Displaying 20 results from an estimated 20000 matches similar to: "voiceglo.com and dtmf"
2004 Jan 26
0
I need some clarification on DTMF
Hello all,
I have a SIP provider that is doing the PSTN bridging for me
(voiceglo.com). I understand the inband, info, etc for DTMF on *my*
network, but what about what they're sending me? Is there anything that
could prevent them from sending me DTMF information? The Asterisk
system is currently NAT'd (no special rules, just a Linux router) and
incoming and outgoing calls work
2005 Jan 05
2
Glophone/Voiceglo and Asterisk
<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P>
<P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P>
<P>I've tried copying the config in this listing with no success. </P>
2003 Dec 23
0
Voiceglo SIP configuration
The call quality is really pretty good. I think better than Vonage over
an FXO bridge. If you are looking for a home provider with direct SIP
support and local phone numbers this is a good choice. If anyone has
questions or comments about my configuration please pass them along. I
have noticed that if you don't put fromuser=phone# then the extension
caller id passes through. Also the
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 Jan 30
1
Cameron Palmer / voiceglo
I found a message in the archives from Cameron Palmer on 23 Dec regarding
his voiceglo SIP configuration. Unfortunately (for me), the archive has
his email address removed.
So, Cameron -- or anybody else using voiceglo with their * box -- please
reply to me so that I can get your email address and ask you a question
about your setup.
Thanks,
Greg
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 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
2020 Aug 26
0
Inband DTMF not detected - bug or config error?
Hi,
we have an Asterisk server basically passing on calls using the Dial
application. In the pjsip endpoint settings, the dtmf_mode is set to audio.
This works with most calls. However, there is a scenario where DTMF tones
don't get forwarded the way I would expect them to get forwarded.
A: Caller without RfC4733 support
B: our Asterisk, version 17.6.0
C: Another Asterisk, with RfC4733
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
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
2009 Jan 29
0
[asterisk-dev] DTMF queuing
[moving to asterisk-users by request]
On Tue, Jan 27, 2009 at 12:56 AM, John Todd <jtodd at digium.com> wrote:
>
> On Jan 26, 2009, at 7:38 PM, James Lamanna wrote:
>
>>> On Jan 26, 2009, at 8:53 PM, James Lamanna wrote:
>>>
>>>> Hi,
>>>> Is it just me, or does DTMF queuing not work properly?
>>>> I'm consistently faced with
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
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
2003 Sep 13
2
SJphone DTMF?
Hi. I have sjphone installed on windows and working
except for dtmf. I read the docs for sjphone and it
uses inband dtmf. I configired dtmfmode=inband but it
still does not recognize it. Someone on the lists
said that inband only works using alaw or ulaw but i
tried only allowing that too but still no go. Hmm..
any other ideas? I can't get any other client to work
on windows :-/
I
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).
2015 Jan 17
1
Fwd: Asterisk pjsip auto dtmf mode
Hello Asterisk Users,
I have been looking for similar auto dtmf mode implementation on pjsip, but
didn't see it coming, so I decided to give it a try.
My basic plan was to do it as simple as possible with minimum changes
because I am not familiar with all Asterisk code. My idea is to use rfc4733
settings, but when going over the codecs check if telephone-event appear
and if not set the dtmf
2005 Jun 21
1
GSM ISDN gateway to quadbri NT port under bristuffed Asterisk - where to set gain (I have unreliable inband dtmf recognition )?
Hi,
I'm getting unreliable dtmf recognition (it works fine for 4-5 digits,
errors (duplicates) on more), when transferred inband from gsm gateway to NT
port of quadbri under bristuffed Asterisk.....
Since Asterisk is claimed to have good dtmf recognizer, I suspect there are
some settings to workarouned... I've tried dtmf relax, but didn't help, so I
suspect gain settings....
Is
2006 Mar 24
1
[1.2.5] DTMF not being set correctly (RESEND)
I apologize if this gets posted twice. Tried once about 5 or so hours
ago, and still have not seen the message on the list....
--------------------------------
I am having trouble getting DTMF mode to be set to inband on incoming
calls.
I have the following set, and for some reason the connection is still
negotiated with rfc2833.
[outbound]
type=friend
secret=XXXXXXX
username=XXXXXXX
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
2003 Dec 01
8
VoiceGlo
Hi,
VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll
Take a loock on http://www.voiceglo.com/
The softphone is IAX :)
Best regards,
Chris HARIGA
Techselesta Inc.
http://www.techselesta.com/
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