Displaying 20 results from an estimated 40000 matches similar to: "SIP & RTP and in-band dtmf"
2007 Feb 09
1
RFC2833 SIP trunks and DTMF
I have a telco providing DTMF inband, they say they can't provide it any
other way. This is creating headaches for me.
What is the common method for SIP DTMF? Kpml, or 2833 or inband?
My handsets don't support inband so I'm tying up some expensive
resources to convert the inband DTMF to out-of-band DTMF...
Can you recommend a vendor in US that provides SIP with DTMF in RFC
2003 May 23
4
SIP and DTMF
Hello,
I am fairly new to asterisk. I am currently using asterisk as a
more convenient sip side voicemail system.
My problem:
I have cisco 7960 phones whose out of band dtmf tones
are recognized properly(when dtmfmode=rfc2833) by asterisk but whose
in-band dtmf tones are recognised poorly(when dtmfmode=inband) . For
example 7999 comes out as 799999, 4242 comes out as 442422 ... etc
I
2003 Oct 30
1
Out Of Band DTMF and SIP
I am currently using Asterisk with G.711 codecs and in-band DTMF for
several Cisco 7960's
and an Audiocodes GW. When allowing out-of-band DTMF, I could use
voicemail menus and
anything else on Asterisk that required DTMF but I could not get the
DTMF relayed out of the
GW. Has anyone verified that this works between 2 SIP devices? If so,
I would be interested
in your settings. Also, I would
2005 May 23
0
How to detect DTMF and change if needed
I have done some searching and not sure this is even possible, but
here it goes...
**Scenario**
Let's say you have an asterisk server that you use to connect to a SIP
provider that you push your PSTN-bound calls to using g711 and
out-of-band DTMF. The SIP phones in use are Cisco 7960's and are set
to also use out-of-band DTMF. For the most part, everything works
great.
However, a few
2007 Jan 10
0
DTMF on Snom
Hi all,
I have problem using DTMF on Snom Phones (300, 320 and 360)
I read they use in preference out-of-band DTMF , and if the remote system
does not support it they default back to inband.
I would like to use DTMF as out of band , and I defined
dtmfmode=rfc2833
in the peer configuration.
Nope, I am no able to access any ouside services using DTMF;
Another kind of phones, ATCOM AT320, can be
2003 Aug 15
1
DTMF SIP
Hello list,
my case is as follows:
SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729.
When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the
keypad on the phone.
As suggested by you, I need to configure the SIP1 with out band dtmf mode ,
what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238
? do I also need to make same kind
2004 Apr 12
0
oob to inband dtmf over rtp
Are there any known problems converting dtmf from oob over iax2 to
inband over rtp/ulaw?
Obviously it works when converting to inband over pri/ulaw et al,
but how about rtp?
I've got packet traces that confirm that 2833 packets are properly
generated when I have 2833 configured for the rtp link, but the other
side seems to be ignoring those packets. So I tried inband on that
link; nothing
2009 Apr 06
1
Off-topic: SIP DTMF most supported method
Hi,
I know it is a bit off-topic, but I'd like to ask the community what is the
current most supported way to deal with DTMF?
I'm looking for an all-SIP system and I'm mostly interested in the end
devices support of the different methods (DTMF in-band audio, DTMF RTP
telephony events packets, SIP INFO, ...)
Thanks in advance.
Cesc
-------------- next part --------------
An HTML
2010 Mar 01
3
Asterisk and Cisco DTMF
Hi,
I have encountered a DTMF issue. My scenario:
Access carrier-----sip---->
Asterisk-1.4.25.1-----sip---->CiscoGW-----ISDN----->TDM Switch
the access carrier sends to Asterisk out of band (rfc2833) dtmf, Asterisk
forwards it with SIP INFO method to Cisco gateway, but on TDM switch every
digit is duplicated. Is it possible that the carrier sends inband along with
rfc2833?
Kind
2005 Jan 18
0
DTMF is being MUTED by asterisk to/from SIP channels from SIP or ZAP
I am having a problem trying to do inband DTMF passthru via asterisk.
