Displaying 20 results from an estimated 10000 matches similar to: "dtmf detection"
2003 Sep 08
8
Callgroup, Pickupgroup and SIP
I have just started to play with callgroups and pickupgroups..
I updates my * from CVS this morning (about 15 mins ago)..
I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf..
I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB..
Have I not configured
2005 Mar 01
5
Polycom Auto-Answer
I am having a problem with Polycom auto-answer. I have the auto-answer
working between PhoneA and PhoneB, but when I try to use the intercom
between more then one phone I start having problems. PhoneA dials *3
which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only
one will pick up, the rest will hang up and I get this error on
Asterisk: Got SIP response 500 "Internal Server
2009 Mar 02
1
SIP dialog matching problem? (1.4.23.1)
Hello all,
Not sure if this mail belongs to this users or dev list. Sorry about
that.
We have the following scenario:
PhoneA OpenSER Asterisk PhoneB PhoneC
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2014 Dec 16
1
Asterisk sends CANCEL to the wrong destination
Hi,
I got a weird behaviour in asterisk (original found in 1.8 but it is
still the same in 11.15.0). I have three phones communicating via
OpenSIPs with asterisk. Phone A dials 100 and asterisk calls
SIP/phone-b. Phone B accepts the call. The User on Phone B places the
call on hold, dials 200 and, while hearing the dial tone of ringing
Phone C, places the handset on hook. Phone B sends a REFER,
2005 Feb 12
2
Intermediary jitter buffering
Hello,
I understand that only the destination of a call should do jitter
buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no
transfers), PhoneA and PhoneB need to perform their own jitter buffering,
and Asterisk will just forward the frames, correct?
What happens if the peer does not support jitter buffering, but is
close by so there's no need for jitter buffering? My
2008 May 26
0
realtime problem with two Asterisk servers
Hi all,
I have a problem with using remote MySQL database server with two Asterisk (1.4.17) servers. PhoneA registers with Asterisk#1 using realtime into MySQL on remote server and everything is working fine and when I call Phone A from Phone B (also registered with Asterisk#1) call is established. Problem is when I call PhoneA
(which is registered with Asterisk#1) from PhoneC (which is
2006 Nov 05
1
asterisk DTMF detection
Hi,
Hi All,
I've just delved into the world of asterisk and I'm having a few dtmf issues.
Internally, amongst sip phones, dtmf is fine.
Externally, if you ring from a GSM mobile, DTMF is fine, however if
you ring from a standard phone, DTMF fails to register.
I am attempting to use a quad port HFC-4S Beronet Card. I've been
searching the web most of the last week and
2003 May 22
0
Call Parking Difficulties
I can't seem to retrieve a parked call.
Here is what I do:
Call from phoneA to phoneB
Answer phoneB
Press Flash/Callwait on phoneB
Press 700 to park the call
A voice says that the call is parked at 701
When I try to dial 701, I don't get connected to the parked
call
Below is the asterisk output when I tried to park
2006 May 11
1
mISDN trouble with a HFC Cologne card, Asterisk Asterisk 1.2.4 on Linux 2.6.16.11 - incoming DTMF detection
Hello everyone. I've got this really annoying HFC Cologne card (or
however I should call it - a single channel ISDN card based on the HFC
chipset).
It wrongfully detects lots and lots and lots of incoming DTMFs, to the
point the card is not usable.
Here's a sample out of CLI:
P[ 1] I IND :DTMF_TONE oad:206361 dad:520101
P[ 1] --> mode:TE cause:16 ocause:16 rad: cad:
P[ 1] -->
2003 Sep 05
2
Transfer (again!)
Hello,
I am building an asterisk PBX with some stuff to make a usable VOIP /
PSTN Gateway. I use the following devices:
SIP Phones from GrandStream for VOIP side
OpenLine4 from voicetronix for PSTN Side
I am building things step by step with some priorities.
I have now a working system able to place and receive calls from/to pstn.
Before attempting to bring other functions (like voice
2003 Aug 15
1
DTMF SIP
Hello list,
my case is as follows:
SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729.
When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the
keypad on the phone.
As suggested by you, I need to configure the SIP1 with out band dtmf mode ,
what is about the sip.conf, should I specify the SIP1 with demfmode=rfc2238
? do I also need to make same kind
2005 Feb 16
4
DTMF inband detection improvement
Hi all,
I have some probleem detecting DTMF send by a GSM phone,
I'm using SIP with ulaw.
do you know what are the options to improve the detection ?
I'm using asterisk 1.05,
is the CVS HEAD version had some improvement about DTMF detection?
Florian.
2007 Oct 24
0
Two DTMF tones on keypress with Handsfree cell
Hello, I am using Asterisk SVN, a cellular phone, and chan_mobile to
run a small home PBX with two analog telephones connected to a Linksys
ATA using SIP. It works great (except for some Bluetooth adapter bugs
that I am still trying to beat...seems the misaligned audio detection
still needs work), but I have encountered an interesting issue.
If I am using an automated system that accepts input
2009 Mar 19
0
DTMF tones mid conversation
Just to add....
P[ 1] Transmitting 128 samples 2 misdn
P[ 1] writing 128 bytes 2 asterisk
P[ 1] Sending :160 bytes 2 MISDN
P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0
P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0
P[ 1] Transmitting 128 samples 2 misdn
P[ 1] writing 128 bytes 2 asterisk
P[ 1] PH_CONTROL: channel:1 oad2:07nnnnnnnnn dad0:820055
P[ 1] --> DTMF
2007 Nov 21
1
quality after call transfer
Hi,
We are using attended call transfer to transfer the call. In the
direct call, the quality of the voice and dtmf are acceptable. After
transfer, the quality becomes worst. Voice can't be heard clearly and
dtmf wrong detection will occur sometime. I wonder call transfer will
affect he quality of the call. Anyone has same experience? Anything
to do in asterisk level can get a better
2004 Dec 06
1
DTMF via PSTN to * to IAX to * challanges.
Ok I have an * server finally setup and acepting calls from freshtel and
I am VERY impressed at how well the Freshtel.net service works but thats
another subject :)
I have it all setup so that I can Dial my DID number on freshtel and
that gets set to my * via IAX.
At the moment I have the demo configured so that I can test it all and
make sure it is all working.
The problem is that I
2007 Oct 04
2
Voicemail/dtmf not working?
Hi,
I am setting up an asterisk server for testing purposes and cannot get
voicemail to work at all.
My host OS is Linux From Scratch 6.3 and the asterisk software versions
I built are zaptel-1.4.5.1 and asterisk-1.4.12.
I am using the Ekiga softphone on my Ubuntu desktop machine. My asterisk
server and client phone are on different computers but are on the same
LAN, i.e. no NAT.
I have an
2008 May 05
3
simple realtime question
HI,
Does asterisk will ignore the setting in files if realtime is
applied, say asterisk will ignore all the setting in sip.conf if
realtime table sip_buddies is applied?
ango
2009 Feb 24
7
multiple asterisks in a server
Hi all,
Is it possible to install more than 1 asterisk in a single server?
If yes, what do I need to set and take care?
Rgds,
ango
2010 Jan 11
2
Sipgate > DTMF not detected
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the general section of sip.conf
This has made no difference. I've tried a range of settings (auto,
rfc2833,info) but no matter what, it plain refuses to pick up key
presses.
Locally, if I call from an