well looks likes we solved it
the rtpkeepalive was set to 5 seconds on the trunk and every time asterisk
sends a rtpkeepalive a cn packet is sent
the same time a cn packet is sent asterisk loses the dtmf it was sent
On Wed, Dec 16, 2020 at 7:43 PM Israel Gottlieb wrote:
> Hi all
> i have a aste
Hi all
i have a asterisk server 16.11.1 (server A) that gets a call (leg A) and
then calls a second server (leg B) server B is a freeswitch server
the servers are configured all thru with rfc2833 for dtmf
the caller enters a number a long 15 digit number like a credit card number
or even a phone n
Have you done a wireshark capture and then seen if the DTMF is coming in
from your provider? What does the SDP show?
On Thu, Dec 5, 2019 at 12:17 AM Carlos Chavez wrote:
> What is the best way to debug DTMF on a PJSIP trunk? I have cycled
> through all available options ('rfc4733','inban
What is the best way to debug DTMF on a PJSIP trunk? I have
cycled through all available options
('rfc4733','inband','info','auto','auto_info') but my IVR does not
recognize any options from the remote end. I have also tried changing
codecs from g729 to alaw or ulaw with the same result.
I agree! I have my SBC and asterisk servers all configured with rfc2833, so
it should be ok! No need for auto mode!
Thanks again!
Cheers
Patrick
On Tue, 1 May 2018, 20:07 Joshua Colp, wrote:
> On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote:
> > Thanks very much for the reply Joshua!
> > S
On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote:
> Thanks very much for the reply Joshua!
> So I guess that setting dtmfmode=auto would be the safest choice in order
> to strip out the DTMFs from the recording, right?
> Cheers!
It should work. Personally I prefer explicit configuring instead
Thanks very much for the reply Joshua!
So I guess that setting dtmfmode=auto would be the safest choice in order
to strip out the DTMFs from the recording, right?
Cheers!
Patrick Wakano
On Tue, 1 May 2018, 19:36 Joshua Colp, wrote:
> On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote:
> > H
On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote:
> 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
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://com
Asterisk is in version 14.7.1. One end is a SIP Trunk to another
Asterisk, the other end a home-made SIP phone. SIP INFO requests are
coming from the other Asterisk.
Both endpoints use chan_sip with "dtmfmode" set to "info".
This is not strictly speaking a one-to-one setup since we're connecti
Hello Jean,
1. Can you describe a bit further how both ends of the above call were both
made of and configured ?
DTMF receiving is Asterisk/SIP channel but which version ?
Is the other end a SIP phone or a SIP trunk ?
2. Do you observe such behaviour in a one-to-one setup (one end emits, the
othe
Hello,
I think there is an issue when DTMF are handled with SIP INFO and direct
media is enabled.
When I receive a SIP INFO, the logs tell me that a "DTMF begin" is
generated, but no related "DTMF end" is generated, unless the call is
ended. Here is an excerpt of the logs :
*--- SIP INFO r
Hi,
I have a Asterisk running in Amazon AWS with an IVR configured. The problem
I'm having is I'm not getting DTMF from mobile phones. The Landlines works
without any issues. I have configure DTMF to rfc 2833. I checked with dtmf
debug but I'm not receiving dtmf from mobile devices. please let me
Carlos Chavez wrote:
I had an old Asterisk installation die recently and we decided to
upgrade to Asterisk 13 to replace the old server. Everything seems to be
working with PJSIP but there is one issue. Asterisk talks to a
callmanager via a SIP trunk and send calls to PSTN (another country).
Most
I had an old Asterisk installation die recently and we decided to
upgrade to Asterisk 13 to replace the old server. Everything seems to
be working with PJSIP but there is one issue. Asterisk talks to a
callmanager via a SIP trunk and send calls to PSTN (another country).
Most of the call
On 9/10/2015, at 5:16 AM, Jamie Rees wrote:
>
>
> I understand this is DTMF talkoff
>
>
> My question is how do people running SIP phone systems mitigate against this?
My personal answer to this question has been to completely avoid the use of any
ATAs at all. Since taking that approach I
Hi all,
I am still receiving reports from some users that calls they make or receive
contain loud deafening beeps that can last a couple of seconds. I understand
this is DTMF talkoff and is being triggered because the phone interprets
speech as a key press (say if someone is pressing 1 at an IV
riginal Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: 08 July 2015 10:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue
Indeed, thanks.
I&#x
Discussion'
Subject: Re: [asterisk-users] DTMF issue
You probably have to reload asrerisk after making the change.
Thomas M. Peters | Systems Administrator | tpet...@mcts.org
Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org
>>> "Jamie Rees" 7/7/2015 3:53 PM >
0 Span 2"
group=0,12
context=from-pstn
switchtype = euroisdn
signalling = pri_cpe
channel => 32-46,48-62
context = default
group = 63
Thanks again,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of To
t: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] DTMF issue
In my humble opinion, adjusting this setting will (for you) do nothing,
since you don't use the dahdi channels for transport.
See this discussion, whi
channel => 32-46,48-62
context = default
group = 63
Thanks again,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [
s-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 19:14
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF issue
It's called DTMF Talk-off. We have it too. Seems worse when talking to
mobile phones but it hap
It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile
phones but it happens at random on many external calls. If this happens to you,
especially on voice peaks (when the outside party said a particularly loud
syllable) then you probably have DTMF talk-off.
I think it's
On 7/6/15 5:53 PM, Jamie Rees wrote:
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an
Asterisk platform where several users hear loud, random beeps during
calls to external recipients. The noises are akin to button press
tones, are very loud and a significant a
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees
Sent: Monday, July 06, 2015 5:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issue
Hello folks,
We have an issue with several Cisco SPA512G phones
Hello folks,
We have an issue with several Cisco SPA512G phones connected to an Asterisk
platform where several users hear loud, random beeps during calls to
external recipients. The noises are akin to button press tones, are very
loud and a significant annoyance.
I've tried changing the DTMF to
Thank you Mathew,
We tested the feature flag workaround and it worked.
We opened a ticket - Asterisk-24459.
If you need any information please get back to us and we will do our best.
Thanks again,
Yaron.
On Mon, Oct 27, 2014 at 3:48 PM, Matthew Jordan wrote:
>
>
> On Mon, Oct 27, 2014 at 1:20
On Mon, Oct 27, 2014 at 1:20 AM, Yaron Nachum
wrote:
> Hello Mathew,
> Thank you for the reply.
>
> I will open an issue and send debug information.
>
> Can you explain more about the workaround? A reference to the
> documentation would be fine.
>
>
>
Sure - really, what you are running into is a
Hello Mathew,
Thank you for the reply.
I will open an issue and send debug information.
Can you explain more about the workaround? A reference to the documentation
would be fine.
Thanks again,
Yaron.
On Sun, Oct 26, 2014 at 10:46 PM, Matthew Jordan wrote:
>
>
> On Sun, Oct 26, 2014 at 3:22 A
On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum
wrote:
> 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 INVI
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
ists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Tuesday, June 17, 2014 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DTMF transmitting letter A
Dear list,
maybe not really an Asterisk question, but... all m
Dear list,
maybe not really an Asterisk question, but... all my users dial in via
PSTN (via SIP DIDs) and enter a target number via DTMF through my
Asterisk 1.4. Out of about 150,000 calls per month I see on average
about 1 call per month where an arbitrary caller enters the letter 'A'
via DT
Hello List,
I am facing some issue while passing DTMF (RFC2833 set globally in
sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two
users tries to pass DTMF simultaneously at the same time from their phones
only one DTMF is detected in asterisk and broadcasted to other users
Hi all.
I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port).
When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not
receive DTMF from caller while the voice is playing. But if user waits to the
end of playing voice, there is no problem.
I`m using As
Hello.
Scenario: 9 servers connectec to each other over IAX trunks. Users
used to call to remote extensions and remote conferences (meetme) via
IAX.
Problem: all extensions from one server (just one) when try to attend
remote conferences had problems with PIN validation. If they use their
local c
Anyone see this before?
I have a main Asterisk box 11.4 connected to Windstream via SIP trunks in my
colo.
So as a did comes in they are routed to appropriate customers, in this case
another asterisk 11.4 box.
All is working well with the exception of DTMF. Losing the last digits so say
someon
Let me try with dtmfmode as auto...
On 28 May 2013 19:32, "Asghar Mohammad" wrote:
> work around was block dtmf.
> set wrong type of dtmf in incoming trunk.
>
>
> On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> So any resolution for this?
>>
>> I su
work around was block dtmf.
set wrong type of dtmf in incoming trunk.
On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> So any resolution for this?
>
> I suspect it could be related to RE INVITE
>
>
> On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote
So any resolution for this?
