On Jun 25, 2004, at 8:51 PM, [EMAIL PROTECTED]
wrote:
I highly recommend the UIP200. Although I haven't used all the phones
out there, I know it performs and sounds a lot better than the Snom
200 and Grandstream...comparable to Cisco 7960 in sound quality. We
use it for our Asterisk set up in
Hi,
I would like to make a call and then when I am connected to the
destination to transfer the call to my coleague in the office. When we
receive the call it is easy using #. But when I am the originator
the # doesn't work. Can you give me some suggestions?
Best Regards,
Hi,
I would like to have different type of behaviour of our IP PBX
depending on the time and the day:
Weekday
Nightly - 18:30 to 08:30
Daily - 08:30 to 18:30
Weekend, Holiday, etc.
For example Daily the IP PBX will rings to some phones, nightly
will work IVR system.
Hi all,
Anyone know where/how I can setup my own menu to work like the
voicemailmain menu.
e.g.
extension.conf
exten = 888,1,mymenusystem
exten = 888,2,Goto(s,6)
then somewhere mymenusystem plays message and give options to goto exten
1, 2, 3 etc
Thanks in advance.
Dee
On Sat, 2004-06-26 at 10:09 +0100, Dee Lowndes wrote:
Hi all,
Anyone know where/how I can setup my own menu to work like the
voicemailmain menu.
e.g.
extension.conf
exten = 888,1,mymenusystem
exten = 888,2,Goto(s,6)
then somewhere mymenusystem plays message and give options
Is opermode set via asterisk or do you need to do
modprobe wcfxs opermode=UK
chris
- Original Message -
From: Nicolas Gudino [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 26, 2004 12:43 AM
Subject: Re: [Asterisk-Users] X101P on a UK BT line txgain issue
Hi
hi everybody,
I'm running my asterisk with a HFC-S card in NT-mode with a modded NTBA
(NT1) (=simply crossed cable) and two ISDN-phone behind it. Now, when ever I
user both phones at the same time, the sound is very, very crappy, as if it
is played at a slower speed (like playing a 7'' single at
lets see if we help Jeremy (and ourselves) to narrow down the timeframe
when this problem startet.
I have the following release running with the recommended pw/openh323 libs.
Audio is working fine and I use faststart (must).
Asterisk CVS-04/13/04-22:41:25
Does anyone have a newer release running
At 16:47 25/06/2004 -0400, you wrote:
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59s-mh4 (2000/Oct/27).
Looks a bit old to me... I'll try to install a newer release.
You need version r this is the only one that works well with asterisk
Hi Folks,
Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk
to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with
port 5036 forwarded:
$IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT
When I call a PBX system and enter digits, Asterisk is
eating away some digits. For example when I call ATT
and when the system prompts me to enter my phone
number, Asterisk eats away some digits, so ATT does
not get the number that I entered. I am using the
extensions.conf as it came from the
Thanks all,
It's working now with version r.
Francois
At 16:47 25/06/2004 -0400, you wrote:
High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3.
Version 0.59s-mh4 (2000/Oct/27).
Looks a bit old to me... I'll try to install a newer release.
You need version r this is the
I need a provider of DIDs with multiple inbounds.
regards
joe baptista
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Just wondering if anyone can give me some pointers, I'm configuring Asterisk to
talk
to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with
port 5036 forwarded:
$IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT
--to-destination
I installed a TDM04b and a TDM40b with aggressive echo suppression
and it's working almost perfectly.
The problem is that all extensions are fax machines and people uses it for
both purposes, voice and fax. AFAIK, I cannot use aggressive suppression
for fax extensions, but when I turn it off
Hi,
I having a problem compiling the wrapper for oh323. I am running Debian,
kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the
openh323 version I have is 1.13.5. I execute the following commands first
before attempting to compile the wrapper:
pwlib_1.6.6:
Andres wrote:
I just tried this myself and it behaves as you have described it. No
need to use a username. When the call comes in on the remote Asterisk,
the iax.conf simply tries to match the password to any entry. The first
entry with a matching password gets used. I suggest
Erm, wont the TIMESTAMP value have changed during the monitor ?
