Displaying 20 results from an estimated 800 matches similar to: "No subject"
2007 Jul 12
0
No subject
an external program, which at this stage, is not customizable ...
I don't know if alternatives (LiMO, Android, ...) would be more open to this
customization but for Symbian, not only Nokia SIP client wouldn't let you
autoanswer to SIP calls, but any other SIP client complying to Symbian
design wouldn't support autoanswer.
PS: Please, note that I'm far from being an expert in GSM
2011 Jan 06
0
No subject
If you don't use 'CERTVERIFY 1', then this will at least make sure that
nobody can sniff your sessions without a large effort (...)
> So, do I misunderstand CERTVERIFY directive ? Or is there a bug ?
>> Can you reproduce such behaviour ?
>>
>
> I'm not sure what is going on. Can you try running 'upsmon' with debugging
> enabled? The following are
2007 Jul 12
0
No subject
That's the main reason I opened this thread as it surprised me a bit ...
>
>
> Any 2-wire analog leg will be a source of echo. Many, many, many calls
> do not have a 2-wire leg.
Even in handset audio circuit ?
I was thinking that any handset is a potential echo source due to this audio
circuit ...
Do you agree ?
> Think cell/mobile or endpoints with PRI or T-1.
>
>
2007 Jul 12
0
No subject
such file or directory" on pure-IP platform in which I installed
asterisk-libpri-dahdi trilogy.
Maybe, it's me while following README instructions, maybe README
instructions could be improved or maybe it's wrongly labeled messages ?
That's why I told myself : I'm waiting too much from doc ? is a pure-IP
platform too specific ? what is the official policy ?
README starts with
2009 Jan 16
0
No subject
...
Thanks, anyway for telling as at least, it reflects your needs.
>
>
> You want NT PtMP and i second that,
>
not being limited on the asterisk
> side is a must in the
> telephony ecosystem, since the legacy PABX aren't alwsys easy to
> reconfigure.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by
2011 Apr 12
0
No subject
supported, beside Idle, On call and Ringing ?
Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE |
INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD)
> Might be worth seeing if other phones do the same.
>
> S
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by
2009 Jan 16
0
No subject
connecting legacy PBX to Asterisk (for the very same reason, those PBX use
TE-PTMP).
If others could join this thread and say if they agree or not with NT-PTMP
being the 2nd most needed mode, would be great.
Please, do not hesitate to comment.
>
>
> Right now, I would not preclude the possibility that NT-PTMP support
> might be added, but I could not give you a concrete time at which
2009 Jul 20
0
No subject
mailboxes).
Are you certain that removing either 612 or 610 mailbox would keep Asterisk
from complaining ?
>
> However, the MWI does not indicate voice mails for 610 and I keep seeing
> this error message:
>
> ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox
> 610 in context a10
>
> However, mailbox 610 is clearly defined in voicemail.conf:
>
2009 Jan 16
0
No subject
could be "hot". Is there any chance this would cause the card to fail after
a while? It appears this site just had 4 port Digium card fail today.
> Also, I am trying to cross connect with another Asterisk system with
> > the normal LBO setting (i.e. span=1,1,0,esf,b8zs) but as of yet the
> > systems aren't seeing each other at all. Could the side with the high
>
2011 May 13
0
[LLVMdev] [ptx] Propose a register class naming convention change
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=UTF-8" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Justin Holewinski wrote:
<blockquote
cite="mid:BANLkTi=Y9EFmWRu-9dQxydq8zTyF7tEbJw@mail.gmail.com"
2010 Oct 06
1
Microsoft ISA Server
A tinc from behind ISA Server can't connect to another tinc, both windows. The error displayed is:
Timeout from name (ipaddress port number) during authentication
I can connect to the ipaddress on the port with telnet - no problem.
The wierd thing is: *all* programs that connect to internet display the
proxy ip address in TCPView as the remote end point. Telnet included.
tinc on the other
2007 Jul 12
0
No subject
1. Is it normal to see :
# lsmod
Module Size Used by
dahdi_dummy 3236 0
Shouldn't it be used by asterisk or is this 0 value meaning something
specific ?
2. How can you check dahdi is running ?
Here, "ps aux | grep dahdi " replies "grep dahdi".
Cheers
------=_Part_2692_19661943.1228286635399
Content-Type: text/html; charset=ISO-8859-1
2007 Jul 12
0
No subject
I'm not aware of any zaptel driver for such HFC USB modem (some Xorcom's
products use USB, so ...) so I'm inclined to think it's not possible but
it's better to ask ...
Cheers
------=_Part_13548_28463665.1207585749504
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable
Content-Disposition: inline
Hi
2009 Jan 16
0
No subject
"Why Siphon doesn't allow to receive a call when it doesn't run
Apple doesn't accept (for the moment) an application runs in the background=
.
So, when Siphon doesn't run, the SIP server of your provider doesn't know
your iPhone."
--0015174c3c60a73ef5046656ca27
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable
2011 Jan 10
0
No subject
takes precedence over a queue's defined moh class.
--=20
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--000e0ce0494051d402049b4247c1
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable
<div class=3D"gmail_quote">On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas =
<span dir=3D"ltr"><<a
2008 Mar 25
0
No subject
sort of standard for getting media players to support dynamically mixing
different tracks and also making it easy for artists to do.
On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote:
> i'll chime in and say that i would love to get music recorded in
> separate tracks, maybe there would be some kind of settings embedded
> in the files so i could hear them
2008 Mar 25
0
No subject
sort of standard for getting media players to support dynamically mixing
different tracks and also making it easy for artists to do.
On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote:
> i'll chime in and say that i would love to get music recorded in
> separate tracks, maybe there would be some kind of settings embedded
> in the files so i could hear them
2008 Mar 25
0
No subject
sort of standard for getting media players to support dynamically mixing
different tracks and also making it easy for artists to do.
On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote:
> i'll chime in and say that i would love to get music recorded in
> separate tracks, maybe there would be some kind of settings embedded
> in the files so i could hear them
2007 Jul 12
0
No subject
described (stop accepting calls and shut down when all calls have
completed). If you don't want to stop accepting calls, but still want to
stop Asterisk when there are no active calls, you can use "stop when
convenient". The same qualifiers ("gracefully" and "when convenient") can be
applied to the "restart" command.
Cheers,
AR
On Dec 10, 2007 7:29 AM,
2007 Apr 01
0
No subject
file. I don''t know why it will work for other hosts but these steps do not
work for this.. there are no extra outputs from using --debug
On 4/9/07, Atom Powers <atom.powers at gmail.com> wrote:
>
> On 4/9/07, Mike Zupan <hijinks at gmail.com> wrote:
> > I recently had a working puppet server serving around 4-5 clients. One
> of
> > the clients needed to