On Sat, Jan 09, 2010 at 11:33:06PM -0800, Nitin Bahadur wrote:
> Hi,
>
> I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing
> a fast clicking sound and now I am not hearing any dial-tone. The FXO
> card has 2 slots: [phone | line ]. I hve connected the wall-phone-input
> to th
Hi,
I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing
a fast clicking sound and now I am not hearing any dial-tone. The FXO
card has 2 slots: [phone | line ]. I hve connected the wall-phone-input
to the "line" slot and "phone" to my home-phone. I do not hear any dial-tone
on
On Saturday 09 January 2010 15:22:29 Benoit wrote:
> I'm playing around with asterisk 1.6.2.0 and the first try was to
> replace my now non-functionning
> 'app-realtime' macro which emulated RealTime with REALTIME_HASH()
>
> There is very few documentation on the subject except for this bug report:
On Sat, Jan 09, 2010 at 01:22:09PM -0200, Valter Nogueira wrote:
> Hi folks,
>
> I just want to give a small contrib to newbies on Asterisk.
>
> To install (or upgrade to) Asterisk-1.4-current on Ubuntu boxes I wrote a
> small script.
>
> To use it, just open a terminal window and type:
aptitud
On 01/10/10 05:03, --[ UxBoD ]-- wrote:
> Hi,
>
> I use VoIPTalk as my provider and unsure of a minor issue. When people call
> me they get a US ring tone instead of UK. Is this a Asterisk configuration
> issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks.
>
>
This is almost a
I want to play soft music in the background while the IVR passes
through various contexts.
In short, I need to mix the script with music and my pre-staged .gsm
or .wav audio.
What tools to I need to use in Asterisk to make this happen please?
exten => s,1,Answer()
;exten => s,n,system(echo "${DATE
Hello,
I just updated my Asterisk Box from 1.6.1.0 to 1.6.1.12 and I have a single
queue to handle the customers, but with this upgrade every time the agents
phone rings it generates a NO ANSWER different CDR entry that is pretty
annoying. Checking the changelog I saw there was a merg in the co
Hi,
I'm playing around with asterisk 1.6.2.0 and the first try was to
replace my now non-functionning
'app-realtime' macro which emulated RealTime with REALTIME_HASH()
There is very few documentation on the subject except for this bug report:
https://issues.asterisk.org/view.php?id=13651
Hi,
I use VoIPTalk as my provider and unsure of a minor issue. When people call me
they get a US ring tone instead of UK. Is this a Asterisk configuration issue
or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks.
Thanks - Phil
--
- "Tilghman Lesher" wrote:
| On Saturday 09 January 2010 08:05:21 Doug Lytle wrote:
| > Michelle Dupuis wrote:
| > > What do you mean internal timing?
| >
| > He probably means he didn't have a timing source setup, a.k.a
| ztdummy or
| > dahdi_dummy.
|
| In the case of 1.6.1 and forward, the
On Saturday 09 January 2010 08:05:21 Doug Lytle wrote:
> Michelle Dupuis wrote:
> > What do you mean internal timing?
>
> He probably means he didn't have a timing source setup, a.k.a ztdummy or
> dahdi_dummy.
In the case of 1.6.1 and forward, the timers are separate modules in Asterisk
and no lon
Hi folks,
I just want to give a small contrib to newbies on Asterisk.
To install (or upgrade to) Asterisk-1.4-current on Ubuntu boxes I wrote a
small script.
To use it, just open a terminal window and type:
>wget http://www.fastway.com.br/downloads/install_asterisk-1.4.sh
>chmod 755 install_ast
Michelle Dupuis wrote:
> What do you mean internal timing?
>
>
He probably means he didn't have a timing source setup, a.k.a ztdummy or
dahdi_dummy.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor S
What do you mean internal timing?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[ UxBoD ]--
Sent: Saturday, January 09, 2010 8:11 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Choppy MOH
- "
- "--[ UxBoD ]--" wrote:
| Hi,
|
| I am running Asterisk 1.6.2.0 and am finding that MOH is playing quite
| choppy. I have enabled format_mp3 from the add-ons package and
| removed mpg123.
|
| Any pointers please ?
|
| --
| Thanks - Phil
|
|
|
| --
| __
Hi,
I am running Asterisk 1.6.2.0 and am finding that MOH is playing quite choppy.
I have enabled format_mp3 from the add-ons package and removed mpg123.
Any pointers please ?
--
Thanks - Phil
--
_
-- Bandwidth and Colocat
Dear All
My Asterisk has sip connection with an external sip server
@192.168.0.139. I have sip inbound and outbound calls as ok . But
there is a problem on
sip incoming calls . To illustrate the problem , please suppose the sip
phone on external sip server dials my Asterisk sip phone @6672019 . Ple
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