I suspect it could be related to RE INVITE
On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad wrote:
> i had this in past there was an ATA configured to send 9 at the end of
> dialing in my case.
>
>
> On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
> gopalakrishna
i had this in past there was an ATA configured to send 9 at the end of
dialing in my case.
On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:
> Hi,
>
> I am receiving DTMF without any reason after call establishment.
>
> The log as follows, and I suspect some
Hi,
I am receiving DTMF without any reason after call establishment.
The log as follows, and I suspect something related to directmedia,
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is
making progress passing it to SIP/MAN-000a4b48
[May 17 00:33:35] VERBOSE[4238] app_di
On Wed, Feb 20, 2013 at 10:49 AM, James Lamanna wrote:
> Hi,
> I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end
> the recording on the recording itself.
> Is there an easy way to truncate the last 200ms of the recording or so to
> eliminate this?
> The DTMF is coming in thr
Hi,
I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the
recording on the recording itself.
Is there an easy way to truncate the last 200ms of the recording or so to
eliminate this?
The DTMF is coming in through rfc2833 and not inband.
Thanks.
-- James
--
__
digits, they are displayed and logged.what's missing?
Regards
--- On Wed, 11/28/12, Joshua Colp wrote:
From: Joshua Colp
Subject: Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Date: Wednesday, No
mohammad aliasgari wrote:
Dear all,
Hola,
having verbose level 5, and enabling dtmf logging in
/etc/asterisk/logger.conf
console => notice,warning,error,debug,dtmf
I receive dtmf detected, in a SIP-PSTN call, as follows
Why don't I receive DTMF that are dialed by a PSTN phone?
How is
Dear all,
having verbose level 5, and enabling dtmf logging in /etc/asterisk/logger.conf
console => notice,warning,error,debug,dtmf
I receive dtmf detected, in a SIP-PSTN call, as follows
[code][Nov 28 16:12:19] DTMF[2532]: channel.c:2351 __ast_read: DTMF begin '1'
received on SIP/16001-0
Necati Demir wrote:
Hello,
Hola,
The service provider wants me to setup dtmfmode to rfc2833 and dtmf
payload to 101.
I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload?
Asterisk already uses payload 101 for RFC2833 so you should be fine with
dtmfmode=rfc2833
Chee
Hello,
The service provider wants me to setup dtmfmode to rfc2833 and dtmf payload
to 101.
I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload?
--
Necati DEMİR
--
_
-- Bandwidth and C
Jakob Hirsch wrote:
Hello everyone!
Hola,
We use Asterisk for various services like voicemail. Our SIP clients
usually use rtp events (rfc2833) for DTMF, which works just fine and
independent from the codec (g711 vs. g726 etc.).
Now we noticed there are some SIP clients that announce telepho
Hello everyone!
We use Asterisk for various services like voicemail. Our SIP clients
usually use rtp events (rfc2833) for DTMF, which works just fine and
independent from the codec (g711 vs. g726 etc.).
Now we noticed there are some SIP clients that announce telephone-event
in their SDP, but send
Sorry the attachment was too big. here is link:
http://www.2shared.com/file/Ola640Pn/doubledigit.html
On Fri, Oct 12, 2012 at 9:24 AM, SamyGo wrote:
> Why am I feeling like I'm the only one here who is not able to see any
> pastebin link or attachments in this thread !
>
>
--
__
Why am I feeling like I'm the only one here who is not able to see any
pastebin link or attachments in this thread !
On Fri, Oct 12, 2012 at 6:18 PM, Vik Killa wrote:
> The trace is attached 3 emails back.
>
> --
> _
> -- Bandwi
The trace is attached 3 emails back.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
as
Well like I said before and you increased my doubt as well by saying that
this happens from callers using E-link internet. Can you share the trace !
On Fri, Oct 12, 2012 at 5:24 PM, Vik Killa wrote:
> Any ideas?
>
> On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa wrote:
> > Call was to 7167436110
>
Any ideas?
On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa wrote:
> Call was to 7167436110
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Th
Can you share your pcap trace !
On Thu, Oct 11, 2012 at 5:16 PM, Vik Killa wrote:
> Only callers calling from Earthlink internet connection
>
> On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly wrote:
> > Is this happening for all callers, or just iPhone callers?
> >
> > --Don
> >
>
> --
> ___
Only callers calling from Earthlink internet connection
On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly wrote:
> Is this happening for all callers, or just iPhone callers?
