Displaying 20 results from an estimated 122 matches for "shmaltz".
2005 Sep 06
2
Asterisk overheating on VIA Epia MSeriesmoth erboard
...) or an OS setting?
I run a lot of Via C3 machines (they are so nifty) but don't remember seeing
this.
-----Original Message-----
From: Technical Support [mailto:support@ocg.ca]
Sent: Tuesday, September 06, 2005 10:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion';
shmaltz@gmail.com
Subject: RE: [Asterisk-Users] Asterisk overheating on VIA Epia
MSeriesmotherboard
You can dramatically reduce the heat from your EPIA board by turning on CPU
scaling! Once we turned it on, the heatsink was cool to the touch. (Even
with asterisk running).
MD
2007 Jan 03
7
SNOM loses server registration
Hello to all
When my SNOM (300 or 320) loses Internet connectivity, it loses its
Asterisk registration (ok, thats normal).
But when the phone is back online, he doesn't try to register in
Asterisk. I believe this happens to avoid flooding the private LANs when
the Internet link is lost.... but the problem is that the phones don't
try to re-register in the future.... Sometimes it stays
2006 Jun 20
10
TE420P/TE415P?
Hi,
I just read a pressrelease from VON that Digium will soon be releaseing
a couple of new cards. What got me interested was: "The TE420P and
TE415P support 128ms of G.168 (2002)-compliant echo cancellation across
their entire 128 channels."
Does anyone know when thease will be released and what they will cost
when released? Thanks!
2007 Feb 06
4
Something wrong with the list?
Since Monday I didn't see much traffic.
2006 Dec 18
3
Billing solution
Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?
I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor does it have
to allow for customers the ability to log in to check their
usage/balances.
I have
2004 Dec 20
2
Grouping SIP channels (Sipura 3000)
Does any body know if it is possible to group SIP channels just like
it is possible with Zap channels? I have a group of FXO gateways
(Sipura 3000's) and I would like to treat them as a group the same as
I would Zap channels. Does anyone know if this is this possible?
2006 Apr 17
1
voicemail use external smtp server for sendingmail
...lem getting Sendmail to use a smarthost, but am
I understanding the Asterisk part of this properly, or is there a way to
get Asterisk to DIRECTLY use a smarthost, so that Sendmail doesn't have
to be running on the local Asterisk box?
Thanks!
-Steve
-----Original Message-----
From: C F [mailto:shmaltz@gmail.com]
Sent: Saturday, April 15, 2006 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voicemail use external smtp server for
sendingmail
Yes, just configure your sendmail to do it.
On 4/13/06, nik600 <nik600@gmail.com> wrote:
> is i...
2008 Mar 03
2
Switchvox feedback
I have a customer that wants to get switchvox, since I have never used
it, I would like to hear some feedback from active users of switchvox.
In specific:
1. Does it use realtime or conf files
2. Is it possible to change it manually?
3. Is SSH access to login to console/shell available?
4. Are you or your customers happy with the user interface?
TIA
2005 Sep 12
5
What have I misconfigured?
I'm getting these messages every 7-10 seconds.
-- Registered SIP '532' at x.x.x.x port 52956 expires 60
-- Registered SIP '532' at x.x.x.x port 56988 expires 60
-- Registered SIP '529' at x.x.x.x port 51444 expires 60
-- Registered SIP '529' at x.x.x.x port 64044 expires 60
-- Registered SIP '532' at x.x.x.x port 52956 expires 60
-- Registered SIP
2005 Jan 11
1
internal caller id on analog phones connected tozap
> -----Original Message-----
> From: C F [mailto:shmaltz@gmail.com]
> Sent: Tuesday, January 11, 2005 4:38 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] internal caller id on analog phones
> connected tozap
>
> How are the analog phones connected to * ? this is where the setting
> s...
2005 Sep 06
1
Asterisk overheating on VIA Epia MSeriesmotherboard
...C3 machines (they are so nifty) but don't remember
> seeing this.
>
>
> -----Original Message-----
> From: Technical Support [mailto:support@ocg.ca]
> Sent: Tuesday, September 06, 2005 10:43 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion';
> shmaltz@gmail.com
> Subject: RE: [Asterisk-Users] Asterisk overheating on VIA Epia
> MSeriesmotherboard
>
>
> You can dramatically reduce the heat from your EPIA board by turning
> on CPU scaling! Once we turned it on, the heatsink was cool to the
> touch. (Even with asterisk runn...
