Displaying 20 results from an estimated 1000 matches similar to: "dialplan contexts syntax and terminology"
2015 Feb 22
0
dialplan contexts syntax and terminology
READ READ READ ....
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html
Regards,
Mitul Limbani,
Business Head,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mitul at enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
On Sun, Feb 22,
2009 Aug 04
0
SIP server behind NAT
Hello.
I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage
to make outbound calls, but the communication drops off after 30 seconds
or so.
I'd really appreciate having some assistance from the mailing list on
this issue.
So, I'm having an Asterisk server behind a firewall and Zoiper
softphones on SIP connecting to Asterisk on the same local area network.
The
2015 Feb 20
2
sipsak 200 for a user, but 404 for a different user...why?
What's the difference between user "123" and "devries"? Based on the
output here, they seem the same..?
tleilax*CLI>
tleilax*CLI> sip show users
Username Secret Accountcode
Def.Context ACL Forcerport
201 password 201
default No Yes
123
2015 Feb 20
2
sipsak 200 for a user, but 404 for a different user...why?
On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote:
> A "sip set debug on" will give you more info on why you are getting the
> 404. It probably has to do something with your context/dialplan.
on tleilax:
tleilax*CLI>
tleilax*CLI> sip set debug on
SIP Debugging enabled
tleilax*CLI>
on doge:
thufir at doge:~$
thufir at doge:~$ sudo sipsak -vv -s sip:devries at
2015 Feb 16
3
LAN sip-to-sip
I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a
starfish on it. In some ways, astonishing that it's not really that
definitive, it's more general -- and it only clocks in at one ream of
paper!
In any event, I'm having some port problems on my home network:
http://security.stackexchange.com/questions/81752/
I need to open ports for
2015 Feb 20
2
sipsak 200 for a user, but 404 for a different user...why?
On Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote:
> This is showing nothing so I don't think your test message even made it
> here. I think it looped in the 'doge' server.
I was wondering the same thing :)
in tleilax, I looked in /var/log/asterisk/messages and see:
[Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19]
<--- SIP read from UDP:192.168.1.3:38154
2015 Feb 18
0
ports, routers and firewalls
I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not
even a call. Ring? Beep? Ping? Some sort of "hello world" connection.
192.168.1.1 netgear router
192.168.1.2 asterisk (vicidial)
192.168.1.3 ubuntu client
192.168.1.4 mac OSX client (not shown)
Do I have a firewall problem which would impact a soft phone from
establishing a connection?
2006 Apr 19
0
Re: new_callback_call and conf disconnect
We are using G711 for phones to talk to Asterisk and G729 licenses at
asterisk to talk to ITSP
Could you please suggest transcoder to use from G711 and G729 and which is
comptible with Asterisk. We will like to avoid using TDM if possible
Also i remember that initially we didn't have G729 and were using only 711
for with vicidial but then also we had same problems. at that time it was
only 2
2008 Dec 23
1
second trunk in extensions.conf
I have a TE210P digium card that has 2 E1/T1 ports.
the code in my extensions.conf file for span 1 is :
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1 ; Trunk interface
TRUNKX=Zap/g2 ; 2nd trunk interface
...
...
; dial a long distance outbound number to SPAIN
; This
2015 Feb 16
1
SIP show peers: UNREACHABLE
I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk
the definitive guide", 4th ed. While I don't have the page handy, I was
reading the suggestion to try SIP to SIP before proceeding to outside
connectivity. I'm aware that SIP trunking is a construct, but am,
obviously, learning the system.
What I'd like to do is from the CLI "ping"
2007 Apr 13
2
FreePBX - Vicidial Integration
Hi all,
I am trying to install Vicidial in an existent FreePBX installation
(I'm using Xorcom packages for Debian Etch), but I didn't find any
documentation, I found only this guide [0], but is for trixbox only,
do you think it will work on FreePBX on Etch?
[0] http://iptn.org/vicidial/index.html
Regards,
Diego Quintana Cruz
2005 Jun 07
2
Gnudialer
Hi there!
Is there anyone out there using gnudialer? I tried vicidial but couldn't get
it to work (does vicidial support SIP trunks anyways?). Gnudialer seems to
be simpler, though their web interface needs a little work (version
2.0seems like a step in the right direction but it isn't out yet). How
do you
register an agent? The documentation is lacking...
Jesus Mogollon
2009 Apr 03
2
New ViciDial Call Center Suite Release: 2.0.5
Hello,
We've released another update to our VICIDIAL/astGUIclient call center
suite: 2.0.5
http://astguiclient.sf.net/
The call center suite client applications run on most modern web
browsers on almost any GUI-capable operating system, and it includes
the VICIDIAL call center suite.
This package is free and AGPLv2.
This package is geared towards Asterisk installations with SIP,IAX or
Zap
2004 Sep 24
2
VICIDIAL and IAX
Hello everybody,
I would like to know if there is a support of IAX in vicidial.
I want to make predictive dialing use vicidial using IAX soft phones.
Thanks in advance
Lamine
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2009 Aug 07
3
Going to VM after 180 seconds in queue
Hello all,
This is a VICIDial server and I am looking to send calls to VM box
2100 after 3 minutes of sitting in the queue(via the VICIDial AGI).
This would be inserted between exten => s,8,Background(open) and exten
=> s,9,AGI.
2010 Oct 27
1
Extension notation in default ViciDial installation
Hello List,
A few days ago I installed ViciDial on a server, and while looking to
the default 'extensions.conf' file, I saw this line:
exten => _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT)
Can someone point me out to the Asterisk documentation part where
explains how to use server IP's as extension number?
I could not see it in the ATFOT2 book, and I would
2007 Dec 03
1
New VICIDIAL astGUIclient Release: 2.0.4
Hello,
We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the VICIDIAL call center
suite and the astGUIclient client-side web app which extends your
phone's functionality.
This package is free and GPL.
(the suite is not an
2007 Jan 26
2
Centos 4.4 perl modules. where are they?
I need to install VICIDIAL (cause is a free outbound call solution)
and vicidial wants the following perl modules:
MD5
Digest::MD5
Digest::SHA1
readline
Bundle::CPAN
DBI (found the rpm perl-DBI)
DBD::mysql (found the RPM perl-DBD-Mysql)
Net::Telnet
Time::HiRes
Net::Server
OLE::Storage_Lite
Spreadsheet::ParseExcel
I do not want to taint the installation if an RPM is available on the
DVD or in
2004 Apr 27
3
New ASTGUICLIENT released: 1.0.1
Hello,
We've released another update to our Asterisk GUI Client suite:
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX and includes a dialer
(the suite is not an asterisk configuration tool)
In addition to the usual bug fixes, this is mostly an update for the
VICIDIAL dialer application.
2006 Dec 22
1
New astGUIclient VICIDIAL Release: 2.0.2
Hello,
We've released another update to our astGUIclient suite: 2.0.2
http://astguiclient.sf.net/
The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's functionality and the
VICIDIAL call center suite.
This package is free and GPL.
(the suite is not an asterisk