On Sun, Sep 18, 2011 at 07:51:43PM -0400, Zeeshan A Zakaria wrote:
This DTMF problem has always been there and there is no real solution
for it, other than using those expensive PBX systems like that from
Avaya, Cisco, etc. This problem happens when you are sending inband
DTMF tones. Via
19 sep 2011 kl. 01:51 skrev Zeeshan A Zakaria:
This DTMF problem has always been there and there is no real solution for it,
other than using those expensive PBX systems like that from Avaya, Cisco,
etc. This problem happens when you are sending inband DTMF tones. Via
softphone you are
From time to time, I have a DTMF problem. The asterisk cannot recognize my
handset key pressed if I press 9 to start with. However, if I press with 6,
it is ok.
On the other hand, if DMTF key is generated from softphone, it is ok.
My dialplan is as follow
exten = 1002,1,Answer
exten =
This DTMF problem has always been there and there is no real solution for it,
other than using those expensive PBX systems like that from Avaya, Cisco, etc.
This problem happens when you are sending inband DTMF tones. Via softphone you
are sending out-of-band DTMF which is basically SIP
Hey there,
I don't think that its DTMF mode issue ! OP say pressing 9 asterisk ignores
while pressing 6 is OK. Using expensive PBX solutions should be never be the
first priority.
So I'd a similar experience in some asterisk version when I used to enter 2
asterisk always took 3-4 seconds to do
I have a simple dialplan. Using the read cmd, I ask caller for his passcode.
If the caller is calling from an plain old analog phone, his dtmf is not heard
until the prompt stops playing. SIP phones work correctly. I've trird
everything I found searching the net. I've tried all the dtmfmode.
First, the scenarios:
Call placed from Boston to locally configured Asterisk Meetme extension:
Cisco 7941 -SCCP- Cisco 2821(CME,Boston) -SIP- Asterisk(Boston)
Call placed from Boston to European Asterisk Meetme extension:
Cisco 7941 -SCCP- Cisco 2821(CME,Boston) -SIP- Cisco
2821(CME,Europe)
On Thu, Jun 4, 2009 at 9:34 PM, Phillip Heller phel...@me.com wrote:
Cisco 7941 -SCCP- Cisco 2821(CME,Boston) -SIP- Cisco
2821(CME,Europe) -SIP- Asterisk(Boston)
debugging enabled on Asterisk, I see that I often get duplicate DTMF
entries. So where I might have dialed 1234#, Asterisk sees
So the PBX in Europe has a local extension and DID configured for the
Asterisk MeetMe such that users in Europe have a local number to
dial
Placing the call from Boston to the European extension is only to
duplicate and hopefully solve the problem.
When I came aboard this company, it
On Thu, Jun 4, 2009 at 10:17 PM, Phillip Heller phel...@me.com wrote:
I have tooled around with the various dtmf-relay options, though to no
positive effect. I'll keep playing with it tomorrow. If you happen
to think of anything else, I certainly appreciate the input.
If you haven't already
Dear Sir,
I have the following Scenario:
1- I have a DID number from Voxbone mapped to my asterisk server with RFC
2833 protocol used for DTMF
2- On asterisk Server I configured an incoming peer that receives calls from
VoxBone and send calls to a2billing context as follow:
*sip.conf*
On Oct 2, 2008, at 5:27 AM, michel freiha wrote:
Dear Sir,
I have the following Scenario:
1- I have a DID number from Voxbone mapped to my asterisk server
with RFC 2833 protocol used for DTMF
2- On asterisk Server I configured an incoming peer that receives
calls from VoxBone and send
Hi;
This problem I suffered from it for long time, it needs some work from ur side
to resolve it, I will give u all the factors that will help u to fix it, and u
need to work on it one after one in care:
1) Disable x-windows, gnome, ... at least for all testing. This is very
important to be
： Re: [asterisk-users] DTMF Problem
for UDP
tcpdump -nnXs 0 udp -i eth0 -w name.cap
Btw, a pcap file (created on a linux server using tcpdump) capturing the
RTP(udp) traffic opened up in wireshark, wireshark doesn't really
format(or recognize) the packets as RTP, unlike the capture done live
from
Users Mailing List - Non-Commercial Discussion;
Asterisk Users Mailing List - No
*抄送：*
*主题：* Re: [asterisk-users] DTMF Problem
At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote:
Hi，
Could you capture the the UDP package
How is this done?
in all of your server, Asterisk A, Asterisk B, ser
.
