Dear, I have Asterisk 1.8 (installed with Elastix 2.4) and I want to
integrate a Vtiger 6.5 server.
In my PBX I have Asterisk 1.8, Java 1.4 and I have not Java Jetty.
What are the requirements in the Asterisk server in order to install the
VtigerAsteriskConnector package and then integrate the
No, Local/queue/ don't work at all :(
2012/2/6, Danny Nicholas da...@debsinc.com:
Local/queue/?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, February 06
No :(
2012/2/6, Danny Nicholas da...@debsinc.com:
Queue()?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, February 06, 2012 1:40 PM
To: Asterisk Users Mailing
Dear Bakko, I use this lines in order to listen and whisper:
[custom-spy]
; Listen
exten = _*84.,1,Set(SPY=${EXTEN:3})
exten = _*84.,n,NoOp(spy an agent: ${SPY})
exten = _*84.,n,ChanSpy(Agent/${SPY},q)
exten = _*84.,n,Hangup
; Whisper
exten = _*85.,1,Set(SPY=${EXTEN:3})
exten =
Dear, I have to let some agents from a call center to spy/coach just a range
of extensions. They must not spy extensions from boss and some other
important people from my company.
I have in extensions_additional.conf:
[app-chanspy]
include = app-chanspy-custom
exten = 555,1,Macro(user-callerid,)
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days when I call from one extension to another of the
same PBX after I dial the number the rings sound after 20 seconds.
In the CLI log, when I debug the AGI, I see always goes good until
dialparties.agi,
Warrem thanksa lotI'll test next monday and I'll tell you.
Regards
2011/8/5 Warren Selby wcse...@selbytech.com
On Fri, Aug 5, 2011 at 7:53 AM, Alejandro Cabrera Obed
aco1...@gmail.comwrote:
Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and
Celeron), and last days
Dear, I have Asterisk 1.6 with an E1 Digium card with echo
cancellation module. So I need to use just the echo cancellation by
hardware and disable the echo cancellation by software. I use DAHDI
for my telephony hardware.
If the lines involved with echo cancel are:
In /etc/dahdi/system.conf:
- Original Message - From: Alejandro Cabrera Obed
aco1...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, May 06, 2011 23:51
Subject: [asterisk-users] Blacklist with *30
Dear, when I dial *30 in order to get instructions
, it is not denied
by the blacklist.
Why could be the problem the blacklist doesn't work ???
Thanks a lot
2011/5/9 Dovid Bender asteriskus...@dovid.net:
Try the Elastix forums.
- Original Message - From: Alejandro Cabrera Obed
aco1...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial
Dear, when I dial *30 in order to get instructions to blacklist an
extension, Idon't get the menu but I get a new dial tone.
What happen please ??? What can I do to solve this ???
Thanks a lot,
Alejandro
--
_
-- Bandwidth and
Dear, we have the following:
- Asterisk A with E1 to PSTN connection.
- Asterisk B with IAX trunk to Asterisk A
- Outgoing routes between Asterisk A and B
- Asterisk A with an outgoing route to PSTN with 9|. dial rule
How can I reach the PSTN from Asterisk B through Asterisk A ???
Thanks a lot
Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both
boxes. I need to connect both PBXs with E1/R2 and UTP cable.
What are the requirements to deploy the UTP cable ??? Straight-through
or crossover ??? What are the pinouts in both peers ???
Thanks a lot,
Alejandro
--
Dear, I always had one E1 card with one span, so I've never had any
problem in set it up through /etc/dahdi/sustem.conf and
/etc/asterisk/chan_dahdi.conf because I put span=1.
But now I have a PBX with two E1 cards with 4 span (8 span in total).
How do I have to define both card in system.conf
Dear, I have an Asterisk PBX with two E1 cards: Digium TE122 and Sangoma
A101D. Sangoma card has SNMP support but Digium card has not, and also SNMP
does't give me ral time information.
Within CLI Asterisk I execute dahdi show channels but I don't get
information about channels usage.
What is
Dear, I have this scenario:
- PBX Asterisk 1.6.2.10 with private IP 192.168.0.10
- Behind a Cisco ASA firewall that connects to Internet
- SIP trunk to Net2Phone with these parameters (nat=no):
host=200.58.113.60
username=DOLLY
secret=123456
port=5060
type=peer
dtmfmode=rfc2833
disallow=all
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Regards
Alejandro
--
_
-- Bandwidth and
Dear all, I need a call center asterisk's based solution and I see
there are two important solution for 120+ agents:
VicidialNow and ViciBox
Can you tell me the difference between these open source call center
solution please ???