My setup:
PSTN gateway MAXTNT v11.0 SIP (T1 PRI/NT2)
Asterisk HEAD or v1.0 makes no difference (I am using HEAD mostly)
12/10/04 and 01/17/05 (no difference)
CAC ABII-T100P to/from analog lines to/from asterisk
BTW, I have used a ABI and it works just like the ABII with asterisk.
What I am seeing is:
I make a call from a
2011 Jun 09
0
Asterisk, attended transfers and DTMF mode
Hi,
Asterisk: 1.8.4.2
I've just managed to configure attended transfers using Asterisk and
Grandstream GXP-2000 phones. The only way I've got it to work is by
using one of the out-of-band DTMF modes on the phone (either RFC or
SIP-info).
I think I can understand why - as Asterisk wouldn't be "seeing" the DTMF
tones during the call if they are inband (or am I wrong)? I
2011 Jul 05
0
DTMF between sip trunks and PRIs
Hi,
I'm looking for some advice on how to solve DTMF issues.
I have 2 boxes, one which is the connection to the PSTN (PSTN) through
PRIs and SIP trunks, and a second (PBX) which has UAs registered to
it.
We have a customer that has an existing pbx that we trunk analog lines
to using a GXW-4008.
The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF.
The issue I'm
2007 Jun 28
1
Avaya IP Office DTMF Issue
Hi
I have a client using a Avaya IP Office PBX that is taking a SIP trunk
from me terminating on a * box. It all works perfectly apart from DTMF.
Although you can hear the tones they don't seem to get recognised. I
have tried DTMF mode auto, inband, out of band and rfc2833 but no luck.
Any ideas?
Regards
Jon
--
Jon Farmer
Telford, Shropshire, UK
2007 Jun 20
1
DTMF doesn't work between Asterisk and Cisco SIP Proxy
Hi buddies,
I encountered DTMF issue when I tried to place call from x-lite to a
sip conference serice,here is the diagram.
X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service
The Call can be established,and I can hear from x-lite the prompt of
the conference,but when I input any digits,nothing happened,the
conference service did not recognize my input.At the same time,in
2010 Apr 26
0
DTMF from SIP phone to FXS/FXO
Hello,
I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my FXS/FXO lines. I am running Asterisk 1.4.21.1
In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly from SIP to FXS, but in the SIP phone I only hear a continuous noise. However, when I press any digit in the pone (FXS) I hear the DTMF tone fine in the SIP phone (the noise goes away for as
2006 Mar 16
1
RFC 2833 and SIP? DTMF? What am I not getting?
Hi again,
I am trying to get my DTMF to use RFC 2833 rather then inband, so that
I can utilize lower bandwidth codecs through my FXO.
After much tinkering I was able to get my gateway (wellgate 3701A)
configured to a point where I have some success, but no real joy.
I have configured the RTP Payload type (or RFC2833 Payload type) to
101. I don't have a clue what this means, but I took
2008 Mar 26
2
DTMF suddenly stopped working on SIP channel
Hi All,
Anyone have any idea what could cause incoming calls on a SIP channel
to no longer be able to use DTMF? DTMF on incoming calls on zaptel and
on local SIP softphones and ATAs all work fine. Nothing gets
registered in the CDR or on the console in verbose level 10, it just
times out. I haven't changed anything on my part and can't get through
to Viatalk tech support to ask them
2008 Mar 19
0
Inband SIP DTMF
I've been searching to a solution to this for a while and can't
figure it out, perhaps someone has done something similar.
I have a Cisco AS5400 sending SIP traffic via PCMU / ulaw directly to
my Asterisk (1.4.19-rc2) box. Jitter and latency are incredibly low
on my lightly loaded switched gigabit ethernet network. One Asterisk
uses Zaptel and a Digium card, and DTMF recognition
2005 Jan 05
3
Sending DTMF to PSTN problem with SIP
Dear All ~
I have * setup & running ok (with two Wildcard X100P's to PSTN). I also have
two analog phones connected into same through a SIPURA 2000. These work fine,
except that when I call out through PSTN & try to send DTMF tones to (say) a
remote PBX to dial an extension, the gain seems to go wild (high), and the
DTMF tones are not recognized at the other end.
I tried setting the
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: