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
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.
rather than inband. Why
one works and the other doesn't I don't yet know.
Cheers Duncan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug
Sent: Thursday, 17 May 2007 2:40 p.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF not working
At 21:40 5/16/2007, Doug wrote:
Has anyone seen anything like this:
I dial *98. Asterisk says Password? I punch in
the password, and the system doesn't recognize the
tones.
However, if I dial my own number and ignore the
incoming call, it goes to voicemail, and then
I can get into voicemail.
You should really include the dialplan snippet that controls *98, so
that people can formulate better responses, so that being said, I have
to ask, did you do an Answer() step?
Doug wrote:
Has anyone seen anything like this:
I dial *98. Asterisk says Password? I punch in
the password, and
Has anyone seen anything like this:
I dial *98. Asterisk says Password? I punch in
the password, and the system doesn't recognize the
tones.
However, if I dial my own number and ignore the
incoming call, it goes to voicemail, and then
I can get into voicemail.
I have a sneaking suspicion
Do you mean, only one originating caller has problems with DTMF, but
other callers do not?
What kind of phone/service are they calling from?
We have had problems specifically with Skype/Skypeout customers not
working correctly with DTMF. I have heard of others who have problems
with certain cell
No, all incoming calls DTMF works as far as I know (sometimes they just
don't tell me). However, there are two numbers that I know of on the
outgoing leg that don't work. What's strange, is that they work fine on
the system we're passing through (we're hooked up to a meridian), but
won't
Anyone have any ideas why DTMF would not work on only one number? Looking
through the logs, anytime a button is pressed, this is what shows up:
2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1
2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on
I thought I sent this earlier this week, but I didn't see it. If I
missed it, I apologize for the resend.
We are running asterisk 1.2.2 with a TDM04B connected to PSTN lines. On
incoming calls from cellphones located overseas, DTMF is not recognized
- we have many single-digit choices in
-Users] DTMF not working
Innocent Evil wrote:
I am having same problem .. DTMF is not working from a SIP phone while
sending to Asterisk cmd VoiceMailMain.
Have you set DTMF to out of band RFC2833?
In band won't work. At least in my version of HEAD
John Novack
Would you please
I had a similar problem that seems to be caused by the DTMF tone lengths
being to short. Try this:
Asterisk generates DTMF tones in do_senddigit() in the file channel.c.
The tones are defined in a const char array called dtmf_tones[]. Each
DTMF tone is a string that looks something like:
I am having same problem .. DTMF is not working from a SIP phone while
sending to Asterisk cmd VoiceMailMain.
Would you please explain this line
!941+1336/100,!0/100, /* 0 */
what value is what and how it affect on DTMF tone generation.
Thanks,
I had a similar problem that seems to be
Hi Mr. Evil,
I'm not sure if the problem that I am describing relates to the problem
that you are having. It seems that when you press a key on a SIP phone
that is set for inband DTMF, asterisk absorbs the tones until you
release the key. This way if you are using DTMF to do things like
Innocent Evil wrote:
I am having same problem .. DTMF is not working from a SIP phone while
sending to Asterisk cmd VoiceMailMain.
Have you set DTMF to out of band RFC2833?
In band won't work. At least in my version of HEAD
John Novack
Would you please explain this line
Subject: Re: [Asterisk-Users] DTMF not working
Hi Mr. Evil,
I'm not sure if the problem that I am describing relates to the problem
that you are having. It seems that when you press a key on a SIP phone
that is set for inband DTMF, asterisk absorbs the tones until you
release the key. This way
Hi all,
I just upgraded from Asterisk 1.0RC1 to Asterisk 1.0.7 and our dtmf no longer
works with external phone systems. I have a Wildcard TDM400P with 4 FXO's?
(it connects to analog lines). No changes were made to the config files.
Here's my config:
/etc/zaptel.conf
fxsks=1-4
loadzone = us
Im trying to configure voicemail, but asterisk doesnt respond to dtmf codes.
I uses kphone with g711u codec (I've tryed the others one) and in sip.conf I
configure dtmfmode=rfc2833 (I've tryied inband and info).
Asterisk seems not to see the tones. Could somebody help me? Thanks
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is all on an internal switched 100mb lan.
Has
On Fri, 23 Jul 2004, Brent Franks wrote:
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 23 July 2004 12:21 pm, Brent Franks wrote:
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's
Brent Franks wrote:
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is all on an internal switched 100mb
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