I think that's a very good idea. When I started to become active in * last
December the list was much less congested and Mark usually responded to
requests, comments and patches within a few hours. Now things are clearly
taking off - good for * and Digium but it's sort of losing the community
The sort answer is no. The ztdummy code is written specifically for
usb-uhci and usb-ohci operates in an entirely different way. However,
there is an alternative to ztdummy that uses the real-time clock. Take a
look at zaprtc from here http://www.junghanns.net/asterisk/page1.html
Iain
Here's a patch that changes the behaviour of # transfers in asterisk. A
single # is transferred to the remote phone/system. Two # in quick
succession will trigger a transfer. This is very useful for users who have
basic analogue phones connected to an ATA 186. For example, when calling a
.. poking head above parapet, venturing correction ..
RTP payload type 13 is comfort noise viz
http://www.iana.org/assignments/rtp-parameters
whereas payload type 19 is reserved. Maybe Cisco is right ;-)
I believe * has a partial implementation of comfort noise but that it's not
complete
I think the quality for music playback on my SIP stuff is pretty good. The
real sound problem is in the voicemail access. I very often get sound
dropouts when * is reporting the number of new or old messages.
Iain
--On Saturday, July 26, 2003 10:39 pm -0500 Mark Spencer
[EMAIL PROTECTED]
The basic call transfer functions, set with the T and t options to the dial
application and triggered by pressing a # work fine for me. Make sure that
you have set the DialPlan on the ATA 186 so as not to grab the # (ie look
for any # character pairs and change the second character or remove
Assuming it is a suitable Fritz card your best bet is to get the CAPI
library/driver from AVM and then check this out
http://www.junghanns.net/asterisk/ - chan_capi is reportedly the best
performing ISDN channel driver for asterisk, although I personally haven't
used it ;-)
Iain
--On
There was a thread on FWD failures yesterday and indeed it didn't work for
me at 9:00 in the morning but by 10:30 all was fine - I'd made no changes
to *. It looks as though there's some tinkering going on at the FWD end.
Iain
--On Thursday, July 24, 2003 12:32:00 -0400 Leif Madsen
[EMAIL
--On Saturday, July 19, 2003 16:30:04 +0100 Darren Poulson
[EMAIL PROTECTED] wrote:
The one thing that I think it could be is the connector to convert from
RJ45 to BT phone socket. I'm using a mod tap that I had lying around.
Not sure what the wiring is like inside it.
That's a pretty good
As far as I know the Sip support for the 7905 has not been generally
released so comments you've seen on this list refer to test versions of the
code.
You can set up a call between two phones on an ATA186 through asterisk.
Iain
--On Wednesday, July 16, 2003 9:28 pm +1000 Steven Honson
Not all of the * wait commands respond to dtmf whilst playing back.
Couldn't you use the Background application to play the music? That does
respond to dtmf whilst playback is in progress.
Iain
--On Friday, July 11, 2003 10:52 am + WipeOut .
[EMAIL PROTECTED] wrote:
Hi..
How do you
If you make an outgoing call to a conference bridge (or anything else that
needs DTMF '#') then you can't use the asterisk 'T' transfer option because
that is triggered by the '# also. Is there already a solution in # for
this eg use two keys to trigger a transfer rather than just the '#'?
.
[EMAIL PROTECTED] wrote:
Iain == Iain Stevenson [EMAIL PROTECTED] writes:
Iain I didn't used to have any trouble with FWD and * is registering
Iain with FWD OK. Has FWD changed or * changed in a way that might
Iain cause this error?
Jeff just announce an upgrade to fwd the other day.
One change
I got this today trying to place a call through FWD:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.11.11.1:5060;branch=z9hG4bK230f856c
From: Iain sip:[EMAIL PROTECTED];tag=as6eaa85fb
To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.3701
I didn't used to have any
RFC3389 is comfort noise. By default the ATA 186 will generate rfc3389
packets. You can turn this off through the ATA 186 web interface.
It looks as though you need to configure that ATA186 properly - several
people have posted guides on this.
Iain
--On Thursday, July 3, 2003 9:29 am
rxgain and txgain are used, for example with the X100P. As I understand
it, the echo problem with a SIP to PSTN implementation in * has two
components:
- echo resulting from the digital to analogue conversion at the X100P
- acoustic feedback within the handset used
The former is reduced by
I had a call today where there were several remote participants using a
speakerphone. They sounded quiet to me. Every time I spoke I got noise at
my end but the respondents never complained of any problems hearing me.