>
> --Don
>
--
_
-- Bandwidth and Colocation Provided by http
-Commercial Discussion
Subject: Re: [asterisk-users] DTMF digits are coming through twice
I'm not sure I follow, the packet capture on the asterisk server shows
double digits being entered. Does that mean it's the source?
On Wed, Oct 10, 2012 at 11:55 AM, SamyGo wrote:
> Hi,
>
After comparing packet captures of good and bad calls. It looks like
the double digit is coming from rfc2833 and dtmf inband. It looks
like the inband tone is splitting the rfc2833 in two? Is there some
way to resolve this???
On Wed, Oct 10, 2012 at 12:28 PM, Vik Killa wrote:
> I'm not sure I f
I'm not sure I follow, the packet capture on the asterisk server shows
double digits being entered. Does that mean it's the source?
On Wed, Oct 10, 2012 at 11:55 AM, SamyGo wrote:
> Hi,
>
> Not exactly a solution, but I'm sure you must've taken pcap traces of a few
> such sample calls. See in the
Hi,
Not exactly a solution, but I'm sure you must've taken pcap traces of a few
such sample calls. See in their RTPs that you are receiving repeatedly same
RTPs which will tell you that any DTMF packet is coming in twice by the
source or not !
just one such simple pcap will help you identify at wh
I've been running an Asterisk server (1.6.2.17.2) for over a year
without any major issues. All of a sudden people are unable to login
to their voicemail because Asterisk is seeing DTMF twice for each
digit the caller pushes. We've noticed the problem only consistently
happens to callers from speci
On 9/15/2012 6:28 PM, Matthew Jordan wrote:
> - Original Message -
>> From: "Vladimir Mikhelson"
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>
>> Sent: Saturday, September 15, 2012 1:11:14 PM
>> Subje
- Original Message -
> From: "Vladimir Mikhelson"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Saturday, September 15, 2012 1:11:14 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
> Can you please quo
> And just to make sure. In both scenarios, normal digit press
> and prolonged digit press, you did not reproduce the problem
> we are discussing with X-Lite. Is that correct?
>
Correct, everything with X-Lite 3.0 and asterisk 1.8.16.0 worked correctly
with short, normal and long key presses
On 9/15/2012 5:16 PM, Alec Davis wrote:
>
>>> [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end
>> '4' received on
>>> SIP/alec-0009, duration 1660
>>>
>>>
>>>
>> Alec,
>>
>> Interestingly in your log DTMF durations are even greater
>> than in my
> > [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end
> '4' received on
> > SIP/alec-0009, duration 1660
> >
> >
> >
> Alec,
>
> Interestingly in your log DTMF durations are even greater
> than in my original sampling. Well, maybe my "duration"
>>
>> Sent: Saturday, September 15, 2012 11:41:23 AM
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>> Please take a look at the case
>> https://issues.asterisk.org/jira/browse/ASTERISK-20424?actionOrder=asc
>> I uploaded the PCAP captur
- Original Message -
> From: "Vladimir Mikhelson"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Saturday, September 15, 2012 11:41:23 AM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
>
&g
>> Hopefully the initial poster still has the configuration to
>> produce the files for you.
>>
>> Are you saying the DTMF logs I attached do not provide enough
>> evidence to support the theory of the DTMF length being the
>> cause of this issue?
>>
>> -Vladimir
>>
> Vladimir,
> What was
.
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] DTMF digits falsely detected
>>
>>
>> On 9/14/2012 11:04 PM, Matthew Jordan wrote:
>>> - Original Message -
>>>> From: "Vladimir Mikhelson&q
sion
> Subject: Re: [asterisk-users] DTMF digits falsely detected
>
>
> On 9/14/2012 11:04 PM, Matthew Jordan wrote:
> > - Original Message -
> >> From: "Vladimir Mikhelson"
> >> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
On 9/14/2012 11:04 PM, Matthew Jordan wrote:
> - Original Message -
>> From: "Vladimir Mikhelson"
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>
>> Sent: Friday, September 14, 2012 10:39:30 PM
>> Subje
- Original Message -
> From: "Vladimir Mikhelson"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, September 14, 2012 10:39:30 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
>
>
>
On 9/14/2012 10:11 PM, Matthew Jordan wrote:
>
> - Original Message -
>> From: "Vladimir Mikhelson"
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>
>> Sent: Friday, September 14, 2012 9:24:41 PM
>> Su
- Original Message -
> From: "Vladimir Mikhelson"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Friday, September 14, 2012 9:24:41 PM
> Subject: Re: [asterisk-users] DTMF digits falsely detected
>
>
rs@lists.digium.com
>> Subject: [asterisk-users] DTMF digits falsely detected
>>
>> Hi,
>>
>> I have a context that basically does:
>>
>> Wait(1)
>> Background(message)
>> WaitExten(10)
>>
>> _6XX,1,DoSomething
>>
>> The prob
rs@lists.digium.com
>> Subject: [asterisk-users] DTMF digits falsely detected
>>
>> Hi,
>>
>> I have a context that basically does:
>>
>> Wait(1)
>> Background(message)
>> WaitExten(10)
>>
>> _6XX,1,DoSomething
>>
>> The prob
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
> Sent: Saturday, 15 September 2012 8:45 a.m.
