Displaying 20 results from an estimated 4000 matches similar to: "No subject"
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:
>
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
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
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
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
>
2007 Jul 12
0
No subject
tnet.itand SIP register messages are not replied.
I suggested to check if your Asterisk box is really sending SIP messages,
you can use a net sniffer.
Did you alerady used different sip client with the same sip account of your
Asterisk box?
Did you use zoiper from the same box?
Marino
p.s.
Are you Italian?
On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo <
gincantalupo at
2018 Mar 30
1
Tinc: performance
2011 Sep 02
0
No subject
penSuse 12.1. Lets check with OpenSuse 12.1.
<div><br />
</div>
<div>Regards.</div>
<div><br />
<br />
<div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan =
N <span dir=3D"ltr"><<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=
2011 May 13
0
[LLVMdev] [ptx] Propose a register class naming convention change
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Justin Holewinski wrote:
<blockquote
cite="mid:BANLkTi=Y9EFmWRu-9dQxydq8zTyF7tEbJw@mail.gmail.com"
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
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<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
2009 Jul 20
0
No subject
-uzzi
PS: If you're not seeing any connection information, be sure to double-check
the IP address is correct. Learned that lesson the hard way =\
On Sun, Jan 31, 2010 at 5:51 PM, Jim Rosenberg <jr at amanue.com> wrote:
> Let's say I have two Asterisk boxes, A and B. I am trying to get A to do
> SIP registration on B, so an extension for A can dial SIP phones covered by
>
2007 Aug 16
0
No subject
sses, that way autoloading works ok and the classes are found, but that see=
ms a bit awkward.
<br></div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid=
rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br><br=
>Note that it's a bit redundant to name your classes that way -- you<br=
>
can just as
2011 Apr 12
0
No subject
be able to setup a SIP trunk. I've been able to successfully integrate a
Cisco CallManager 7.x system with Asterisk using SIP trunking, so I imagine
you should be able to do the same here.
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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<div
2007 Jul 12
0
No subject
2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant <russell at digium.com>
* CREDITS, include/asterisk/http.h, main/tcptls.c (added),
main/manager.c, channels/chan_sip.c, doc/siptls.txt (added),
main/Makefile, main/http.c, include/asterisk/tcptls.h (added),
configs/sip.conf.sample, CHANGES: Merge changes from
team/group/sip-tcptls This set of changes