On Mon, Jul 18, 2011 at 11:02:16AM +0530, mahesh katta wrote:
Sorry boss
Best Regards,
Mahesh, I'm afraid that at some point Ashirwad will become annoyed
that you are including the asterisk-users list on these emails.
--
Barry
--
Hello
I'd like to build a compact, affordable, fanless x86 solution to
handle my home landline.
I know about the following two platforms:
1. www.pcengines.ch/alix.htm
alix1d + case 100
Does Availability 500 mean that it's just not possible to buy just
one item?
2.
- Original Message -
On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
--[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing
as
A J Stiles asterisk_l...@earthshod.co.uk writes:
Really, building packages from source *IS* *NOT* *HARD*, and it doesn't even
take long anymore (on any target system with the grunt to run Asterisk).
The only thing to beware of is, if configure complains that you need a
package that you
Hi, is anyone else having problems with the reload command crashing
Asterisk 1.6.2.19? I've run a few tests and 1.6.2.18.2 doesn't have
this problem but 1.6.2.19 after a few reloads is just dumping and
restarting.
Thanks
Lee
--
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu celular
Eduardo Carpes
E-mail: car...@bsd.com.br
www.freebsd.org
--
_
-- Bandwidth and
Gilles wrote:
Does Availability500 mean that it's just not possible to buy just
one item?
I read it as they have over 500 in stock.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
On 18 July 2011 12:03, Lee Archer lee.arc...@thebigword.com wrote:
Hi, is anyone else having problems with the reload command crashing Asterisk
1.6.2.19? I’ve run a few tests and 1.6.2.18.2 doesn’t have this problem but
1.6.2.19 after a few reloads is just dumping and restarting.
Thanks
On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote:
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu celular
Eduardo Carpes
E-mail: car...@bsd.com.br
www.freebsd.org
The magic sauce that you
Hi Steve, I think it's related to my ODBC connection. I started with a random
hang where it looked ODBC related which led me to try a few things. Reloading
the config a few times is causing core dumps which 1.6.2.18.2 just doesn't
have, however my main reason for using 1.6.2.19 is a fix to
Just about any of the HP thin clients, either new or used off eBay, with
AstLinux installed do a wonderful job, especially if you are not going
to need a PCI card.
The older units will need a larger flash. Transcend has several
different sizes that are direct replacements
Looks like some of
On Mon, 18 Jul 2011 08:04:31 -0400, John Novack
jnov...@stromberg-carlson.org wrote:
Just about any of the HP thin clients, either new or used off eBay, with
AstLinux installed do a wonderful job, especially if you are not going
to need a PCI card.
The older units will need a larger flash.
First they came and said that instead of offices, doors and hallways,
we should have massive, open-plan seating or grungy, industrial
cubicle farms, because open spaces mean open companies!
It's safe to say the advice did not fall on deaf ears. Now, we're
ready to take openness to the next
Sent from my Computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Lee Archer
Sent: Monday, July 18, 2011 7:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Seg Faults with
Boy if only it was Enron :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, July 18, 2011 8:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Requires
First
Hi Eric, are you using ODBC?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
Wieling
Sent: 18 July 2011 13:54
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
HP still does make Thin Clients, often with XP Embedded, though I have
had very good results with many older used ones sold on eBay. With a new
larger flash from Transcend, they simply work. Consider the used units
not worn out but simply ones with more experience that probably won't
fail. New
Seems to be an already reported problem but since no more fixes for 1.6
it's back to 1.6.2.18.2 for me.
https://issues.asterisk.org/jira/browse/ASTERISK-18103
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On 18 July 2011 13:00, Lee Archer lee.arc...@thebigword.com wrote:
Hi Steve, I think it's related to my ODBC connection. I started with a
random hang where it looked ODBC related which led me to try a few things.
Reloading the config a few times is causing core dumps which 1.6.2.18.2 just
On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote:
Seems to be an already reported problem but since no more fixes for 1.6
it's back to 1.6.2.18.2 for me.
https://issues.asterisk.org/jira/browse/ASTERISK-18103
Regards
Lee
If it is a regression introduced in 1.6.2.19, then
On Mon, 18 Jul 2011 09:03:52 -0400, John Novack
jnov...@stromberg-carlson.org wrote:
there are other low cost solutions around as well.
the ALIX boards I have seen do not impress me. I think they are somewhat
overpriced. Jut one opinion
Thanks for the feedback. I'll read what HP has to offer.