Special thanks
Alejandro
--
Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for
mobile phone calls coming from a GSM Gateway.
All the components are set up in DTMFMODE = RFC2238, and so when the
caller from mobile touches the IP phone LAN extension, the call is
succesfully established. Everything is OK
set up
with DTMFMODE = RFC2238 by now, and I can't understand if you suggest
me I change the DTMFMODE from RFC2238 to INBAND just in the GSM
Gateway or everywhere.
Thanks again.
2010/6/28 Gareth Blades list-aster...@skycomuk.com:
Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.23
be doable for your
outliers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Thursday, June 17, 2010 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
for a call center like ACD, real-time status of E1
and PRI lines and a so more I don't know they are in Asterisk Now or
Vicidialnow.
Regards,
Alejandro
2010/6/23 Carlo Taguinod cvtagui...@gmail.com:
VicidialNOW (http://vicidialnow.org/)
On Wed, Jun 23, 2010 at 2:21 AM, Alejandro Cabrera Obed aco1
want to install dahdi with wget, I prefer to maintain the original
zaptel installation.
Special thanks
Alejandro
2010/6/16 Doug Lytle supp...@drdos.info:
Alejandro Cabrera Obed wrote:
Is it necessary to install or update any Asterisk/Zaptel/Any extra
module or the default installation is good
Dear all, I need to build a PBX based on Asterisk for a call center. I
have worked with raw Asterisk but it's hard to work for big
implementations think.
Also I have worked with Trixbox CE for a small bussines and it was
prette good, but I have not have many features like ACD. I know there
is
it ??? Or do you suggest another test I can implement
???
Thanks again
Alejandro
2010/6/17 Danny Nicholas da...@debsinc.com:
The physical location of the phone (access to towers) can vastly affect
the
quality of DTMF pass...
--
Alejandro Cabrera Obed
aco1...@gmail.com
Dear all, I have a GSM Gateway that let me connect from cell phones to
the Asterisk's IVR I've created. The IVR let dial any extension you
know, so you can dial to the range 1000-1050.
When the cell phones are in the metropolitan area evertythin is
correct, you dial 1000 and you call to 1000
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Thursday, June 17, 2010 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IVR extension dialing error
Dear all, I have a GSM Gateway that let me
transmission quality, transcodings, etc.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-17 11:25 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote:
Danny, so you say it's a problem of the cell phone and not the
Astreisk or GSM Gateway ???
OK, in this case if I call from a fixed phone
Dear all, I have to install an E1 card in my Asterisk 1.4.23 server
and here is my short question:
Is it necessary to install or update any Asterisk/Zaptel/Any extra
module or the default installation is good enough to just plug and run
the E1 card
Thanks a lot
Alejandro
--
Dear all, I've read that Asterisk supports only the G.729 A audio
codec. I have several Grandstream IP phones with G.729 A/B codec
implementation.
Does G.729 A/B mean both version A and version B, or A/B is a new
version different from A and B and it's not supported by Asterisk ???
Thanks a lot
Dear, I have an Asterisk PBX with 3 SIP extensions (1000, 1001 and
1002) and a GSM Gateway with SIP extension . Two cell phones call
to the GSM Gateway number and after that they get a ring tone to dial
to the SIP extensions.
Is it possible to consider the GSM Gateway SIP extension as an
this sounds REAL simple - just dial
1000, 1001 or 1002 from dialplan or do a Goto to the IVR context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro
Cabrera Obed
Sent: Monday, April 26, 2010 3:24 PM
Dear all, I have an Asterisk SIP server in a LAN environment and I want your
opinion in order to decide the use of an audio codec:
What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip
calls ???
Thank you !!!
Alejandro
--
Dear all, if I use the CustomContext module in Asterisk in order to create
new customized contexts for my extensions to managed outbound/inbound calls,
do these custom contexts replace the original context defined in sip.conf,
like context=from-internal ???
In other words, does a custom context
Yes, Custom Context is a module from FreePBX in order to define calling
routes.
Thanks.
2010/3/22 Leif Madsen leif.mad...@asteriskdocs.org
Alejandro Cabrera Obed wrote:
Dear all, if I use the CustomContext module in Asterisk in order to
create new customized contexts for my extensions
Dear all, I have Asterisk managed by a FreePBX web console, and I want to
add an external dial code, in order to dial 9 to get external line/tone for
outgoing calls to the GSM network through my GSM gateway.
