed Asterisk and have got the majority of it
configured (what an excellent piece of software it is, too), but I'm
having a couple of problems.
The first one is with music on hold! I've downloaded and
installed mpg123 as specified:
># whereis mpg123
>mpg123: /usr/local/bin/mpg123
It's the correct version:
>#
2009 Jan 24
3
Passing DTMF
Hello:
I need to be able to reliably send out touchtone to any calling party who comes
into my pbx. The standard things to help with this have been done as far as I
know:
1. dtmfmode is rfc2833.
2. The phones themselves are set to rfc2833.
3. allow=ulaw
4. On internal calls between extensions, touchtone works fine.
Also, I have reviewed sip.conf with my carriers.
Now for the
2007 Feb 23
1
Asterisk and DTMF
Hi list!
I have an Asterisk server (1.2.14) connected to a E1 line via a TE410P, and
some
PAP2NA connected to it. The PAP2 DTMF configurations is set to INFO and
Asterisk
to INFO too. At first, is INFO method different from RFC2833??
Well, I have two problems. The first is that when I place a call to outside,
via
E1 trunk, sometimes I get some DTMF tones and I'm sure nobody hit any key.
Seems
2008 Feb 25
1
Realtime Queue Status for Agents
Hello,
I am having a problem which stems from the fact that the Agents that
handle my call queues are unaware if there are are people waiting in the
queue. I have found the following configuration options but they are
not very helpful.
in queues.conf:
announce=xxx
The "announce = XXX" option in queues.conf makes Asterisk play the XXX
announcement to the member of the queue who
2004 Jan 22
3
Grandstream 101
Just got GS 101 phone and plugged into the network.
Got ip setup however, the following problems arise:
1. when dialing an extension, I cannot further send any key tone to
Asterisk.
2. there is no sound coming from the other end.
I have a sip.conf setup for GS:
[General]
disallow=all
allow=ulaw
allow=alaw
[gs]
canreinvite=no
dtmfmode=info
In the GS101 setting
rtp port = 5004
sip port = 5060
2011 Dec 08
1
random digits dialing during call
Hi List,
When a user is on a call, sometimes they hear digits dialing as if the
other end is randomly pressing the keypad with their face...but they
aren't. It has happened while I've been on calls also, very odd and
annoying.
Has anyone come across this on Asterisk before?
TIA,
Skyler