On 07/18/2011 09:00 AM, Robert Huddleston wrote:
Boy if only it was Enron :)
Baby steps. Success is not built overnight; you have to work your way
up the totem pole of fleecing people. Start small: persistently ask
basic, RTFM-grade newbie questions while assigning yourself pompous,
Alex you are my role model... Next time I'm in Atlanta - let's do lunch!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, July 18, 2011 9:08 AM
To: asterisk-users@lists.digium.com
No. The only database stuff we do is MySQL CDRs
Sent from my Computer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Lee Archer
Sent: Monday, July 18, 2011 9:01 AM
To: Asterisk Users Mailing List -
Hello,
I'd like to run Asterisk on an embedded device, where space is scarce.
It should be able to handle calls from a VoIP provider in SIP, calls
from the PSTN through Dahdi, and voicemail.
If someone's already done this, I'd like to know which
directories/files are required for a basic
So Steve
I looked this, but, i didn't understood the difference between enable T.38
and T.38 Gateway, this site ttp://www.voip-info.org/wiki/view/T.38 talk
--Asterisk *1.6* support G.711 and T.38 FAX origination and termination.
T.38 gateway features are still in development. --
I know that
On 07/18/2011 08:07 AM, Steve Davies wrote:
On 18 July 2011 14:05, Lee Archerlee.arc...@thebigword.com wrote:
Seems to be an already reported problem but since no more fixes for 1.6
it's back to 1.6.2.18.2 for me.
https://issues.asterisk.org/jira/browse/ASTERISK-18103
Regards
Lee
If it
- Original Message -
Hello
I'd like to build a compact, affordable, fanless x86 solution to
handle my home landline.
I know about the following two platforms:
1. www.pcengines.ch/alix.htm
alix1d + case 100€
Does Availability 500 mean that it's just not possible to buy just
I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an
attended transfer. The transfer is going to an outbound number (normally AA on
another IP PBX). the audio on the first transfer is fine. But if the user
requests a transfer from AA to another department, I loose audio
- Original Message -
- Original Message -
On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
--[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine.
Similar if not the same behavior still observed as of 1.8.5.0 with FreePBX.
See https://issues.asterisk.org/jira/browse/ASTERISK-17498
-Vladimir
On 7/18/2011 8:07 AM, Steve Davies wrote:
On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote:
Seems to be an already reported
Hi Kevin, the ticket below was closed as it doesn't happen with 1.8. It
can't be related to my ODBC connections if others are having it. Should
a new ticket be opened?
Regards
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi,
When starting Asterisk (1.8.5.0) I see in messages:
[Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module
'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol:
ast_aji_get_client
[Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be
- Original Message -
From: --[ UxBoD ]-- ux...@splatnix.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 18, 2011 11:42:25 AM
Subject: [asterisk-users] chan_gtalk load error
Hi,
When starting Asterisk (1.8.5.0) I
- Original Message -
- Original Message -
From: --[ UxBoD ]-- ux...@splatnix.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 18, 2011 11:42:25 AM
Subject: [asterisk-users] chan_gtalk load error
Hi,
On Mon, Jul 18, 2011 at 03:20:03PM +0200, Gilles wrote:
Hello,
I'd like to run Asterisk on an embedded device, where space is scarce.
It should be able to handle calls from a VoIP provider in SIP, calls
from the PSTN through Dahdi, and voicemail.
If someone's already done this, I'd like
Hello!
I am wondering if the Libss7 add-on for Asterisk also translates ss7
variables into the dialplans for routing, accounting, etc?
Thanks,
Elliot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Dears;
If I need to login using as agent using the AddQueueMember(team,) then what
to be the second paramter? How to be written?
For example, if the agent id is 8000 then it will be:
AddQueueMember(CustomerSupport,Agent/8000) or something else?
Regards
Bilal
---
you
El 18/07/11 18:03, bilal ghayyad escribió:
Dears;
If I need to login using as agent using the AddQueueMember(team,) then what
to be the second paramter? How to be written?
For example, if the agent id is 8000 then it will be:
AddQueueMember(CustomerSupport,Agent/8000) or something else?
Short answer is: dont use it. For the long answer wait for others to
answer that.
On Mon, Jul 18, 2011 at 7:20 AM, Eduardo Carpes car...@bsd.com.br wrote:
Hello guys
I need some help to do works FAX using SIP, anybody know the secret to
this? Have asterisk 1.6.
Thanks!!
--
Enviado do meu
I resoundingly second that.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jul 18, 2011, at 11:12 PM, C F shma...@gmail.com wrote:
Short answer is: dont use it.
On Mon, Feb 21, 2011 at 1:21 AM, Vladimir Mikhelson v...@mikhelson.comwrote:
William,
It still looks like something is not properly set with your account on
Google Voice. Have you had a chance to follow the recommendations I
gave you earlier in the thread?
If the account is properly set
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