Where from Asterisk/FreePBX can I setup this feature ???
Thanks a lot.
Alejandro
--
Dear all, I use Trixbox as my PBX. Until a couple of days I've installed a
GSM Gateway to communicate with our three cellular phones:
15 6422
15 6422
15 6422
The GSM Gateway has just one SIM.
I use the Free PBX web interface in order to set up the route and trunk
parameters:
Trunk:
, I get new tone, dial ANY phone
number and the call is established.
How canI restrict my calls through the GSM Gateway to just our three
cellular numbres cited above ???
Thanks a lot,
Alejandro
--
Alejandro Cabrera Obed
aco1...@gmail.com
www.alejandrocabrera.com.ar
Dear all,
I'm planning to buy some IP phones with GSM audio codec support in order to
use with an Asterisk SIP server I have implemented and nowsuccessfully
running with softphones like Eyebeam and Twinkle.
A vendor offer to me the SNOM 300 IP phone, that support GSM 6.10 audio
codec. I've
Dear all, does anybody know about a complete set of neutral Spanish sounds
to use in my Asterisk voicemail ???
Because when I get a Spanish sounds package, it always is incomplete.
I live in Argentina, so I prefer neutral voices.
Special thanks
Alejandro
Dear all, I have Trixbox 2.6 (Asterisk 1.4) installed in my voip server. I
have the following short questions about the usage of G.729 codec:
1) Does Asterisk have installed the G.729 codec by default ???
2) If I don't want to pay for a codec license, using Asterisk in
pass-through mode for G.729
Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in
voicemail sounds files (I have Spanish sounds).
But now I have a problem because I have to use G.729 mandatory at peers, and
I have GSM in voicemail sound files. I can't let Asterisk do trascoding
because I have no a DSP in the
Just a short question: I will have Asterisk using G.729 codec and connected
to some voip devices such IP phones (GarndStream) and a GSM gateway
(Portech).
Do IP phones and GSM gateway include valid G.729 licenses or do I have to
pay for them ???
Thanks a lot
Alejandro
On Fri, Jun 26, 2009 at 4:21 PM, Kevin P. Fleming kpflem...@digium.comwrote:
Alejandro Cabrera Obed wrote:
Because sounds files in /var/lib/asterisk/sounds are a lot as I see.
If you are using the Spanish sounds distributed by Digium, they are
already available in G.729 format from
Dear all, I've read some news about Sisky
(http://www.yeastar.com/Products/SiSkyEE.asp), a service to
interconnect Skype clients with SIP clients.
Does anybody test Sisky and can tell me about his experience ???
(Sisky runs on Windows because Skype and its API are more stable on this OS).
Dear all, I want to know if anybody has implented an Asterisk server
(1.4 or 1.6) with SRTP and SIP with TLS, in order to encrypt both
signaling and voice packets.
Is it possible ??
And in the affirmative case, does encryption increase the delay and so
the voice quality becomes wrong ???
Thanks
Dear all, I want to know if anybody has implented an Asterisk server
(1.4 or 1.6) with SRTP and SIP with TLS, in order to encrypt both
signaling and voice packets.
Is it possible ??
And in the affirmative case, does encryption increase the delay and so
the voice quality becomes wrong ???
Thanks
mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Alejandro Cabrera Obed
aco1...@gmail.com
www.alejandrocabrera.com.ar
___
-- Bandwidth and Colocation Provided by http://www.api
/ulaw for everything that's local, gsm or g729 only for
remote extensions.
On 2/24/09, Philipp Kempgen philipp.kemp...@amooma.de wrote:
Alejandro Cabrera Obed schrieb:
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones
Yes, there is a WAN among the hardphones and my Asterisk server.
I know the GSM bitrate is about 31 Kbps.
Thanks
On Tue, Feb 24, 2009 at 1:41 PM, Olivier oza-4...@myamail.com wrote:
2009/2/24 Alejandro Cabrera Obed aco1...@gmail.com
Thanks for your comment about codecsI tell you I
calls the sound quality was BAD not poor BAD.
the digium transcoder is GREATE 0 cpu was gone for transcoding.
keep this in mind.
