Not the most elegant but since I have a generic context for my IVRs I
simple check the date there.
exten => s,n,GotoIfTime(*|*|1|jan?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|10|apr?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|25|may?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|3|jul?closed-
> So at this point, it seems like it boils down to this:
> Soekris
> PCEngines
> Atcom (IP01: £160+VAT)
> Herologic (HL-463: $259)
> uCpbx (235.00 EUR)
> AstBoxes (168.00 EUR)
> Gumstix
> HP Thinclient t5720
I recently custom built an Intel Atom 330 Mini-ITX based system for a client
who then too
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of James Van Vleet
> Hmm would this somehow get to the the D channel and stop incoming
> calls from coming in the PRI? My box does not show the D channel
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Cary Fitch
> Can any one suggest a little code to either ring a cell phone when a
> new VM
> message is recorded, or send a text message?
>
> Basically out
Not an LDAP user but perhaps using AGI to access your LDAP server may be
a solution? Looks like it may require some work but I wouldn't think it
to be too hard.
sl
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Klaverstyn
Se
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of David Backeberg
> Sent: Monday, September 21, 2009 10:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] dig
Mike –
It looks like you have externip set but no localnet setting.
You need to set localnet for your internal networks so that Asterisk knows when
to properly apply the externip setting.
sl
___
-- Bandwidth and Colocation Provided by http://www.api-d
Mike -
Uncomment and set appropriately for your network. If you're using
192.168.1.0/24 as your internal network then that's what it should be set to.
Be sure to include any private networks that may interact with the server over
VPN or private circuits as well.
Then be sure to reload or resta
Subject: Re: [asterisk-users] INVITE Sending Local IP
Still no luck. I'm almost ready to start over with a fresh sip.conf and
extensions.conf. Does anyone kno where I can find one without all the comments
and other "fluff"?
On Thu, Oct 1, 2009 at 9:22 AM, Scott L. Lykens
Mike -
If your router/firewall does not have any kind of SIP protocol-specific support
then you need to set up port forwarding on your router.
Forward udp/5060 for signaling, and the matching udp ports as listed in your
rtp.conf, to your Asterisk box. Keep the externip and localnet settings in
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
> I am working on getting this situation resolved and should have new
> releases of FFA out at the end of this week, but in the meantime if
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
> Sent: Monday, October 05, 2009 10:21 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] digi
Apologies for the top post - Outlook really is braindead with HTML
email.
I've been thinking about this problem for a project I am working on and
what I think I am going to do is create a table that I insert a record
into just before the dial statement that includes where the call is
going then
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
> Sent: Monday, October 19, 2009 9:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] digiu
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Bill Shaw
> Sent: Tuesday, November 17, 2009 11:34 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] newbie question
> When typing 'help
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Thorolf Godawa
> Sent: Thursday, December 03, 2009 2:49 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Source-I
> Subject: Re: [asterisk-users] Mixing commercial/SVN Asterisk
>
> On Dec 16, 2009, at 10:08 AM, Richard Kenner wrote:
>
> > Am I correct that if I'm running an -rc or from an SVN release tree
> > that there's no way I can use any commercial add-ons from Digium,
such
> > as Skype, Cepstral, or G.
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Neeraj Chand
> Sent: Monday, January 04, 2010 1:17 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] MYSQL queries from dial plan
[mysql
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Thursday, January 07, 2010 10:30 PM
> What about:
>
> 1) Fixing the slow responding DNS server?
>
> 2) Tweaking /etc/resolv.conf opt
Not sure it is relevant, however, I have an E52 I use occasionally with
my * and I've found that without an active SIM in the phone the SIP
profile will ring silently.
I'm sure there's a way to fix it I just haven't been bothered enough to
work on it.
sl
From: asterisk-users-boun...@lis
> My idea is to use a well know port like port 80 (that is not blocked).
Skype for example uses this port.
If you are in a situation where the ISP/government is blocking VoIP you
are probably going to have to encrypt it to get it through, and that may
not even work. I have a client who has facili
0 Machine check exceptions
MCP: 1 1 Machine check polls
ERR: 1
MIS: 0
Any ideas on how I can further diagnose and pursue this? Google does not reveal
much related to this issue that is useful.
Thank you!
--
Scott L. Lykens
Keystone Medical Management So
00 0 0 200 420 0 0 99 0 0
0 0 0 7714712 42752 17632400 0 4 205 414 0 0 99 0 0
0 0 0 7714760 42804 17632400 023 216 430 0 0 98 2 0
0 0 0 7714756 42812 17632400 0 4 201 409 0 0 99 0 0
Tha
On Jun 1, 2014, at 11:01 AM, jg wrote:
> Yes, I can see this. Another thing to check would be to start from a
> different OS (eg from a USB stick) and see how the card behaves on the
> otherwise same hardware.
>
> Since your ProLiant G2 server is almost 10 years old, and the TE410P works
> w
On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere
mailto:j...@jeff.net>> wrote:
I wrote earlier today about a new PRI installation in the Caribbean, where all
outbound calls are functioning fine *except* calls to Sprint phone numbers,
which get rejected immediately as "busy".
I don’t know what ex
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