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).
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
Deployed a new server different mobo and problem went away. Same version of
asterisk, same sangoma card.
Sent from my android device.--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
I am still having major issues with dtmf recognition. My setup is Polycom end
points. Tried this with different models, firmware and cfgs. Outbound calls are
not going out reliably. Phones are set to rfc2833. I have had sangoma and
elastix support look at it.. No better. Running asterisk 1.8.4.
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
Latest firmware is on the card
Sent from my android device.
-Original Message-
From: Jim Dickenson dicken...@cfmc.com
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
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
Hi,
We have few systems with asterisk 1.4.22.1 and we use sip trunking for them
not PRI's, one of our system is giving a problem of dtmf (rfc2833), like
when we dial the number that have IVR and enter the extension or access
code, it some time takes it and some times does'nt recognize the digits
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 sandesh...@gmail.com wrote:
Hi,
We have few systems with asterisk 1.4.22.1 and we
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
--
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
I have a block of DID's that I ported to Vitelity about 7 days ago. The
problem is if a POTS caller dials into the system, his dtmf is not heard
at READ() or Background() while a prompt is played. After the prompt is
finished, then dtmf is heard. I've been working with their support, but
it
- Original Message -
From: Barton Fisher bpvoip...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 07, 2009 18:25
Subject: [asterisk-users] DTMF Issues
I have a block of DID's that I ported to Vitelity
Ok, I'm still rather new to Asterisk, and I'm sure there is a simple fix
here, but I can't see it.
Client with a small system, AsteriskNow 1.4, 10 Polycom IP330 phones. Has
been up and running flawlessly for about a year.
This morning I logged in to make a couple of extension changes, and
I am having a big problem with DTMF. I have a customer using an
Asterisk 1.4.20.1 system with ZTDUMMY as the timing source. The problem
is that when they dial into a conference bridge or IVR where they have
to enter a code they always get an error. Either some numbers are
duplicated or
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
Hello, all!
Trying to figure out an issue with DTMF recognition with 1.4.19. This
is somewhat similar to the issue described here:
http://bugs.digium.com/view.php?id=11740, but it might be a different
issue altogether.
I have 1.4.19 running with TE212P on a US PRI.
I'm sending digits
Date: Fri, 2 May 2008 15:14:54 -0500
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issues in 1.4.19 with missing digits
Hello, all!
Trying to figure out an issue with DTMF recognition with 1.4.19. This
is somewhat similar to the issue
. April 2008 11:34
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] DTMF Issues
Hi all
I am getting the feeling you are going to be hearing alot more from me in
the near future.
I have yet another issue here. This time its with DTMF (again).
Ok the setup
I am missing DTMF digits on a PRI with 1.4.11
I added dtmf logging in logger.conf. I can see that if I
enter 205 I dont see the 2 but all I see is 05.
I have added the dsp.c patch that was recently added to bugs but
that doesnt seem to help my situation.
What can I do to provide more
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
I have a pri connection to the phone company.
Sending DTMF out over the pri I hear on my phone when I call it.
However, a second box uses a SIP connection to talk to the first box.
When the second box is trying to do the function sendDTMF(1) over the
SIP connection and then out the PRI I do not
Hi all,
I am trying to indentify a problem: I have 2 machines, one with Asterisk
1.0.11, the second with Asterisk 1.2.17. Both running with the same zaptel
(1.2.16). Asterisk 1.0.11 running on Sarge with AMP's dialplan and the 1.2.17
running on Etch with FreePBX's dial plan.
Now on both
Is anyone else experiencing problems with Asterisk 1.2, the ivr does not work. I have tried it on Linksys RT31P2 and Grandstream Handytone 496. After a call goes through you're not able to enter any of the prompts on a IVR. and cannot enter pin numbers when using a calling card or anything that
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc. Has anyone else been having these issues? I'm
only accepting ulaw
@lists.digium.com
Subject: [Asterisk-Users] DTMF issues with SIPPhone?
Does anyone else have DTMF issues with SIPPhone? When calling into my
DID, and entering, say, 1002. Sometimes it will recognize it properly
(rarely), other times it will receive something different. Such as,
1102 or 1000, etc
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=inband,
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?
Louie,
On 8/8
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
: [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
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-
: [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 PROTECTED] wrote:
RFC2833 is sent out of band
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
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
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 have one.
morn all,
I ran into a strange issue last night and have not been able to find
resolution either in documentation (wiki) or experamentation.
Using a handyton to feed dial tone to a pbx I am able to connect both ways
with no problem. If I make a call from the pbx through asterisk I can
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
: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's set to avt
or
auto. The internal dtmf used for voicemail
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 unsuccessfully with Cisco right now on this, but they
cant
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 unsuccessfully
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: [Asterisk-Users
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Greg Hill
Sent: Tuesday, August 10, 2004 1:32 PM
To: Asterisk
Subject: Re: [Asterisk-Users] DTMF issues
On Tue, 10 Aug 2004, AJ Grinnell wrote:
I am now at a total loss. Using Sipura spa-2000s connected
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 anyone had a similar issue with Asterisk
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 of SJPHONE?
SJPhone Evaluation Version
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.
Is there a way to not
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
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