David
2009/2/24 Kristian Kielhofner kristian.kielhof...@gmail.com
On Mon, Feb 23, 2009 at 6:59 PM, Alejandro Cabrera Obed
aco1...@gmail.com wrote:
Dear all, I
Dear all, I have Asterisk 1.4 with SIP. I have a voicemail implemented
with GSM sound files.
The problem is I have IP phones Utopix HyperPhone 202 which support
only G.729a/u and G.723.1 high/low, but not GSM.
If I choose G.729A the pass-throu calls among users are OK, but
Asterisk can't
Dear all, just a short question:
What is the best CPU hardware requirements (CPU, memory, hard drive) to
install Asterisk with SIP/RTP protocol for 100-150 users, and routing
the RTP traffic by itself (no direct RTP traffic client-to-client)
Special thanks
Alejandro
Dear, I'm looking for IP phones (directly connected to the RJ-45 port
from my LAN) that support any level of encryption for use with an
Asterisk 1.4 SIP server we have.
What branch and type can I use
What is the encryption mechanism I can have with this equipments ???
Greetings
Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users
and it works very well only in an intranet environment (no connections
to the PSTN world).
But in the near future, we have to plan a telephone system that works in
the intranet (voip) and also it must be connected to the PSTN
Dear people, does anybody try the ZRTP patch for Asterisk in order to
have ZRTP encrytion among SIP/RTP calls ???
In other words, did anybody succesfully implement ZRTP in Asterisk ???
Any documentation about it ???
Special thanks
Alejandro
___
--
Jeff Peeler wrote:
On Fri, 2008-06-27 at 11:36 -0300, Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP
clients using ZRTP support (with Zfone module in Windows and libzrtp in
Linux).
People say that it's necessary to use an Asterisk patch
Dear all, I have Asterisk 1.4.13 as a SIP server for my company with 100
peers (I mean users) and everything work fine.
I have the following question: what is the maximum number of peers that
I can reach with Asterisk ??? I know Asterisk is not a SIP server
basically like OpenSER, so I'm
Dear all, I have Asterisk 1.4.13 as our SIP server and several SIP
clients using ZRTP support (with Zfone module in Windows and libzrtp in
Linux).
People say that it's necessary to use an Asterisk patch in order tu
support ZRTP encryption.
Is it true ??? Or maybe if I use the last version of
Dear all, I have installed asterisk 1.4.13 and configured all the
/etc/asterisk files very well. Always I enter the CLI (with asterisk
-r) and when I make a change after that I execute module reload
and everything is OK.
But a few days ago, without make any change, I execute module reload
Dear all, I have installed asterisk 1.4.13 and configured all the /etc/asterisk
files very well. Always I enter the CLI (with asterisk -r) and when I make
a change after that I execute module reload and everything is OK.
But a few days ago, without make any change, I execute module reload
Dear all, I have Asterisk 1.4.13 with the default configuration for
Music On Hold. I have this in /etc/asterisk/musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/moh
and in /var/lib/asterisk/moh I have the default wav files:
fpm-calm-river.wav fpm-sunshine.wav
Dear all, I'm using Asterisk 1.4.13 with voicemail feature. When anybody
receive a voice message, he/she receives a mail with the audio attachment.
After that I dial the voicemail number and I hear the envelope message
that is correct (America/Argentina/Buenos_Aires) which is GMT-3, but
when I
Dear all, I have Asterisk 1.4 and I'm using the voicemail feature. I
edit /etc/asterisk/voicemail.conf with envelope=yes and after that I
left a message in a given mailbox near 11:00 AM. When a dial the
voicemail number in order to hear the message, the Astreisk server close
the cal and I get this
Dear all, I have an Asterisk 1.4.13 with SIP protocol and several voip
clients (Twinkle, X-Lite and SJPhone). Every call among voip clients
pass through the Asterisk server, so there isn't any voip packet
client-to-client.
Can Asterisk control the RTP open ports the voip clients use ??? Or the
Dear all, sorry for my OT but I need to know if Avaya voip server uses
SIP or H.323 ???
Anybody can't tell me this...so I'm here for thei reason.
Thanks a lot
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users
Dear all, I have to implement a linux/iptables firewall between my SIP
clients and the Asterisk 1.4.13 SIP server. There is no NAT in my
implementation, so in sip.conf I have canreinvite=no.
I have iptables 1.3.6 version.
Does iptables need any SIP special module or something like this in
order
Anthony Francis wrote:
Pepo wrote:
Hi friends.
How do I can use Asterisk 1.4 with LDAP? I need it because the system must
use
just one password for each user for everything.
