How can I mute the message "please enter the number you wish to call and
press the # key" in a2billing???
I tried
use_dnid = YES
but still I keep getting the message prompt...
thanks
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On Fri, Oct 22, 2010 at 9:24 PM, Jerry Geis wrote:
> I am running /usr/sbin/asterisk its an executable not a script or anything?
> How do I stop using safe asterisk?
>
Confirm that your init.d script is not referencing it. Otherwise, if
'core stop now' does not work, perhaps you have a deadlock.
>
> Stop using safe_asterisk, it has logic to restart killed processes.
>
Paul,
I didnt think I was running "safe_asterisk".
I am running /usr/sbin/asterisk its an executable not a script or anything?
How do I stop using safe asterisk?
Thanks,
Jerry
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_
On Fri, Oct 22, 2010 at 8:53 PM, Jerry Geis wrote:
> It seems to respawn itself. Even on a kill -9 it respawns itself.
>
Stop using safe_asterisk, it has logic to restart killed processes.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Fre
Hi!
> but all of a sudden we have all calls origination from one sip
> extension opening channels which have the name of another sip extension
> in the channel name.
Do the devices of this extension happen to have the same IP address?
Philipp
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In old 1.4 you can kill -SIGINT # and stop the process.
I dont seem to be able to do this in 1.8 any more.
I know about /usr/sbin/asterisk -rx "core stop now"
but in case it doesnt respond. How do I kill it .
It seems to respawn itself. Even on a kill -9 it respawns itself.
Thanks,
Jerry
--
On Sat, Oct 23, 2010 at 2:33 AM, Baha @ SH wrote:
> I mean sip trunk
>
Then this functionality is dependent on the type of SIP phone you have.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
__
I mean sip trunk
thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Friday, October 22, 2010 6:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-us
On Fri, Oct 22, 2010 at 7:10 AM, Baha @ SH wrote:
> How can I let asterisk immediately dials a trunk when off hook?
>
For DAHDI:
chan_dahdi.conf
[channels]
immediate=yes
channel => 1
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenod
On 10/22/2010 04:05 PM, Zeeshan Zakaria wrote:
> Hello list,
>
> (Resending this email due to a typo in previous copy)
>
> I need to do E1 to T1 conversion for a project, and was wondering if
> there exists a card with both E1 and T1 on it. Or is it possible to use
> two separate cards in an aste
Hello list,
(Resending this email due to a typo in previous copy)
I need to do E1 to T1 conversion for a project, and was wondering if there
exists a card with both E1 and T1 on it. Or is it possible to use two
separate cards in an asterisk box, one for E1 and one for T1? (Please don't
mention ac
Hi,
I have been experimenting with CEL in a trunk version of asterisk for some time
and have upgraded my test machine to 1.8.0 today.
Made a few calls and it looks like the eventtype field is missing in the CEL
insert query when using ODBC. I see the following errors on the console:
[Oct 22 21
Hello list,
I need to do E1 to T1 conversion for a project, and was wondering if there
exists a card with both E1 and T1 on it. Or is it possible to use two
separate cards in an asterisk box, one for E1 and one for T1?
(Please don't mention aculab or adtran)
Zeeshan A Zakaria
--
www.ilovetovoip
I am helping a friend on one of his sip trunk and couldn't find the way
to resolve his problem.
His asterisk's problem is like this:
0. When incoming call to one of his sip trunk, Asterisk reply with "488
Not acceptable here". So the call get dropped.
1. Recently upgraded Elastix with Asterisk 1
Rob, you are the man. Thanks for pointing me in the right direction.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-10-22 12:28 PM, "Rob Coward" wrote:
Any reason you cant change the asterisk server to bond the 2 nics together ?
We use bonded nics a lot to provide re
On Fri, Oct 22, 2010 at 10:02 AM, Bruce B wrote:
> Hi Everyone,
> For some reason a few phones connected to a pfSense box can't make or allow
> in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP
> and it works fine. I would like to know if there is any disadvantages to
> usi
Any reason you cant change the asterisk server to bond the 2 nics
together ? We use bonded nics a lot to provide resilient networks, and as
far as any apps on the server are concerned, you are only talking to a
single interface bond0 instead of eth0 and eth1.
Rob
On Mon, 18 Oct
2010 17:03:45
I am using a sip softphone
But anyway, my question in specific:
If my sip phone registered as a trunk, then went off-hook, is there anything
that asterisk run or recognize that this trunk now is offhook?
I am doing this because I want it to go to a a2b right after hook-off, I
don't want to press
Hi Everyone,
For some reason a few phones connected to a pfSense box can't make or allow
in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP
and it works fine. I would like to know if there is any disadvantages to
using TCP over UDP for the tunnel when using Asterisk or is ju
Hi,
When upgrading-downgrading from one asterisk version to another, you may
need to edit config files.
For instance, when upgrading sip.conf from 1.4 to 1.8, it is tempting to use
a single common file and add some #ifdef-like statements in it so that
1.8-specific statements are not mixed with 1.4
On 22 October 2010 14:24, Miguel Molina wrote:
> I think the OP is asking for the old MoH sound (fpm-world-mix, etc) that
> came with asterisk. I wonder why the change from the fpm sounds to the
> opsound ones, it was a licensing issue?