木木
2007-11-16
发件人： Benjamin Jacob
发送时间： 2007-11-16 12:55:51
收件人： Asterisk Users Mailing List - Non-Commercial Discussion
抄送：
主题： Re: [asterisk-users] DTMF Problem
for UDP
tcpdump -nnXs 0 udp -i eth0 -w name.cap
Btw, a pcap file (created
：
主题： Re: [asterisk-users] DTMF Problem
At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote:
Hi，
Could you capture the the UDP package
How is this done?
in all of your server, Asterisk A, Asterisk B, ser, Asterisk C.
And you can find that server who lost the DTMF (RTP EVENT).
--
Amy
Hi
Here is my setup:
USER -- PSTN - Asterisk A IAX2 Trunk Asterisk
B - SER Asterisk C
I'm not able to receive DTMF passed by USER on Asterisk C.
All my asterisk boxs are configured with same DTMF type (auto) but no luck.
Please help on this issue.
Users
抄送：
主题： [asterisk-users] DTMF Problem
Hi
Here is my setup:
USER -- PSTN - Asterisk A IAX2 Trunk Asterisk
B - SER Asterisk C
I'm not able to receive DTMF passed by USER on Asterisk C.
All my asterisk boxs are configured with same DTMF type
Kumar
·¢ËÍʱ¼ä£º 2007-11-15 20:30:45
ÊÕ¼þÈË£º Asterisk Users Mailing List - Non-Commercial Discussion; SER Users
³ËÍ£º
Ö÷Ì⣺ [asterisk-users] DTMF Problem
Hi
Here is my setup:
USER -- PSTN - Asterisk A IAX2 Trunk Asterisk
B - SER Asterisk C
I'm
Hello All,
Does anyone knows a good carrier who can pass DTMF tone while doing Call
Back? Currently, the Call Back system works within US, but as soon as
international users tries to enter phone number the system does not
understand the tones.
I tried to change the sip config to inband, auto,
Hello,
I got a little problem here. My Setup is like this,
Service provider - ser - asterisk - termination provider.
I have a sip trunk on asterisk pointing to ser,
allow=ulawalaw
context=from-trunk
disallow=all
dtmfmode=inband
host=ser ip address
insecure=very
type=peer
This works with
I am having the same issue with 1.2.17. Only certain toll free numbers do we
have issues with.
- Original Message -
From: ismir saljic
To: [EMAIL PROTECTED]
Sent: Thursday, April 12, 2007 5:42 PM
Subject: [asterisk-users] DTMF problem with inbound calls on Toll-Free number
Hi all,
I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call on
Toll-Free number asterisk accept DTMF digits but dial only first in context.
Per instance:
When i press 1 it is OK,but when i try to dial extension 700 asterisk dial only
first digit(1) and i receive from
From: ismir saljic [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 07:42:13 -0700 (PDT)
Hi all,
I have asterisk 1.2.13 and problem is about DTMF.When i have incoming call
on Toll-Free number asterisk accept DTMF digits but dial only first in
context.
Per instance:
When i press 1 it is OK,but when i
You can use the following to display what you receive from user (dtmf):
exten= 1,1,Read(test)
exten= 1,2,NoOp(DTMF Received: $test)
exten= 1,3,Hangup
On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote:
I upgraded to 1.4.1 and my DTMF has stopped working, I tried
rfc2833compensate=yes and
it shows empty string
On 4/3/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
You can use the following to display what you receive from user (dtmf):
exten= 1,1,Read(test)
exten= 1,2,NoOp(DTMF Received: $test)
exten= 1,3,Hangup
On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote:
I upgraded to 1.4.1
That would be because $test is not a valid dialplan variable. You would
want ${test}
Nitin Gupta wrote:
it shows empty string
On 4/3/07, *Rizwan Hisham* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
You can use the following to display what you receive from user (dtmf):
I upgraded to 1.4.1 and my DTMF has stopped working, I tried
rfc2833compensate=yes and relaxdtmf=yes etc but none working.