A lot of thanks.
What exactly in asterisk would your LDAP be authenticating? Sip
registrations?
Dear all, I'm using Asterisk 1.4 as my SIP server of my LAN users. The
SIP clients are using different operating systems such Debian, Gentoo
and Windows XP so they use different SIP softphones like SJPhone,
Twinkle and X-Lite.
In order to let SIP clients to see the presence status to each other,
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server
connected to Twinkle and X-Lite clients. I have to use the GSM codec for
all of my clients, and it was set up in the sip.conf specifically in
allow=gsm line.
Twinkle has GSM codec built in, but when I open X-Lite audio codecs
SIP wrote:
Alejandro Cabrera Obed wrote:
Dear all, sorry for the Off-Topic but I have an Astreisk 14 voip server
connected to Twinkle and X-Lite clients. I have to use the GSM codec for
all of my clients, and it was set up in the sip.conf specifically in
allow=gsm line.
Twinkle has GSM
Dear all, I have Asterisk 1.4.13 and I need to use encryption among
Asterisk and my SIP users, and with the RTP data interchanged among
users. I prefer the use of ZRTP/SRTP because we use Twinkle and
X-Lite/Zfone as our voip clients and they support these encryption
mechanism.
My question is: do
I have Asterisk 1.2.13 installed as a Debian package and I edit only
sip.conf and extensions.conf in this way:
sip.conf:
[general]
realm=work.com.ar ; Realm for digest
authentication
bindport=5060 ; UDP Port to bind to (SIP standard port
is
Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP
with Linux/Debian Etch???
I'd like to see if my intranet contacts are available, busy,
disconnected
Thanks a lot
Alejandro
___
Sign up now for AstriCon 2007! September
/AVP 34 31
What can be the problem ???
Thanks
Alejandro
--
Ing. Alejandro Cabrera Obed
Interconexion
SINTyS
Sistema de Identificación Nacional Tributario y Social
Consejo Nacional de Coordinación de Políticas Sociales
People, I have an Asterisk 1.2 server and a Jabber server in different
hosts. I need to implement voip+presence+instant messaging knowing that
Asterisk does not support presence+IM.So is it possible to use a
softphone client (Gaim, X-Lite, etc.) to give to my users
voip+presence+IM connecting
Hi people, I have Asterisk 1.2.13/DebianEtch as my VoIP server, using SIP.
I need to use IM (instant messaging) among X-Lite clients, but when I
send a message to any other client I get the error Error: method not
allowed. I read Asterisk does not support instant messaging,
so.What's the best
Dear all, I have an Asterisk server with SIP and IAX softphones clients, and I
need to encrypt the voip calls among them:
*For SIP clients I use Twinkle which implements the ZRTP/SRTP encryption
mechanism client-2-client; I read it's the better security mechanism nowadays
created by Phill
Dear all, I have a Debian/Asterisk server and I connect several
softphones using SIP in a first test and IAX in a second test. They work
OK in both cases; I use Twinkle client for SIP conversations and Kiax
for IAX.
But now I want to have IM also, I mean a voip client with a chat
messenger
People, I've setup Asterisk in a basic mode with SIP protocol. In the
future I wanna connect several offices each one with an own Asterisk
server, using IAX because I read it has no firewalling problems using
just one UDP port for control and data -aming other advantages- . SIP
has NAT problems I
People, I have a Debian box with Asterisk and I've installed the Destar
package in order to get web managing of my voip system.
After I installed Destar, it runs on localhost:8080, but my server
does not have X-Window to access to it so I can engter the web interface..
So how can I change
People, I will install asterisk on my Debian Etch box without a PSTN
interface card. I want to use only softphones for the moment.
My question are:
1) Is it enough to install with apt-get the asterisk 1.2 or do I have
to get asterisk 1.4 manually ???
2) Do I have to configure a dummy PSTN
Tzafrir Cohen wrote:
On Tue, Apr 10, 2007 at 10:58:45AM -0300, Alejandro Cabrera Obed wrote:
People, I will install asterisk on my Debian Etch box without a PSTN
interface card. I want to use only softphones for the moment.
My question are:
1) Is it enough to install with apt-get
server.
Really thanks.
Alejandro
--
Alejandro Cabrera Obed
Interconexion
SINTyS
Sistema de Identificación Nacional Tributario y Social
Consejo Nacional de Coordinación de Políticas Sociales
Presidencia de la Nación
Julio A. Roca
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