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] DTMF digits falsel
Hi,
I have a context that basically does:
Wait(1)
Background(message)
WaitExten(10)
_6XX,1,DoSomething
The problem is that when I reach this context and press some digits (eg.
6566604) then I can see in the log that Asterisk reads 6655666.
So it's actually reading the digits twice.
How can
Up?
2012/8/20 Luis H. Forchesatto
> Thanks for your answer.
>
> The logs where posted at pastebin, here the links:
>
> - Working Phone: http://pastebin.com/q3pHcwna
> - Not working phone: http://pastebin.com/iiCHPMmn
>
>
> 2012/8/20 Rusty Newton
>
>> On 8/20/2012 7:19 AM, Luis H. Forchesatto wr
Thanks for your answer.
The logs where posted at pastebin, here the links:
- Working Phone: http://pastebin.com/q3pHcwna
- Not working phone: http://pastebin.com/iiCHPMmn
2012/8/20 Rusty Newton
> On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
>
>> Hi
>>
>> I've got a little issue with DTMF/I
On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
Hi
I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of
ATA on the network who autenticate the phones: Linksys PAP2,
Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the
VoIP server at the same network all with g
Hi
I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA
on the network who autenticate the phones: Linksys PAP2,
Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP
server at the same network all with g729 codecs and rfc2833 for the DTMF.
Making calls via
*David Matías Hernández didi you have any luck?*
*I have the same problem.*
--
_
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> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Noah Engelberth
> Sent: Thursday, August 02, 2012 1:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [as
risk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] DTMF transmission problem
> >
> > On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote:
> > > I am having difficulties with customer-bound DTMF being very short &
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Shaun Ruffell
> Sent: Thursday, August 02, 2012 11:06 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [as
On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote:
> I am having difficulties with customer-bound DTMF being very short
> & clipped off (and basically unusable, as systems on the customer
> side aren't recognizing the DTMF digits, and I can barely tell
> that DTMF is there when I list
I am having difficulties with customer-bound DTMF being very short & clipped
off (and basically unusable, as systems on the customer side aren't recognizing
the DTMF digits, and I can barely tell that DTMF is there when I listen on a
handset).
My system set up as follows:
PSTN <--> Metaswitch
Nevermind,
I checked the code, and A* is not using the "F" option in MeetMe for
Page(), so it's not working by default.
Attached is a patch which fixes the problem for me, if anyone needs it.
Matteo
Il 11/02/2012 13:53, Matteo Fortini ha scritto:
Noone knows that? Where/whom could I ask?
Tha
Noone knows that? Where/whom could I ask?
Thanks
Il 10/02/2012 12:30, Matteo Fortini ha scritto:
Hi,
I'd like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a MeetMe() c
Hi,
I'd like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is connected
to the prev
o in that case you need to observer the call flow in Server-B, i.e what is
the length of sound file playing. what DTMF it requires etc etc and once
you detect the call flow for a successful IVR traversal then mimic the
behaviour of the call from Server-A.
Thats all you can do.
Think of it exactly t
In server B if I use SendDTMF then it means I am changing programming at
server B. Actually I don't have right or permission to change programming
in server B.
otherwise your suggestion is best for channel base communication.
On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind wrote:
> Easy, use Re
Easy, use Read() to capture the incoming DTMF from Server-B
Server-A <> Server-B
Initiate-Call -> AnswerCall()
SendDTMF(5)--> Read()
Read()<-SendDTMF(4)
SendDTMF(3)--> Read()
Read()<
I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.
*A server code is below:-*
exten => 43689956,1,Answer()
same => n,Wait(5)
same => n,SendDTMF(1)
same => n,NoOp(== ${CHANNEL(state)}==> state)
same => n,wait(2)
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I do
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