Iain
___
Asterisk-Users
Whilst in a call using the mec3 echo canceller today I had period of about
20 seconds of speech distortion. It's hard to describe but to me the call
sounded as though we were having the conversation in a bathroom with some
extra noise bursts and echo thrown in. I could carry on the call, with
... but it still only works on x86? I get a failure to find asm/i387.h at
line 69 of zaptel.c on my ppc box.
Iain
--On Sunday, June 29, 2003 11:55 am -0500 Mark Spencer
[EMAIL PROTECTED] wrote:
I've been working on a fixed point mec3 echo can. The old mec3 which
crashed a lot of peoples
... thanks - seems to go now. I'll test some more.
Iain
--On Sunday, June 29, 2003 3:30 pm -0500 Mark Spencer [EMAIL PROTECTED]
wrote:
Oops, a remenent of when it was still FP. Should be fixed now.
Mark
On Sun, 29 Jun 2003, Iain Stevenson wrote:
... but it still only works on x86? I get
I think the problem is more fundamental than this. The state machine in
the X100P assumes that nothing at all happens before a ring - so it will
simply ignore everything (eg UK caller ID tones) until it gets that first
ring to wake it up. Handling UK caller ID needs a re-write of the X100P
Roy Sigurd Karlsbakk posted a php utility to calculate call costs to this
list a while back. I hacked it for my own use and you can have that if
you'd like to improve it/make it general purpose.
Iain
--On Tuesday, June 24, 2003 11:18 am -0400 Marcus Adolfsson
[EMAIL PROTECTED] wrote:
--On Sunday, June 22, 2003 14:38:20 +0200 Hervé Thibaud
[EMAIL PROTECTED] wrote:
i have an error when i start asterisk in :
chan_modem.so (Generic Voice Modem Driver)
-- Parsing /etc/asterisk/modem.conf': Found
-- Loading modem driver chan_modem_i4l.so = (ISDN4Linux Emulates Modem
Driver)
--On Saturday, June 21, 2003 06:28:32 + WipeOut .
[EMAIL PROTECTED] wrote:
So far I have just got it to the point where I am able to make calls and
have not had the serious echo problems that everyone warns about when
using a passive card.. .
... you're using chan_capi - maybe that's the
A colleague called me through my * system via FWD using SJPhone and the
quality was distinctly poor - a lot of hum and delay. Looking at the debug
log the codec used was miscellaneously numbered 0, 4 and 8. I thought I'd
disabled 4 (g.723) but it appears not. My sip.conf has this:
general]
--On Thursday, June 19, 2003 17:24:21 +1000 Adam Goryachev
[EMAIL PROTECTED] wrote:
One problem I had with this problem is when I dial out through asterisk,
once I have dialled, the remote end doesn't detect my dtmf key presses.
ie, I can diall (eg a bank) but when they ask to press 3 for
There has been a lot of discussion about ISDN BRI on the list - a search
will turn up plenty of discussion!
You're right about there being a lot of ISDN cards available that are
certified for use in Europe. They fall into two categories - active and
passive. Passive cards are cheap and
Don't forget to set the database permissions. These need to agree with
whatever is in /etc/asterisk/cdr_mysql.conf.
Iain
--On Friday, March 28, 2003 3:14 pm + WipeOut .
[EMAIL PROTECTED] wrote:
Hi,
I see in /ect/asterisk there is a cdr_mysql.conf to configure the CDR
logging to a
--On Monday, March 24, 2003 1:00 pm +1100 Adam Goryachev
[EMAIL PROTECTED] wrote:
Does anyone know the location of the kernel patch to disable isdn dtmf
detection?
The patch below should do that.
Also the location of the asterisk patch for doing the dtmf detection?
Pauline Middelink posted
Do you know for sure whether the PBX issues a call termination pulse (ie
zero or reverse battery) on completion of a call?
Iain
--On Friday, March 21, 2003 8:56 pm +0100 Florian Overkamp
[EMAIL PROTECTED] wrote:
Hi guys,
So, now I've made a small demo box to do some IVR apps and hooked
Sounds like the i4l dtmf problem. Assuming you are using i4l, the kernel
dtmf detection routines are poor and quite frequently misinterpret speech
as dtmf tones. You need to patch asterisk to handle dtmf and i4l not to
detect dtmf (or silence). There are a few posts on this list about fixing
--On Thursday, February 27, 2003 1:54 pm +0100 Marian Danisek
[EMAIL PROTECTED] wrote:
this mean that i need 2 different patches ? I already found isdn_audio.c
and isdn_audio.h patch... this is for i4l. You meat that i need another
patch for asterisk ?
If you want asterisk to handle dtmf then
101 - 132 of 132 matches
Mail list logo
