Displaying 20 results from an estimated 20000 matches similar to: "Stange behavior with g729 and DTMF"
2005 Jun 23
5
INBAND DTMF G729 ASTERISK
Hi all.
Why don't Asterisk support inband DTMF with G729? Is there a way to do
that!?
Are you using RFC2833? Doesn't it a security hole?
Thanks.
Denis.
2006 Feb 28
2
Sipura SPA-3000 and PSTN dtmf
Greetings,
What is the recommended settings for using SPA-3000's FXO port for
dialing out to PSTN in regard of the DTMF?
The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports
registered to the Asterisk box with unique username/passwords.
The inbound PSTN DTMF works excellently, e.g. people calling from PSTN
into the * box are able to pick IVR items with DTMF reliably.
The
2004 Jan 22
1
Grandstream transfer solution + DTMF translation possible?
The solution to the problems with the Grandstream 1.0.4.39 firmware is
to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to
work.
However, this presents another problem. When I'm using g729 to place
a call, I get the warning "Unable to process inband DTMF" because
inband is not supposed to work with g729 (although it does seem to
work when I've tried it so far).
2009 Sep 25
3
disable dtmf on SIP peer
Hello,
I have one problem and I need to disable dtmf (disable rfc2833, info and
inband) on one (other peers must support dtmf) SIP peer . Is it possible?
Workaround would be use g729 codec with dtmfmode=inband.
Maybe there is better solution?
Thanks for help.
--
Pagarbiai / Best Regards,
Giedrius Augys
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2005 Aug 16
1
Issue with DTMF Tones - Codec Issues
Topology:
PSTN<-T1 PRI->NEAX2400<-T1 PRI->Cisco 3825<-Ethernet-> Asterisk VoIP server
When I make a call to a VoIP user from the PSTN, the call gets routed
through the PBX, and Cisco. Because of that the DTMF tones are passed
inband, which I can hear on the VoIP end of the call. However, I have
one extension on asterisk set up so that I can check voice mail when
away from my
2004 Apr 29
1
SIP DTMF signaling to VM
Hello all,
I am trying to get voicemail to work for sip phones using g729, yes I did
buy the licenses.
I can get it to work using other codecs like G711 and dtmfmode=inband.
But when I make a call using g729 I get "Apr 29 09:47:14
WARNING[1209214400]:
dsp.c:1452 ast_dsp_process: Unable to process inband DTMF on 256 frames"
on my console.
I can't seem to get anything to work when
2005 Oct 03
0
Weird Problem - SIP/POLYCOM/DTMF
Hi All,
I've been having a weird issue with one of my servers and it's Asterisk
installation.
The server is running Slackware, and Kernel 2.6.12. I'm running the
latest CVS-HEAD
edition of *. I also have 4 other asterisk servers with the same
software configuration,
but different hardware and they have no issues like this.
I have one block of users behind a PF firewall at the
2007 Jun 22
10
inband DTMF for g729
Does anybody know why Asterisk does not support inband DTMF for G.729?
Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system.
Any suggestion to solve this problem?
Gary
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2004 Apr 09
2
g729 and dtmf
HI,
quick and simple question: is it possible to use inband dtmf with g729?
What I would like to do is to have sip clients connected to asterisk and a zaptel
card to make pstn phone calls.
My concern is to allow sip users to use digits for call destinations that
do require menu actions while retaining low bandwith occupation.
Tnx !
--
Best regards,
Alessio
2010 Jul 08
2
DTMF issues/redial tones with rfc2833
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip trunking for them
not PRI's, one of our system is giving a problem of dtmf (rfc2833), like
when we dial the number that have IVR and enter the extension or access
code, it some time takes it and some times does'nt recognize the digits
dialled. We also tried auto and info for dtmf but could not get the dtmf to
work reliably, can
2004 Apr 13
1
SIP->h323 problem DTMF
I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711).
But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message:
-- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack
--
2014 Oct 26
1
DTMF behavior in asterisk 12 with PJSIP
Hello all,
We have recently upgraded some of our services to Asterisk 12 with PJSIP.
We have 2 issues related to DTMF:
1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF
settings according to the incoming INVITE - RFC2833 or inband. The is no
such settings in PJSIP. Do you know is there is a plan to develop it?
2. When we setup 2 peers, one RFC4733 and the other inband,
2004 Nov 23
4
oh323/g729 and DTMF
Hi everyone,
Could somebody enlighten me on this one? I have
configured my asterisk to run on oh323 using codec
g729. Incoming calls are working okay. But the thing I
want to work is say pressing some options, say dial 1
to go to voicemail or dial a certain number to dial a
specific extension.
I have a config for this and tried calling from a
normal PSTN and is working. But i just can't seem
2004 Nov 03
5
FireFly Problems
How come FireFly doesn't give me an Inband DTMF option? Only RFC2833 and
Info. RFC2833 is the default, so I left it that way. I also unchecked all
the codecs except g711ulaw to force that codecs usage. However, when I go to
place a call, I get this:
Nov 3 13:18:44 WARNING[53641241]: dsp.c:1468 ast_dsp_process: Inband DTMF
is not supported on codec G.711 u-law. Use RFC2833
Nov 3 13:18:44
2009 Jul 20
0
No subject
device somewhere in your communication path, and since voice is picked up as
DTMF, some device is also set to listen for inband DTMF.
What is the origination source of incoming calls to your system?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote:
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip
2005 Jan 05
4
Broadvoice / * re-register issues
HELP!
Ok, so I have the following SIP.CONF:
[general]
context=default
port=5060
bindaddr=10.1.1.200
externip = XX.XXX.XX.XX
localnet=10.0.0.0/255.0.0.0
disallow=all
allow=ulaw
allow=g729
allow=g726
allow=alaw
register =>
##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
2006 Feb 12
1
dtmfmode=auto, but doesn't work
Hello everybody,
I have set dtmfmode=auto in my sip.conf, but that does not work and I still got the following message:
WARNING[4980]: dsp.c:1422 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833
According to http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+dtmfmode:
auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones
2008 Oct 03
1
DTMF issues...
I am having a big problem with DTMF. I have a customer using an
Asterisk 1.4.20.1 system with ZTDUMMY as the timing source. The problem
is that when they dial into a conference bridge or IVR where they have
to enter a code they always get an error. Either some numbers are
duplicated or missing.
They use Teliax for calls to the USA and Protel in Mexico. Both
carriers have the same problem so
2005 Aug 05
0
ATA186 can not generate dtmf
Hello:
I have problems sending dtmf signal to an ATA186 my configuration is:
ATA186 --> asterisk --> ATA186 --> FXS to FXO Converter --> PSTN
The ATA186 are set to send dtmf RFC2833, but it seems that the ATA186 can't
generate dtmf so I can dial to a PSTN number.
Is there a setting that can fix my problem, inband dtmf does not work
because I'm using G729 codec
Thanks
2007 Aug 28
2
G729 Confusion
I've purchsed 5 g729 licenses from digium, but am a little confused
about why things are acting the way they are.
When I do show g729 I see:
0/0 encoders/decoders of 5 licensed channels are currently in use
My sip.conf starts out:
[general]
disallow=all
allow=ulaw
allow=g726
then my hosts look like this:
[6016716]
username=6016716
accountcode=75415
type=friend
secret=obsurified
qualify=yes