Displaying 20 results from an estimated 1000 matches similar to: "Integration PBX"
2005 Feb 15
1
Integration Panasonic PBX
Hi,
I was woredering if you could help me to put into practice this solution.
The idea: Create a IVR-Voicemail
The scene:
PSTN------/6------PBX--------/12--------- Internos
|
/4 ports
|
IVR-Voicemail
The Operation:
1)Where a call enters from the PSTN, the PBX flashes and
2005 Jan 31
3
cisco 7960 image
I don't have a CCO account with cisco and I want to use asterisk for my
cisco 7960 phones
But I could not get myself a firmware image for it.
If anyone can share a copy of it, would be most appreciated
Or if you could point me to where I can download it
Thank you
,jm
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2014 Oct 01
1
JABBER_STATUS CODE 7
Hi all,I hope to find a solution with the help of the list, I'm trying
to get the status of my extensions with ejabberd , the idea is to
visualize my users ejabberd incoming calls or missed.
I'm testing with my operator extension with this code but only get the
missed call notification does not show me where the call is coming.
my piece of code
[operadora]
exten =>
2005 Aug 24
0
Asterisk, Maximiliano J. Goldsmid has invited you to try Google Talk.
I've been using Google Talk and thought you might like to try it out.
We can use it to call each other for free over the internet. Here's an
invitation to download Google Talk. Give it a try!
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talk to each other for free over your
2006 Apr 12
1
Recording queue transfers
Regarding this article (1) I have one question to make. What can I do to
record the call if the agent makes a transfer using the "flash" button
instead of "transfer button" or using blindxfer or atxfer defined in
features. conf
If the agent makes the transfer with "flash", the comunication between the
person who is calling and is already in the queue and the target
2005 Mar 25
0
Dial command problem(VOIP+*+TDM400P+Legacy PBX)
Hello,
I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings,
PSTN <-- PanasonicPBX--TDM400P(FXO)--AsteriskPC --> Internet
* is for AA / Voicemail and VOIP in/out
Currently the AA / Voicemail function for incoming PSTN calls are working well.
My problem is for the incoming VOIP call. It can ring my internal extensions and talk without problem.
But
2015 Jan 21
0
asterisk-users Digest, Vol 126, Issue 18 mtr
You could use MTR command.
Its a trace route improved.
Marlon Araujo
> On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote:
>
> Send asterisk-users mailing list submissions to
> asterisk-users at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or,
2012 Jan 13
0
Preços por serviços e equipamento
Senhores algu?m pode me passar os valores aproximados para realizar essas
configura??es,
uma amigo n?o sabe como e quanto cobrar para fazer as configura??es para um
cliente e fica me perguntando, eu cobro por hora, e o senhores ?
O servi?os s?o:
- Configura??o do PABX digital
- Configura??o de URA
- Configura??o de redirecionamento para celular por operadora
ATT
--
Cl?udio Duarte
(61)
2015 Feb 26
0
situation with ivr and four-channel gateway
I'd recommend using DEVICE_STATE
On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not
'NOT_INUSE' then dial it, Otherwise dial SIP/102
exten =>
101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40))
same => n,Dial(SIP/102,40,t)
same => n,Hangup()
On Wed, Feb 25, 2015 at 2:08 PM, ricky gutierrez
2015 Feb 25
5
situation with ivr and four-channel gateway
Hi list, I need your help ,I have an incoming call x the ivr and the
operator takes the call. ext "101" , If a second call reenters and the
operator is talking, I want to send to the extension 102 I use the
Variable DIALSTATUS , but not working
check IVR
[IVRINMA]
exten => s,1,Wait(1)
exten => s,n,Set(CHANNEL(language)=es)
same=> n,Set(TIMEOUT(digit)=4)
same=>
2003 Sep 20
0
Asterisk with Samsung SKP 816H PBX !
Hi,
Having Asterisk-0.4.0 with 2FXO port and Samsung SKP 816H PBX in 2 offices.
I am able to make call between two offices. But the problem is that call
dosen't hangup.
Office A [Asterisk+2FXO+SamsungPBX] <------------- I A X ------------>
Office B [Asterisk+2FXO+SamsungPBX]
Configuration files are given here..............
------------------------
zapata.conf
2019 May 06
2
disable libvirt-nwfilter
Hi,
i want to disable the nwfilter functionality of libvirt.
It's surely nice for some people, nevertheless i don't want libvirt to
alter any netfilter rules, neither i want the according functionality
even available.
I know about nwfilter-undefine, but what i'm looking for is an option to
globally disable this functionality at all. Some config flag or similar.
How can i achieve
2014 Nov 24
2
Re: Libvirt Live Migration
I will try a newer version.
On 24 November 2014 at 11:30, Michal Privoznik <mprivozn@redhat.com> wrote:
> On 24.11.2014 11:23, Dhia Abbassi wrote:
>
>> Thanks for your answer,
>>
>> 1) In this case i'm not using shared storage, the migration is happening
>> between two non-shared storage with full disk copy
>> 2) I already created the file with same
2006 Oct 12
1
routing between two isp
I have three ethernet card and two of them connected to diffrent ISP and
has diffrent gateways. And one ethernet card is connected to local lan. I
have installed iptables and configured the firwall to work as nat gateway
for the users.
Now I want to route only mail traffic to the perticular ISP and which I am
not able to do it. I have done iproute configuration what ever lartc said.
But still I
2004 Apr 29
1
line 521 command not found
Hi
I install shorewall and every seed ok but when I start firwall I got the =
followings
..........
/sbin/shorewall: line 521: --: command not found
..........
..........
Regards,
John Fong
2004 Jun 25
1
SIP extension outside of IP tables firewall
I have an Asterisk PBX on the private lan, which is protected
from the public Internet with a Linux iptables machine. The
firwall is it's own seperate box running NAT with SPI.
I want to drop a SIP phone at my brothers house, and have it be an
extension off my Asterisk box. I've been looking around at some FAQ
info on forwarding ports, and also looked at siproxd.
Anyway, I'm
2003 Mar 04
2
port usage
Our Client software used samba protocol,If there is no firewall case , this
protocol work properly from remote site.All of firwall configuration open (
this mean is all of port open ).
Samba work very well, but if some constrain put in, it doesn't work.
1)I would like to know about port usage of samba protocol.
I think snmpd, nmpd daemon used some port number. i can see /etc/service
file.
2007 May 24
4
design patterns
Hi everyone
I''m Maximiliano Guzenski from Brazil and I am programing on ruby on rails
only a couple of months.
I read a lot of books and articles about it but I could not clarify some
doubts about design patterns:
1) Is it really good put some sql command on controller? because all
articles that I see use sql into controllers, like:
# my controller
@categories = Category.find
2007 Jul 03
1
res_config_mysql.c: MySQL RealTime: Failed to connect database server ..
Hi, I don't explain very well what my problem, but i can't make calls.
i analise my log full and i found two errors
Jul 3 19:02:08 ERROR[4670] res_config_mysql.c: MySQL RealTime: Failed
to connect database server on (err 2002). Check debug for more info.
Jul 3 19:02:08 VERBOSE[4670] logger.c: -- Added extension
'exit-FAILED' priority 1 to macro-vm
Jul 3 19:02:08
2014 Jul 04
1
Disable user login for a few seconds ?
Hi all,
I need to move e-mail files of my users from one file system to another.
I use a MySQL backend on Dovecot 2.1.
How can I do to disable temporary an account while moving files ?
I have an "active" bit in my DB, but if I set it to 0, my MX servers are
bouncing mails.
How can I do to disable temporary an account ? In fact, I need dovecot
not to do any change on files...
I