>
I think the original 'fpm' files were not as freely licenc
El 22/10/10 07:06, Tzafrir Cohen escribió:
> On Fri, Oct 22, 2010 at 12:44:18PM +0300, Aurimas Skirgaila wrote:
>> Hi,
>>
>> I wonder if I may freely use the default soundfiles that came with asterisk
>> (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server?
>>
>> Are there any offi
Hi.
We use GROUP and GROUP_COUNT to track if the peer is engaged in a call. If
so we use Busy()
On 22 October 2010 01:28, GBR Icasiano, Ryan A. <
raicasi...@globalbridgeresources.com> wrote:
> Hi,
>
> I have modified the way agents are being treated since they are using
> mobile phones. Having t
Hi
I'm using asterisk 1.4.17 and have recently found an odd issue. When
processing the CDR data on outbound calls I've been using the channel
field to extract which sip extension has made the call as I was under
the impression that the channel name for SIP channels was always
SIP/-
This has work
This can be done through some agi scripting, for example in asterisk-java
you can get the channel events if the channel event is off-hook you can
initiate the server to dial a call. With normal asterisk dial plan I dont
think so it can be done. But through manager API or AGI it is possible:)
Thanks for this info.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 7:45 AM, "Andrew Latham" wrote:
Have a look...
http://www.digium.com/en/company/casestudies/
Contact John Todd jt...@digium.com with your case studies...
~
Andrew "lathama" Latham
lath...@gmail.com
On Fri, Oct 22
On Fri, 22 Oct 2010, Zeeshan Zakaria wrote:
> I think you are the first person ever to ask this question. Of course you
> can use them, they are royalty free for a purpose.
Not the first person and I recall that the music source was changed
recently due to some countries not honouring the royalt
Thank you guys for making me sure about this question and pointing to useful
resourses.
* yes, I might be the first one because googling didn't give me any certain
answer
On Fri, Oct 22, 2010 at 3:06 PM, Tzafrir Cohen wrote:
> On Fri, Oct 22, 2010 at 12:44:18PM +0300, Aurimas Skirgaila wrote:
>
On Fri, Oct 22, 2010 at 12:44:18PM +0300, Aurimas Skirgaila wrote:
> Hi,
>
> I wonder if I may freely use the default soundfiles that came with asterisk
> (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server?
>
> Are there any official sources of royalty free music?
http://downl
Have a look...
http://www.digium.com/en/company/casestudies/
Contact John Todd jt...@digium.com with your case studies...
~
Andrew "lathama" Latham
lath...@gmail.com
On Fri, Oct 22, 2010 at 8:26 AM, Zeeshan Zakaria wrote:
> I didn't know about Digium's cool case studies. Will my realtime virtu
I didn't know about Digium's cool case studies. Will my realtime virtual PBX
with partially javascript based GUI and Voice Reminder service fit into cool
case study?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 7:17 AM, "Andrew Latham" wrote:
The sound files for MOH, just like the vo
I think you are the first person ever to ask this question. Of course you
can use them, they are royalty free for a purpose.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-10-22 5:53 AM, "Aurimas Skirgaila" wrote:
Hi,
I wonder if I may freely use the default soundfiles that came with asteri
The sound files for MOH, just like the voice files of Alison and
others are open and free. You of course can always donate your
royalty free sounds or pay for some new sounds. If your language is
not included in Asterisk, please contact a quality voice actor and
submit some sound files for your l
I think you will have better response if you can provide the actual
dialplan text file, instead of the format you
attached.
Regards
HASSAN
On 2010-10-22, Jigar Joshi wrote:
> I am reattaching the file one of the svg file is pending for moderator's
> approval,
> but I am here attaching vdp file.
I am reattaching the file one of the svg file is pending for moderator's
approval,
but I am here attaching vdp file.
On Thu, Oct 21, 2010 at 3:13 PM, Jigar Joshi wrote:
> It seems to have some server configuration with it, Its not getting parsed
> if i stop server.
>
> I am attaching svg format
Hi,
I wonder if I may freely use the default soundfiles that came with asterisk
(fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server?
Are there any official sources of royalty free music?
--
Mvh,
Aurimas Skirgaila
--
On 22/10/10 11:05, Hans Witvliet wrote:
> On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote:
>> On 21/10/10 22:04, Hans Witvliet wrote:
>>> For suse there is a precompiled version on the OBS (vitsoft)
>>>
>>
>> Package search on the OBS shows nothing for 1.8.0 at all.
>> Perhaps you know where i
On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote:
> On 21/10/10 22:04, Hans Witvliet wrote:
> > For suse there is a precompiled version on the OBS (vitsoft)
> >
>
> Package search on the OBS shows nothing for 1.8.0 at all.
> Perhaps you know where it is hidden.
>
> Dave Cotton
>
> http://s
On Fri, 22 Oct 2010, Baha @ SH wrote:
> How can I let asterisk immediately dials a trunk when off hook?
If it's a SIP phone then read the manual for the phone. Every one is
different.
If it's an analogue phone connected to a TDM400 type card, then google for
batphone mode.
Gordon
--
_
On 21/10/10 22:04, Hans Witvliet wrote:
> For suse there is a precompiled version on the OBS (vitsoft)
>
Package search on the OBS shows nothing for 1.8.0 at all.
Perhaps you know where it is hidden.
Dave Cotton
--
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