2004 Dec 28
3
Sending call to analog then to Vmail after timeout?
I have one analog line hooked in my Asterisk box using an x100p (I think that's the model number).
When I do this in my extensions.conf:
exten => 1200,1,playback(pls-wait-connect-call)
exten => 1200,2,Dial(Zap/1/5555551212,20,rTt)
exten => 1200,3,VoiceMail(u100@lightwavetech.com)
exten => 1200,4,Goto,t|1
The phone rings beyond the 20 second timeout and never really goes to the *
2007 Jul 12
0
No subject
..._33086575.1224161135545
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit
Content-Disposition: inline
<div dir="ltr"><br><br><div class="gmail_quote">2008/10/16 C F <span dir="ltr"><<a href="mailto:shmaltz at gmail.com">shmaltz at gmail.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
* Live call screening - Yes there is a hack that...
2006 Jan 16
2
Call Center and Predictive dialing
I know this question has been asked a lot before, but please I would
like to know from personal experience.
I'm looking to use Asterisk in a call center environment, where most
of the calls will be outbound calls. They will have at start 100
agents.
I have looked at vicidial and looks promising, however I would like to
hear from users what they use and how they like it compared to other
2007 Dec 11
1
Fw: asterisk performance
...e suppose to give high priority to RTP in our router? What sort of QoS and traffic shapping would you recommend?
5.. How many users can we expect to use voip(with good quality) with 512kbps outbound connection?
Regards,
jorain
Date: Fri, 7 Dec 2007 10:27:36 -0500
From: "C F" <shmaltz at gmail.com>
Subject: Re: [asterisk-users] asterisk performance
To: jorain <jorain at caliber.com.sg>, "Asterisk Users Mailing List -
Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Message-ID:
<81000b5a0712070727s32156f31y4986abc9054144 at mai...
2006 Feb 09
1
Possible for Asterisk to output CLID to invo ke 3rd party app?
Specifically:
exten => s,1,System(/usr/sbin/myperlscript.pl ${CALLERIDNUM})
will execute myperlscript.pl with the caller id as an argument as the first
priority.
hth
-----Original Message-----
From: C F [mailto:shmaltz@gmail.com]
Sent: Thursday, February 09, 2006 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Possible for Asterisk to output CLID to
invoke 3rd party app?
Have you tried System?
On 2/9/06, lists.digium.com@al-najjar.co.uk
<lists.digium.com@al-...
2007 Oct 19
1
Glare on Incoming Calls
...Hangup
exten => 44,n,Playback,ggestion/varios/moment
exten => 44,n,SetMusicOnhold(dialtone)
exten => 44,n,Set(TIMEOUT(response)=10)
exten => 44,n,Set(TIMEOUT(digit)=5)
exten => 44,n,WaitExten(25|m(dialtone))
> Date: Thu, 18 Oct 2007 17:07:03 -0400
> From: "C F" <shmaltz at gmail.com>
> Subject: Re: [asterisk-users] Incoming calls
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <81000b5a0710181407x54f4f7f2g6fbdd7aaba5845c8 at mail.gmail.com>
> Content-T...
2005 Sep 09
9
adding DNIS digits
Situation:
8 POTS lines, 3 companies, 1 system. Channel banking the POTS lines
onto a T1 thru an ADIT 600.
The only way our carrier will provide DNIS is thru Analog DID #'s.
Anyone know of a piece of hardware that can add DNIS digits to a
particular line?
-Darren
2006 May 23
6
Best VoIP provider for Asterisk
Hi Friends,
Can you please tell me who is the best VoIP Service Provider using Asterisk (With trail version for sometime) . Waiting for your quick response. Thank you.
Regards,
Chandra.
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2010 Aug 03
0
asterisk-users Digest, Vol 73, Issue 5
...e
are no re-invites issued too.
i am bit new to sip and rtp stuff and don't know what is going on. how
asterisk is issuing re-invites for devices behind same router and not for
device behind another router?
Nasir Javaid
Message: 12
> Date: Tue, 3 Aug 2010 07:21:06 -0400
> From: C F <shmaltz at gmail.com>
> Subject: Re: [asterisk-users] RTP stream not passing through router
> with port forwarding
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID:
> <AANLkTin9G14ipFL3yVM...