Anthony,
Yes, it does work with echo canceller off. But without echo
canceller, when we bridge 2 calls together, we have echo on the line for the
voice and
the DTMF. For the DTMF, its crazy, one DTMF can loop
forever. I’m genna contact sangoma to do a follow up on that, but
do you have any T1
card provider that works without any problem that you guys work
with?
Thanks.
SL
From:
freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org]
On Behalf Of Anthony Minessale
Sent: Wednesday, June 25, 2008 10:30 AM
To: freeswitch-dev@lists.freeswitch.org
Subject: Re: [Freeswitch-dev] DTMF issue on playback command with
openzap
just for fun try it with the
echo canceler off.
I know for a fact that sangoma's echo canceler messes up the soft dtmf.
see the power of details? ;)
On Tue, Jun 24, 2008 at 6:43 PM, Steve Laroche <slaroche@ip5.com> wrote:
I'm driving the call with an IVR outside of the freeswitch using
event_socket. I'm using
digital lines. In fact, they are T1 with DMS100 protocol connected to
a sangoma card.
The driver I'm using for the sangoma card is wanpipe-3.2.5. This card
have echo
canceller activated on.
IVR originate a call through event_socket :
bgapi originate openzap/1/a/5147884250 &park()
The call is originated to a PSTN phone.
When IVR receive the events from event_socket confirming the call is
connected, IVR
send a playback command :
SendMsg uuid
call-command: execute
execute-app-name: playback
execute-app-arg: /usr/local/freeswitch/sounds/mission_vm.wav
Audio file is about 25sec long, so when the playback is executed on line, I
enter DTMF
on my PSTN phone.
I'm collecting the DTMF events with my system that drives the event_socket.
It's
basically my test.
Thanks.
SL
From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org]
On Behalf Of Anthony Minessale
Sent: Tuesday, June 24, 2008 6:39 PM
To: freeswitch-dev@lists.freeswitch.org
Subject: Re: [Freeswitch-dev] DTMF issue on playback command with
openzap
I understand your problem so you need not repeat anything you have already posted.
please describe in detail exactly what you are doing and how
when you say openzap specify if it's analog or digital etc
I still cannot tell how you are using openzap in this scheme
are you saying you are doing something like:
originate openzap/1/a/12121231234 1000
to call your pots phone and xfer to 1000
where an ivr is collecting dtmf?
On Mon, Jun 23, 2008 at 2:04 PM, Steve Laroche <slaroche@ip5.com> wrote:
Anthony, do you need more info ?
From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org]
On Behalf Of Anthony Minessale
Sent: Monday, June 23, 2008 2:33 PM
To: freeswitch-dev@lists.freeswitch.org
Subject: Re: [Freeswitch-dev] DTMF issue on playback command with
openzap
please describe the entire path of the call and
what protocol is involved on the example that does not work.
starting with the telephone that placed the call all the way to the telephone
or ivr that revived it.
On Mon, Jun 23, 2008 at 1:20 PM, Brian West <brian@freeswitch.org>
wrote:
You still didn't answer what was receiving the audio. If its OpenZAP
that means its inband. But can you elaborate more on the path? If
the audio has ANY gaps in it the DTMF will be double detected but I need to
know more about that before I can tell what is going on.
/b
On Jun 23, 2008, at 1:14 PM, Steve Laroche wrote:
Instead of
123456789012345678901234567890* I receive 1344455677890223444567890233344455567890*.
While for all the other test that I did I do receive
123456789012345678901234567890*. I did this test many time and always the
same result, I have an issue with DTMF when playback is running and the call is
originate through openzap.
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net
#freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
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googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400