Don't you need to set a CALLFILENAME var, just once and re-use it.
T.
Carlos Medina wrote:
Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix
on my linux shell it works so i can mixed two calls into one
Rich Adamson wrote:
Reading way between the lines and taking an educated guess, I'd suggest
the reasoning behind Mark's architectual thoughts are likely to relate
to providing peer-to-peer call completion capabilities, as opposed to
forcing all * systems to pass through some
Hi!
- method a) SetGroup() and GetGroupcount() in extensions.conf
- method b) incominglimit= and outgoinglimit= in sip.conf
But could you actually *prevent* the transfer? Or would you have to wait()
and dial() again?
I have the feeling you are looking at this the wrong way - probably
Hi!
I would like to have different type of behaviour of our IP PBX
depending on the time and the day:
Do some reading about context and how they can be included based upon the
time of day:
; This includes the context daytime - timing list for includes is
;
; time range|days of week|days
Hi all
I have just got a P2000w and experience several problems. Hopefully there is
someone out there that has got it working. I saw it on Cebit and the person
demonstrating it there told me that it was connected to an Asterisk server
on the stand -so it should work.
Problem 1: it does not
Hi all,
I was excited to see the announcement on the list regarding the fix for
the echo problems on Digium FXO cards!
I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite
softphones. A few months back,I had gone thru the recommendation on the
list to remove echo from the SIP
If this is their wireless model similar/identical to the wisip. I would suggest
using g729 to get rid of the choppy sounds. Not sure what the issue is with
registration off the top of my head. I had some problems with a test phone last
week with wep turned on. If possible you might try testing
I was excited to see the announcement on the list regarding the fix for
the echo problems on Digium FXO cards!
I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite
softphones. A few months back,I had gone thru the recommendation on the
list to remove echo from the SIP
the problem not only occures when I use both phones - when I'm using phone 1
and annother calls knocks on for example - the sound is also not ok.
any hints? I'm using a VIA C3 600 MHz with a dual-riser from VIA (make 2
PCI-slots out of one). maybe this is the problem ?!
- Original Message
Chris Stenton wrote:
Is opermode set via asterisk or do you need to do
modprobe wcfxs opermode=UK
You need to do modprobe wcfxs opermode=UK
This will only work if you have the TDM400 FXO modules. The X101P is a
600 ohm US/JATE card only.
Richard
___
T. Chan wrote:
Jeremy, any way to fix that? Thanks again.
I've spent many many days trying to duplicate any of these problems and
absolutely cannot.
I have tried everything from my mini-itx to my celeron based laptop to
my dual xeon dell 1750s and every single one of them work 100%
On Sat, 2004-06-26 at 15:27, Jeremy McNamara wrote:
I even manage a few different production systems with 5300s and they are
running absolutely perfectly on asterisk cvs -head with chan_h323.
Can you post the config from your 5300s?
--
Eric Wieling * BTEL Consulting * 504-899-1387
What you describe as slow sound is most likely not a problem with the
driver.. atleast.. very unlikely. Neither with hardware...
You need proper 100ohm termination on your bus, please check that.
Slow sound is a very very wierd problem... you can experience clicks,
noise and/or frame (voice)
Eric Wieling wrote:
Can you post the config from your 5300s?
They are customer owned gateways, but I can try.
Jeremy McNamara
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Hi, Jeremy, thanks for your help and dedication in resolving the problem.
There must be something that could have caused the problem. Why don't I
provide detailed information on what hardware I use and how I installed the
Asterisk and I would suggest that other colleagues who had or are having
On Sat, 2004-06-26 at 16:32, Jeremy McNamara wrote:
Eric Wieling wrote:
Can you post the config from your 5300s?
They are customer owned gateways, but I can try.
Heck, post the Cisco configs, the chan_h323 config, and sample Dial
lines. The more info the better. 8-)
--
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