What version of Asterisk are you running? What setup ?
I've just hit issues with 1.4.19.1 (see previous post)... Same
symptoms-ish.
Adrian
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Sent: 30 April 2008 10:34
To: Asterisk Users M
Sorry
We are running Asterisk 1.4.18, libpri 1.4.3 and Zaptel 1.4.7.1 on an
Intel core 2 duo 1.6Ghz, 2GB ram under
Ubuntu server.
Regards
Ian
Adrian Marsh said the following on 30-Apr-08 01:24 PM:
What version of Asterisk are you running? What setup ?
I've just hit issues with 1.4.19.1 (s
Not shure if it helps here, but we had nearly the same dtmf problem in an
asterisk 1.4 install with SNOM 320/360 phones.
After hours of fiddling arround, we used relaxdtmf=yes in sip.conf and the
problem went away.
Guido
_
Von: Ian [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 30. A
Hey Carlos,
>What is the best method to debug DTMF issues? Do I have to sniff the
> SIP packets?
The best method to debug DTMF issues depend on how you receive those
DTMF digits. Assuming you can use SIP INFO for the DTMF, that means
the DTMF digits are not really DTMF :-), that is, is n
- Original Message -
From: "Barton Fisher"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, October 07, 2009 18:25
Subject: [asterisk-users] DTMF Issues
>I have a block of DID's that I ported to Vitelity about 7 days ago. The
> problem is if a POTS call
On Tue, 10 Aug 2004, AJ Grinnell wrote:
> I am now at a total loss. Using Sipura spa-2000s connected to *, I get
> DTMF working just fine for internal extensions, voicemail, etc. If
> making an outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I
> get no dial tone. I am working unsucces
TMF sounds very distorted
on the other end of the call. I am trying to use rfc2833. If you have any
ideas, please let me know. Thank you.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Greg Hill
Sent: Tuesday, August 10, 2004 1:32 PM
To: Asterisk
Subject: Re:
isn't there, then it's not a real
option and would probably be ignored by the config-parsing algorithm.
Greg
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Greg Hill
> Sent: Tuesday, August 10, 2004 1:32 PM
> To: Asterisk
ED]>
Sent: Tuesday, August 10, 2004 1:23 PM
Subject: RE: [Asterisk-Users] DTMF issues
> On Tue, 10 Aug 2004, AJ Grinnell wrote:
>
> > I hadnt heard of that setting until today either, but it still doesnt
work.
> > I am using dtmfmode-rfc2833 in sip.conf, and I have my spa'
Hi Mike,
> I have tried the Handytone set for DTMF info and rfc-2833 (as well as exp
>with inband) as well as the sip.conf entry for it.
>From my experience DTMF with any Grandstram device works well only
with SIP INFO method ... give it a try (remember to set it up on asterisk as
well).
Best r
I had a very strange problem with a Sangoma card that I had both Sangoma (about
3 hours) and Digium (about 2 hours) look at. When I got a different Sangoma
tech to look at the problem it went away. I told the tech he did something and
he said I alway verify the firmware on the card is updated an
Latest firmware is on the card
Sent from my android device.
-Original Message-
From: Jim Dickenson
To: asterisk-users@lists.digium.com
Sent: Fri, 08 Jul 2011 5:59 PM
Subject: Re: [asterisk-users] DTMF issues still
I had a very strange problem with a Sangoma card that I had both
Has anyone had a similar issue with Asterisk Voicemail being unable to
detect the digits sent from an SJ Phone connection. I have included
dtmfmode=inband and it works fine when calling other phones though not with
Voicemail. Voicemail doesn't regonise the password.
I am using SJPhone, and work
What ver of SJPHONE?
Thanks for the voicemail stuff :-)
- Original Message -
From: "Girish Gopinath" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, February 28, 2004 7:48 PM
Subject: RE: [Asterisk-Users] DTMF Issues with SJPHONE
>
> >Has
What ver of SJPHONE?
SJPhone Evaluation Version, release Jul 31, 2003, Build: 1.10.187c
Girish
_
All the news that matters. All the gossip from home.
http://www.msn.co.in/NRI/ Specially for NRIs!
__
Same as mine. Strange!
I'll keep trying. Cheers.
