Displaying 20 results from an estimated 20000 matches similar to: "Save Key tone to MySQL DB"
2005 Feb 23
4
Vonage <---> Asterisk Working Config!
Hi Nitesh, check out my config that I have for the Faktortel config in
the asterisk@home sourceforge forum, you'll probably be able to work out
how to set it up from there.
Cheers,
Dean
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Wednesday, February 23, 2005 4:12 PM
To:
2007 May 23
16
WiFi SIP phones
Greetings list,
What are people's experiences with WiFi SIP phones?
When I last looked into them about 18 months ago, they were incredibly expensive, had very limited range and poor battery life. In the end, it worked out much more cost effective to simply use ATAs + DECT cordless phones where there was a requirement for portable devices.
I assume things must have moved on somewhat since
2007 Aug 11
1
LumenVox Speech Recognition
Hello All,
While looking for solution to solve my Callback DTMF problem, I came
across LumenVox Speech Recognition software.
Has anyone tried out? Need some feedback before I purchase it... Please
help...
Cheers,
Nitesh
2005 Mar 26
5
Click-to-Talk with Asterisk?
Hi Nitesh,
Take a look at this
http://www.microappliances.com/site/html/index.php?section=Products&page
=clienthowto.php
I've never implemented it though so I would appreciate some feedback on
if it works.
Cheers,
Dean
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nitesh
Divecha
Sent: Saturday,
2007 Jun 24
3
Nokia N95 + Dial Plan
Hello All,
Recently I added some Nokia N95 customers and it worked pretty good.
Now the customers are complaining about the dialing rules...
They are used to dialing +12486543210 and +4479XXXXXX for long distance
calls.
Is there anyway to create a "+" sign dial plan which will allow them to
dial a number with "+" sign.
Cheers,
Nitesh
2007 Jun 14
4
Que on A2Billing
Hello All,
I got one quick question on A2Billing.
Specs: -
- A2Billing v1.3
- OS CentOS 4.5
- Asterisk 1.2
- Zaptel 1.2
Did the installation and everything is working as it suppose to...
Using the A2Billing documentation, I created the RateCard, SIP Trunks,
and SIP Customers. I was also able to login using XLite Dialer and was
able to call out to my SIP Trunk also.
Now how can I remove the
2007 Jul 19
2
Upgrade Procedure
Hello All,
I would like to upgrade my recently installed Asterisk 1.2.21.1 to
Asterisk 1.4.8?
My OS is CentOS 4.5 with Linux 2.6.9-55.0.2.plus.c4smp #1 SMP Fri Jul 6
05:25:07 EDT 2007 i686 i686 i386 GNU/Linux
Is there any detail step by step procedure to uninstall the current
version and install Asterisk 1.4.8, Zaptel 1.4.4, Libpri 1.4.1, Addons
1.4.2?
Cheers,
Nitesh
2005 Jul 18
5
TDM04B - Takes long to initialize...
Hello All,
I got my TDM04B card installed and configured.
Everything works fine I can receive calls and route to appropriate
extensions.
The only problem I am facing is Slowness.
When I dial the PSTN number which is connected to Zap 1-1 after two
ring it answers and then run the AGI script. What I did was assign it
to a specific extension. So all inbound call on that PSTN number
should
2008 Jan 02
1
Password protect a queue from callers?
Hi, We currently testing a trixbox/asterisk installation and have used Freepbx to set-up and configure the box and it is running tremendously well. We have an generic IVR configured to which can transfer callers to a child IVR. This child IVR has a number of options to send the caller off to various queues. However we would like to protect some of the options with a password/pin number so that
2007 Jun 20
1
Asterisk RealTime
Hello All,
I manage to configure Asterisk RealTime and now it loads the SIP
users/peers from MySQL DB. The table I am using is of A2Billing DB
"cc_sip_buddies".
Now the only problem I am facing is incoming calls are failing... The
ATA which is assigned this DID number is behind NAT and according to
Olle's explanations he said "*there's no support for NAT keep-alives
2005 Jun 05
1
DTMF Tone Lengths
Good day,
I am hoping that someone can assist me with a work around.
