2012/1/12, Kevin P. Fleming kpflem...@digium.com:
On 01/12/2012 06:39 AM, Olivier wrote:
Hi,
I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.
I read
Am 12.01.2012 18:50, schrieb mahesh katta:
I was search for free license but for this Digium require purchase any
Hardware then they can provide Free License.
But I have no Digium Device , I am using Grand stream FXO Gateway and
Asterisk.1.8.XX .
I was connected like
On Fri, Jan 13, 2012 at 1:58 PM, Ruben Rögels
ruben.roeg...@jumping-frog.org wrote:
Am 12.01.2012 18:50, schrieb mahesh katta:
I was search for free license but for this Digium require purchase any
Hardware then they can provide Free License.
But I have no Digium Device , I am using Grand
Hi, we have a queue with some 20 agents. Agents are defined statically
in queues.conf (in an include, actually). One of the agents is showing
up as NOT AVAILABLE in queue show... when his client is disconnected
from Asterisk. However, the moment his system gets connected, his
status in
Hello,
I am using ODBC storage for the message deposited through voicemail()
application. Everything works fine for the audio message if the format given in
voicemail .conf is any audio format . e.g.
in voicemail.conf give the format as
format=gsm
This case works fine.
But if I
Hello,
I have the following in dialplan :
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[CheckOnNet]
include = TrunkAccounts
But when a call for 32380837 enters CheckOnNet, it is not found.
Hi,
I'm looking for a SIP hardphone with 2 network interfaces at 1 Gbps. All the
ones I've seen only have dual 10/100Mbps ethernet ports (eg. Grandstream
products).
Any suggestions?
Thanks,
Vieri
--
_
-- Bandwidth and
The new snom 7 series and maybe the 8 series have Gig ethernet
-Original Message-
From: Vieri rentor...@yahoo.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 13 Jan 2012 04:45:12
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial
On 01/13/2012 05:17 PM, mahesh katta wrote:
On Fri, Jan 13, 2012 at 1:58 PM, Ruben Rögels
ruben.roeg...@jumping-frog.org
mailto:ruben.roeg...@jumping-frog.org wrote:
Am 12.01.2012 18:50, schrieb mahesh katta:
I was search for free license but for this Digium require
purchase
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as I'm sure.
[TrunkAccounts]
exten =
On 1/13/12 2:32 PM, Jonas Kellens wrote:
So the context TrunkAccounts is not included.
Do you know why ?
Does reloading the dialplan (dialplan reload) give any useful output
relating to these two contexts?
--
Andreas Sikkema
--
On 01/13/2012 02:37 PM, Andreas Sikkema wrote:
On 1/13/12 2:32 PM, Jonas Kellens wrote:
So the context TrunkAccounts is not included.
Do you know why ?
Does reloading the dialplan (dialplan reload) give any useful output
relating to these two contexts?
I include this context in 2 other
Jonas Kellens wrote:
Everything works fine when including context 'TrunkAccounts' in
context 'PROVIDERin
dialplan showPROVIDERin
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
On 01/13/2012 02:59 PM, Doug Lytle wrote:
Jonas Kellens wrote:
Everything works fine when including context 'TrunkAccounts' in
context 'PROVIDERin
dialplan showPROVIDERin
Doug
Meaning ?
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_
-- Bandwidth and Colocation
Jonas Kellens wrote:
Meaning ?
Meaning I want to see the dialplan order of that context. I'm guessing
that's your inbound context. With includes that also include sub-contexts.
Usually, there is something ordered differently then expected.
Also, what version of Asterisk?
Doug
--
Ben
On 01/13/2012 03:07 PM, Doug Lytle wrote:
Jonas Kellens wrote:
Meaning ?
Meaning I want to see the dialplan order of that context. I'm
guessing that's your inbound context. With includes that also include
sub-contexts.
Usually, there is something ordered differently then expected.
Jonas Kellens wrote:
Does this mean the Return() comes before Asterisk looks into the
context [TrunkAccounts] ??
No, I believe the includes are read first, but the order in important.
