Displaying 20 results from an estimated 20000 matches similar to: "ZAP extension, DTMF?"
2006 Jan 12
2
SIP phones unbeatable echo
Hey all again, I'm wrestling with echo problems on our sip extensions. I've
set these items in zapata.conf but tweaking these values doesn't seem to
make much difference
echocancel=yes
echocancelwhenbridged=yes
echotraining=2500
rxgain=8.0
txgain=1.0
are there other settings that can help me tame this beast? Been searching
but not turning up anything that'll work here.
Thanks
2006 Feb 15
6
asterisk silence suppression?
Hi all, I'm getting some noise gate like effects on our sip lines & I think I need to disable silence supression, I'm searching docs & not finding where this can be set, does * have a setting to turn this off? basically what's happening is when we stop talking, the other end hears total silence, but when we talk, they can hear the background noise in the office, this sounds odd
2006 Feb 06
4
two tellabs 2572 echo board in a 253c mounting assembly?
Anyone gotten two of the 2572 echo canceller cards to work in a 253c mounting assembly? I can get one to work, but when I install two, one always fails. I've tried all my cards solo in the enclosure, on each side, and they all work properly when only 1 is installed, however, when I install two, one of them will come up, but the other always fails. Anyone know what might be causing this?
2006 Feb 07
2
Re: two tellabs 2572 echo board in a 253c mounting
30 says it's view only in the docs & I can't seem to change it, any other options?
> Option 30 allows to set Module Shelf Address/ID.
2006 Jan 13
2
zapata.conf for non pri T1?
Hi again, I'm trying to setup our non pri T1 (they call it a Long Distance
T1), our current pbx has the signaling set to E&M, I can set em in
zapata.conf, but I'm trying to track down the proper entries for the
zaptel.conf file. The digium docs only show a PRI example. Our current
system has these settings:
Signalling: E&M
Framing mode: ESF
Line Coding: B8SZ
here's my
2006 Jan 12
2
Asterisk crossed lines?
Hey all, been noticing some oddness on a new AAH install... occasionally an
incoming zap line with automatically connect with an outgoing extension,
even though the incoming line hasn't specified what extension it's aiming
for (i.e. haven't tapped in the ext # yet)... so someone's trying to call
out from inside the office & are automatically connected with an incoming
line.
2006 Mar 10
2
Disable flash transfers?
Is there an easy way to disable flash transfers? I'd prefer the users hit # to transfer, since some users are hanging up a call, then dialing another one without giving the handset enough time to actually hangup the call, so it appears that they are transfering the 'ended' call to the new number that they are calling.. I'd like to keep flash functionality for call waiting, but
2006 May 31
1
Upgrade ONLY asterisk from an AAH install
Hey all, is it safe to run the asterisk-update.sh script that comes with AAH
to upgrade only the asterisk binaries? Doug has chimed in a few times saying
'upgrade' when I post problems, but Aah makes this really painful. I'm using
AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in
my installation. Can I safely upgrade just asterisk and not any of
2006 May 25
4
No rings before auto attendant
Hi all, been searching & not finding an answer to this, although I'm
guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0),
which had been using POTS lines via a channel bank.. Now when I call the new
T1 circuit, there are no rings, the Autoattendant just picks up right away..
Any clue on how to make it ring twice before getting picked up? I tried
immedate=no and
2006 Jun 14
2
Calls keep ringing after being picked up
Hi all, using * 1.2.9.1 and this week all of the sudden calls keep ringing
even after they've been picked up... Here's one users summary:
When I pick up the phone, I hear a dial tone and I am able to dial out.
But for some odd reason, the receiving line picks up while the outgoing line
is still ringing.
And the receiving line can hear everything while the phone is still ringing.
I tested
2006 Jun 15
1
Dropped calls continued
Hi All... Well, I'm still experiencing LOTS of dropped calls since
installing the new (non pri) T1 here... I keep noticing a few things in the
logs when this happens, namely the "Wink/Flash" statements and the "Didn't
get a frame" messages...
Anyone got any ideas on if this is a telco issue, a wiring issue, or an
asterisk issue? Been trying to track this down via all 3
2006 Oct 08
3
Tellabs and a PRI
Another question,
Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our
analog lines to a PRI, I thought it would be simple stuff moving the EC
to the PRI. Changed signaling, made sure that channel 24 wasn't being
ECd and everything came up. But, I was getting complaints of random echo
on the PRI. Local echo. Also, we weren?t able to do any kind of modem
dial-outs (Adit
2006 Jan 20
1
Hardwiring a Tellabs echo canceller - help req
Hi All, Greg has been a huge help getting me going with this tellabs echo
can, but I'm still having some problems getting it to work... I suspect I
wired it up incorrectly, so I thought I'd see if anyone can point me in the
right direction. Digium tech support pointed me to this doc for a standard
T1 cable:
http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm
which looks like it
2008 Sep 17
1
DTMF detection problem on DISA
Hi everybody,
I am having DTMF detection problem on DISA with my callback system. For many
users, it keeps playing the dialtone even after they have input their
number. I have trunk setup to both g729 and ulaw. What could be the reason
for this problem. Some users have to dial a few times before the system can
recognize their dialed number.
--
Zeeshan A Zakaria
-------------- next part
2006 Nov 19
4
reduce dialtone volume on zap channel.
Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work.
I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized.
Any other thoughts on how to solve this are also
2005 Mar 08
4
Nortel ATA not passing dtmf tones to fxo
I am trying to integrate a Nortel Norstar system with an Asterisk service
using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel.
The problem is when someone dials from the Nortel PBX to the Asterisk server.
Asterisk answers the call and provides a dialtone (with DISA) but appartently
the DTMF tones are not passed to asterisk and the call cannot proceed.
This only
2004 Sep 07
1
Monitored outbound dialing via Zap interface?
I'm using a T100p to interface to a channel bank and from there to analog
PSTN lines. Because of my particular setup I have to do post-connect inband
DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming
0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an
'outside transfer' voice prompt before commencing dialing my users are
getting
2007 Jul 10
3
ZAP TDM and DTMF issue
Hi,
I'm curious if there is any other option beside relaxdtmf in zapata , or any where else to tune dtmf detection on TDM400 fxo boards.
in one of our sites provider is giving us 4 analog lines out of Adtran router and Asterisk often recognize DTMF wrong.
Obviously playing with relaxdtmf was not helpfull.
What do we know anout 1.2 and 1.4 DTMF handling diffrences?
At this time i'm using
2006 Jun 05
2
DTMF and DISA
Hi Folks,
I'm trying to test out Asterisk overall.
I'm having some problems with DTMF. Currently I'm playing with DISA,
but I'm worried this will happen when I get to implementing AAs etc.
I have a free SIP trunk from IPKall that I'm trying to make work.
I'm able to receive calls, and I've now setup and extension with DISA
and a password.
I connect ok from the
2005 Aug 02
1
Strange DTMF issue with callback
Hi
I'm trying to implement a Callback mechanism whereby I generate a Call
file and connect an arbitrary extension with my cellphone (via a SIP
Channel).
If I create a .Call file that connects the channel
"SIP/12345678@Provider.net" with a local extension/context I get some
weird issues with DTMF tones.
I've set dtmf=2833 and the codec in use is G711a.
For example - I create