- Original Message -
From: "Girish Gopinath" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, February 28, 2004 9:53 PM
Subject: Re: [Asterisk-Users] DTMF Issues with SJPHONE
>
> >What ver
We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband and
make sure you're using a ulaw connection. If you use a lossy codec, it will
scramble the DTMF tones.
Your config would change like so,
[sipphone]
type=peer
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
Louie,
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> We encountered the same issue. change dtmfmode=rfc2833 to dtmfmode=inband
> and make sure you're using a ulaw connection. If you use a lossy codec, it
> will scramble the DTMF tones.
Are you using SIPPhone? When I use dtmfmode=inban
in rfc2833 mode that you're
having.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 12:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone?
So the way I understand this is with rfc2833, DTMF is sent out of
band. So does this mean that SIPPhone is interpreting the tones
incorrectly? Asterisk shouldn't be doing any actual tone detection
with this method, right?
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> Yes we are. I just d
t: Re: [Asterisk-Users] DTMF issues with SIPPhone?
So the way I understand this is with rfc2833, DTMF is sent out of
band. So does this mean that SIPPhone is interpreting the tones
incorrectly? Asterisk shouldn't be doing any actual tone detection
with this method, right?
On 8/8/05, Tarpo, Lou
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> RFC2833 is sent out of band. What's the output on your asterisk console?
I don't see any output during this time on my asterisk console.
Unless there's additional logging I'd need to enable?
Thanks for the help!
-JD-
_
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jason
DiCioccio
Sent: Monday, August 08, 2005 5:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] DTMF issues with SIPPhone?
On 8/8/05, Tarpo, Louie <[EMAIL PROTE
I guess the problem is with SIPPhone then. I opened a ticket with
them. I'll post their response when I have one.
Thanks!
-JD-
On 8/8/05, Tarpo, Louie <[EMAIL PROTECTED]> wrote:
> I find a verbosity of 10 (asterisk -rvv) gets me adequate logging for
> my purposes. I've been really pou
On 8/8/05, Jason DiCioccio <[EMAIL PROTECTED]> wrote:
> I guess the problem is with SIPPhone then. I opened a ticket with
> them. I'll post their response when I have one.
>
I wouldn't bet money on that yet...
I've seen identical DTMF problems (doubled and mangled) digits and
I've never used SIP
I'll give it a shot.. Do you know if they have any plans to merge this in?
On 8/8/05, Gary Reuter <[EMAIL PROTECTED]> wrote:
> On 8/8/05, Jason DiCioccio <[EMAIL PROTECTED]> wrote:
> > I guess the problem is with SIPPhone then. I opened a ticket with
> > them. I'll post their response when I hav
Hi Jerry,
Please post your Zapata.conf configuration.
Joseph
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Wednesday, September 19, 2007 8:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dtmf issues on PRI and 1
Sandesh,
Review the bug logs in particular ID: 0017571..there's a recent patch
that may apply to your issue.
http://issues.asterisk.org
Thanks,
Al
On Thu, Jul 8, 2010 at 4:18 PM, das sandesh wrote:
> Hi,
>
> We have few systems with asterisk 1.4.22.1 and we use sip trunking
>From what you explained, it seems obvious that there exists some non-SIP
device somewhere in your communication path, and since voice is picked up as
DTMF, some device is also set to listen for inband DTMF.
What is the origination source of incoming calls to your system?
Zeeshan A Zakaria
--
ww
Thanks Zeeshan.that server is located at the headquaters and phones are
at different locations, even with default rfc2833 mode, other party IVR
prompts was not able to detect the tones, also 'Info' works good but not
with internal options like voicemail, etc. And inband is not being used as
we
Hi,
I got the captured packet traces and we could see that it was coming from
our asterisk server. Is there any other things that I need to look into,
also we have updated from Zaptel to Dahdi 2.0.2.2, but no luck, still the
random redial dtmf tones are coming in between calls...Can anyone sha
I had a similar problem when using the TDM03B card with 3 fxo module. In my
cas,e the issue stemmed from a noisy analog line from AT&T, so I had to tune my
TDM card by using fxotune utility. I hope this helps. Check this link out:
http://www.voip-info.org/wiki/view/Asterisk+fxotune
-John
>
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
33 matches
Mail list logo