I have an IVR system that I am attempting to connect asterisk
to, however the IVR was written
Some time ago, and requires tones to be approximately 2 seconds
in length. I am using SIP
Phones, Is there any way to adjust the Length of the Tones that
asterisk sends out the audio path?
Are there any phones that would allow
2014 Nov 25
0
Prohibit transfer to one extension
Hello all, first post, need help. I'm running a complex asterisk 1.8 install with five machines. I inherited it and don't fully understand it, nor the deep mysteries of asterisk either. I would appreciate any insight you might have. I scoured the 'net and the Digium wiki and my Google-Fu has failed me.
I've been asked to somehow prohibit transfers to extension 3232. It has to be
2007 Jul 19
0
Blank Voicemails/Vonage Problem
Regarding this message, I've actually been told one caller who has
consistently had this problem was using Vonage, but calling from his
Verizon line, it worked. This skewed my survey.
Therefore I do believe it's the same callers having the issue, and in
which case, I think Vonage is to blame.
I found this thread:
2008 Feb 21
1
IVR No sound on other provider
Hi All,
I have setup 2 trunks using 2 different voip providers using sip.
the first one i have no problem calling inbound then redirected to an
IVR, i can hear the IVR.
the second one has issues, inbound works going to IVR as i can see it on
the CLI, but i don't hear anything. i tried redirecting it to an
extension not an IVR just to see if inbound really works, and it rings
the
2015 Jun 16
0
small pbx for the office [it was: small homebrew pbx]
On 16/06/15 08:52, lucio at sulweb.org wrote:
> Steve Edwards wrote:
>> 0) I hope you mean you want to run Asterisk at home instead of
>> 'Asterisk at Home.' A at H was an ancient distribution from around 2005.
>
> Yes of course I didn't mean an ancient distro from 2005.
>
>>
>> 1) Rent a DID (a 'PSTN number') from a reputable SIP provider.
2004 Apr 26
3
dtmf tone clamping in calls to external ivr
Hello,
I'm having trouble working out how to send DTMF tones to an external
IVR. My system has an analog phone connected to a TDM400P card, a SIP
software phone (Zultys LIPZ4) and is connected to a BRI in Australia
with a NETjet-S card. I'm using ISDN4Linux and a 2.4.25 kernel patched
with the ISDN audio patch from Traverse (which allows the card to do
voice).
DTMF works fine between
2016 May 03
2
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Whoops, email client auto-filled dev previously. Let's try this again.
Michael Maier wrote:
<snip>
> Ok - but this doesn't seem to answer my main question:
>
> Why must
>
> progressinband=never
>
> be applied especially if asterisk uses it by default? The big difference
> between w/ and w/o it is:
The default in 13 is "no" which still
2010 Jul 14
2
Distinctive ring for INTERNAL calls only? How to do it?
Hi Everyone,
Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
how can one receive distinctive ring tones for INTERNAL calls ONLY?
Even though FreePBX Inbound has an option for Alert_INFO but that doesn't
work when the call comes into an IVR or Queue. The calls has to go directly
to extension for external ringtone to be different. So, I am looking for
internal calls
2016 May 03
3
Migrating asterisk 11 to 13: some callers get no ringback tone any more
Hello!
I migrated asterisk 11 to 13 as user of FreePBX 12.0.76.2.
As customer of German Telekom, I have three numbers and therefore three
trunks - each number is bound to one trunk.
After migration, some callers complained about missing ringback tone:
they didn't hear any ring tone and where surprised that they suddenly
got me anyway :-). The connection afterwards was as expected.
Deeper
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All,
I am trying to setup a small system where Nextone Softswitch will send
traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog
Gateway but for some odd reasons the call are flashed back from
Grandstream to Asterisk and creating a Black loop...
I did follow the instructions provided by Grandstream support but it
doesn't seems to be working...