Everything seems to work fine with 1.2.10
Is there any way I dump the dtmf data packets received by asterisk on
console?
Any idea or pointers to debug the issue will be
Subject
[asterisk-users] dtmf problem --
17/01/2007 15.38 second part
I realize I cannot use inband audio for phones (voicemail and internal ivr,
password for external trunks and other thing not working)
So I put everywhere rfc2833.
Doing this, anyway, make any EXTERNAL IVR NOT working.
I see a lot of posts about this, but no solution, becouse using inband
audio
[EMAIL PROTECTED] writes:
Bart,
We have has similar issues with BroadVoice in the past. From what I
understand they had problems with DTMF depending on which proxy you register
to. This is a bug that related to their session border controllers which
should have been resolved.
... snip
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] dtmf problem
hi everyone,
I do have 2 lines with broadvoice.
From 2 days on one line my dtmf tones are not passed to asterisk server.
It siply goes through the extensions routine acting link it did not
receive
any tone. Could
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] dtmf problem
hi everyone,
I do have 2 lines with broadvoice.
From 2 days on one line my dtmf tones are not passed to asterisk server.
It siply goes through the extensions routine acting link it did not
receive
any tone. Could
hi everyone,
I do have 2 lines with broadvoice.
From 2 days on one line my dtmf tones are not passed to asterisk server.
It siply goes through the extensions routine acting link it did not
receive any tone. Could it be problem with my config???
It looks like this:(it worked for last 1.5 year)
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Wegrzyn - asterisk
Sent: Tuesday, December 20, 2005 8:31 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] dtmf problem
hi everyone,
I do have 2 lines with broadvoice.
From 2 days on one line my dtmf tones are not passed
Hi, I got this message in the asterisk console while sending the dtmf from a phone.Dec 8 14:55:50 WARNING[29098]: codec_ilbc.c:163 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Please help me to solve this.Thanks Jibu
Yahoo! India
Hey all,
I have a problem that seems reminiscent of a previous CVS issue that
appeared that caused DTMF not to be interpreted on OUTBOUND (i.e,
Originated) calls from Asterisk. If the call was Originated from a SIP
client, or 'pass through' via the other attached PABX (ZAP/g2) it all
works
Hi everyone
Im using Asterisk as a transcoding gateway between
G711 and ilbc, and when I send out of band DTMF (rfc2833), the DTMF packet sent
has the same timestamp as the last RTP packet received by Asterisk. The problem
is that when I stop sending RTP, and only want to send DTMF
Guys.
Im testing the default asterisk demo setup that comes after installing, but
I have a problem with dtmf tones... I dial 1000 and listen to the welcome
voice but if I try to enter any keys like 2, 500 etc.. nothing happens...
Why are my dtfm tones not been recognized? what would be the
Hi,I'm running a Voicetronix openswitch12 card
under linux with asterisk.It's configured to have 8 loop start and 4
station ports. I've got afew ariavoice and grandstream phones running
off it without any problems.I've also got 4 analog phones running off
it too. They work fine, except fora
I'm really new to asterisk, but everything works fine beside DTMF recognition. As you can
see in output generated by asterisk, I constantly receive following line
"Private structure not found in send_digit." for every DTMF digit I enter on
h323 side. I have analog phone atached to
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
Very weird problem,
got a channelized T1 with SBC. There is a toll free number that points
to a local number on that T, when the local number is called, *
recognizes dtmf tones, when the toll free number is called, * does not
pick up dtmf..
any thoughts?
lizardbox
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