Displaying 12 results from an estimated 12 matches for "sasami".
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masami
2003 May 13
9
Semi-ot: voip provider with 800-service?
Semi-offtopic,
Anyone know of voip providers who can provide tollfree number service?
E.g. route 800-xxx numbers to our * ?
Even better if they are familiar with * or can speak IAX ...
-Dan
2005 Aug 15
2
warning: dovecot list is being harvested
...t listmembers. The list is being harvested.
test3943395 is a unique address I created only for communication to the
dovecot list.
The following spam came from:
Received: from dial-dynamic-62-69-52-187.surfdial.murphx.net (dial-dynamic-62-69-52-187.surfdial.murphx.net
[62.69.52.187])
by sasami.anime.net (8.11.6/8.11.6) with SMTP id j7FLxtv03775
for <test3943395 at anime.net>; Mon, 15 Aug 2005 14:59:56 -0700
Date: Mon, 15 Aug 2005 22:59:46 +0100
From: Neateye <nitaigouranga at aol.com>
To: Test3943395 <test3943395 at anime.net>
Subject: Gouranga
Call out Gouranga...
2004 Jun 21
0
If you're against .ogv, don't bother even reading this. Was (Re:
[Vorbis] Extension proposal - partly serious)
In-Reply-To: <Pine.LNX.4.44.0406201942590.21207-100000@sasami.anime.net>
References: <Pine.LNX.4.44.0406201942590.21207-100000@sasami.anime.net>
Message-ID: <40D70E0E.5090807@ellisfoundation.com>
>Anyone else want to take this discussion to another mailing list? We can
>hash out a standard there without interference from the people here...
2003 Sep 25
1
additional ports for devkit tdm400p(tdm10b) ?
I got the devkit tdm400P(tdm10b) + x100p bundle.
Where does one buy more ports for the tdm400p? I couldnt find anywhere on
digium site or resellers to buy them?
-Dan
2004 Apr 26
3
Sipura SPA-3000
Anyone have one of these yet?
http://www.voxilla.com/shop/index.php?action=item&id=38
-Dan
2003 Sep 24
8
VIA vs Intel
Has anyone successfully run asterisk with a VIA processor ?
I have tried unsucessfully, do I have to run make with any specific switches
?
2006 Mar 24
8
Snom 360 problems
Anyone have a Snom they're happy with? How did you manage that? :)
I have a system of:
Asterisk 1.2.3
2 Wildcard TDM400P Rev I and E/F
1 Snom 360 + sidecar
~15 Sipura/Linsys SPA-841
~15 Grandstream 101
Everything (currently) is on the same network, not a router to be seen
between any two. Also everything, except the snom, is working sweetly.
The main problem is ECHO.. awful echo and
2003 Sep 12
2
problem with * and Howlink CL-100 ip phone
I'm trying to use a Howlink CL-100 ip phone with *
It's h323 phone with very limited protocol support. But it's enough that I
can use it to dial netmeeting client and artisoft pbx just fine.
When I try to dial my * with it using either chan_h323 or oh323, it seems
to fail on negotiating H245. Maybe this phone doesn't support it?
I've used all different versions of
2006 Mar 02
0
RE: Asterisk-Users Digest, Vol 20, Issue 13
...-----
Message: 7
Date: Thu, 2 Mar 2006 10:26:11 -0800 (PST)
From: asterisk@anime.net
Subject: Re: [Asterisk-Users] Re: sipura 841 mass provisioning
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <Pine.LNX.4.63.0603021025520.12962@sasami.anime.net>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
On Thu, 2 Mar 2006, Matt wrote:
> I am guessing you need to escape the <'s. Possibly with a \ but I'm
> not sure. So
> |\<9;\>
No. Use > and <
-Dan
------------------------------...
2006 Mar 27
0
Question about Polycom 601 and expansion module.
...?
>
>
>
> ------------------------------
>
> Message: 11
> Date: Mon, 27 Mar 2006 11:55:39 -0800 (PST)
> From: asterisk@anime.net
> Subject: [Asterisk-Users] queue caveats
> To: Asterisk-Users@lists.digium.com
> Message-ID: <Pine.LNX.4.63.0603271153340.18525@sasami.anime.net>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> According to http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue, under
> the "Notes" section:
>
> "Transfers of calls that are answered out of a queue must be done using
> Asterisk...
2004 Apr 06
0
Ericsson Webswitch 100 Model G4 with * ?
Does anyone have info on using the Ericsson Webswitch 100 Model G4 with *?
The documentation and brochures make it sound like it's some kind of
specialized FXO device purely to provide legacy PBX with access to VOIP,
and not a general purpose FXO-VOIP gateway.
Is it possible to use it to route incoming PSTN calls into * with full
callerid, and establish outgoing calls from * to the PSTN