Since you may be matching against another context that may cause
failure. For example, I have the
- Original Message -
From: shalu dhamija shalu.dham...@rancoretech.com
To: asterisk-dev asterisk-...@lists.digium.com, asterisk-users
asterisk-users@lists.digium.com
Sent: Friday, January 13, 2012 3:56:45 AM
Subject: [asterisk-users] odbc storage for video message
Hello,
I am
On 01/13/2012 04:22 PM, Doug Lytle wrote:
Jonas Kellens wrote:
Does this mean the Return() comes before Asterisk looks into the
context [TrunkAccounts] ??
No, I believe the includes are read first, but the order in important.
Since you may be matching against another context that may cause
Good afternoon,
My current Asterisk is a FreeBSD 8.2-STABLE machine with
asterisk18-1.8.7.1 installed from ports collection and . My old asterisk
was a pre-configured system (Brazilian Disc-OS), and on both systems i
have the same problem, sometimes some of my DAHDI channels get stuck,
here
On Fri, Jan 13, 2012 at 5:45 AM, Vieri rentor...@yahoo.com wrote:
Hi,
I'm looking for a SIP hardphone with 2 network interfaces at 1 Gbps. All
the ones I've seen only have dual 10/100Mbps ethernet ports (eg.
Grandstream products).
Any suggestions?
Cisco, Polycom, Yealink, and MANY others
On 01/13/2012 02:12 AM, Olivier wrote:
2012/1/12, Kevin P. Flemingkpflem...@digium.com:
On 01/12/2012 06:39 AM, Olivier wrote:
Hi,
I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech
Jonas Kellens wrote:
there is only one include-statement
Then I don't know. I am still on 1.4.x and my PRI context contains all
that I'm matching against (No sub contexts).
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
Doug Lytle wrote:
Then I don't know. I am still on 1.4.x and my PRI context contains
all that I'm matching against (No sub contexts).
One thing does come to mind; the inbound call is coming into your s
extension and then your doing a gosub, in which case, you might be
matching
Le 13/01/2012 14:32, Jonas Kellens a écrit :
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
222, Arunvihar,Sector-28, Noida 201301
GSM +91.3 45699 | Phone +91.12.0431.0581
Web http://www.buzzworks.com
On Fri, Jan 13, 2012 at 6:27 PM, Steve Underwood
Senhores alguém pode me passar os valores aproximados para realizar essas
configurações,
uma amigo não sabe como e quanto cobrar para fazer as configurações para um
cliente e fica me perguntando, eu cobro por hora, e o senhores ?
O serviços são:
- Configuração do PABX digital
- Configuração de
Hi all,
I've got a queue with two agents and everything works great so far.
However, I would like to use the option 'R', which stops moh and rings
once an agent is ringing (Asterisk Trunk) [1] according to voip-info.org
However, using this option changes nothing. The moh is played until an
agent
Hi all again,
I've got a problem with sporadic one way audio calls, which means
sometimes I can't hear the calling party (call is established, but audio
is missing). Today I received ~90 calls, one of them got this problem.
I've got two networks involved, without NAT:
- 192.168.1.X, in there
Hello,
I think this option work only with Asterisk 1.8.X
On the Asterisk 1.8.X CHANGES files:
* Added 'R' option to app_queue. This option stops moh and indicates
ringing
to the caller when an Agent's phone is ringing. This can be used to
indicate
to the caller that their call is about
Hi,
I think this option work only with Asterisk 1.8.X
Ah yes, I see. Now Asterisk Trunk in the description makes sense...
Thanks,
Georg
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hi!
Maybe I am missing something or am a little blind at the moment, but I
didn't find out how asterisk can place a call on hold when acting as user
agent client to another SIP server.
Scenario:
--
Asterisk registers to another SIP server (provider) as user agent.
An inbound call from
Hi everybody
I have been presenting a periodic problem, do not know if anyone listed has
happened something similar,I'm using the asterisk, asterisk-1.6.2.13, in
different locations works well, but every so often fails, hangs on Asterisk
server or simply asterisk, SIP requirements do not answer,
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