Displaying 20 results from an estimated 200 matches similar to: "(Fwd) Re: Configuring Softphone"
2010 Dec 08
3
Configuring Softphone
Hi,
I'm trying to get a softphone configured. In Sip.conf [general] I found an example
that said I need:
nat=yes
localnet=192.168.xxx.xxx
Is localnet supposed to be a LAN IP or a WAN IP?
Thank you,
Gary
2008 Jan 11
5
Congestion/Forbidden issue with new carrier
Hi everyone,
having a issue with asterisk and my new Voip providers service.
Iv set up many asterisk systems before but never seen this and have
tried to fix this with no luck..
I have used this exact same sort of setup for 5 other providers and
never had this issue, If i replace the trunk login details with my works
voip account and set it to IAX then it works perfect, Just not the new
2007 Sep 10
2
Failover SIP logic
I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status
Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy)
extensions.conf:
[globals]
trunk_1 => SIP/trunk1
2009 Feb 12
1
Problem with parking
Hi,
I'm having problem with call parking.
When I park call, either via transfer to xten or park digit sequence from
features.conf, I hear the parking lot number read to me and the user gets
transferred.
However, MOH stops for the caller the moment user is transferred.
The user can be retrieved by dialing the parked extension and voice resumes.
If the parked user hangs up, the channel state
2007 Sep 13
1
Problems with two trunks
Hi,
I am attempting to setup an asterisk server, current specs:
CentOS release 5 (Final)
Asterisk 1.4.11
Asterisk-gui checked out from SVN last week
I started with a fairly basic setup involving one VOIP provider who
provided one dial in number, and a couple of handsets. Config files are
below. It was pretty much totally built by Asterisk-gui, except for the
fact I had to add
2011 Jan 18
3
Calling rules
Hello.
I don't know if this is a problem, but I was expecting a different behavior.
Users, have to dial "0" to get an external line, and afterwords the number they want to dial (exe 12345). The thing is:
1-If user dial "012345" there is an error and the call isn't made and the error is "handle_request_invite: Call from 'XXX' to extension '012345'
2009 Feb 26
2
Problems with Outbound Calls
Hi everyone!
I'm quite a newbie at this Asterisk stuff so please bear with me.
We've recently decided to start training in Asterisk via AsteriskNow!
Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
The box we have is paired with a Digium TE110P and we've managed to get
it to the point where incoming calls via a single DID (from NTT Japan)
can be received and answered
2006 Dec 08
1
cal recording with email
I'm trying to set on-demand call recording. Here's a snippet of the
pertinent dialplan. The purpose of this is to allow one user in
particular to be able to receive an email recording of the call
everytime he dials *91 + number. The problem is that the email is not
going out or being generated when I use the ${CALLFILENAME} variable.
When I use the actual file name of the gsm recording,
2009 Jul 27
1
disposition "answered" after authenticate??????????
Hi,
I have the following dialplan.
Problem is, if the user authenticates, * starts counting as billable
seconds even if i hangup the phone before the called party answers..And
also
as disposition.. it accepts all calls authenticated as 'answered'
If i commentout the authentication line everything works as it should be.
How should i use authentication that, it should accept it as aswered by
2007 Sep 13
2
FW: Problems with two trunks
Update on this:
I found that by changing insecure = very to insecure = invite, adding
the second trunk no longer stopped calls working.
I've read the documentation on this switch and still don't see how it
applies/is meant to get used.
Anyway, with this change in place, the following may help:
asterisk*CLI> sip show registry
Host Username
2008 Oct 10
2
Configuring Bandwidth.com SIP trunks to prevent one-way audio
Hello,
We have 2 SIP trunks from Bandwidth.com and if both are in use and someone
tries to dial out, they cause another call to get one-way audio (the caller
hears us, we cannot hear them). This happens 100% of the time and
Bandwidth.com doesn't offer any support. I don't see any setting that tells
Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm
2006 Nov 20
4
Auto recording calls?
Howdy, folks.
I'm having a problem finding a way to auto-record calls (both incoming
and outgoing). I know how to make it so either party can initiate
recording, but I want it done as soon as both ends are connected (or
prior to that if that's what it takes). It's probably right in front
of me and I'm just missing it. Any help would be much appreciated.
Thanks,
Jay
2004 Mar 02
1
someone please unsubscribe this person from freebsd-security?
Forwarded message:
> From R.v.Gogh@kappe-int.com Wed Mar 3 07:54:28 2004
> Message-ID: <0FDD52D38220D611B7CC0004763B37441B2572@HNTS-04>
> From: "Gogh, Ruben van" <R.v.Gogh@kappe-int.com>
> To: Darren Reed <avalon@caligula.anu.edu.au>
> Subject: RE: IPFilter and FreeBSD (was Re: mbuf vulnerability)
> Date: Tue, 2 Mar 2004 21:54:23 +0100
>
2011 Sep 26
0
Farmhouse in Provence
The Building Painting (http://www.micaroo.com/home-and-garden/wall-art/building-paintings.html) Farmhouse in Provence also known as Entrance Gate to a Farm with Haystacks was made in 1888 by Vincent van Gogh in Arles in Provence at the height of his career. Oil Paintings for sale (http://www.micaroo.com/home-and-garden/wall-art.html) Partially due to having been inspired by painter Adolphe
2010 Nov 22
5
Someone has hacked into our system
Someone has hacked into our system and is making calls overseas.
How can I:
1. Find out the where the calls are originating from?
2. Block all calls that are not authorized?
Our system is in the USA.
Only calls from inside our LAN are allowed.
Thank you,
Gary Kuznitz
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2010 Jun 06
0
Strange problem with zap channel.
I am trying to help a guy out with his Atcom IP04. He has set it up like this.
He has a handful of IP phones all connecting via SIP. He has two phone
lines connected to the FXO ports one from telecom, another from
vodaphone. He has set up the dialplan so that one of the trunks fails
over to the other trunk. Everything seems to be working OK except for
outgoing calls. He can call from
2007 Jul 12
0
No subject
Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D =
31999)
-- Remote UNIX connection
Verbosity is at least 8
-- Executing [00425298582 at numberplan-custom-1:1]
Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new
stack
-- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20",
"SIP/trunk_3/0425298582")
2005 Mar 21
2
Best Wireless configuration
Hi,
I wonder if anyone has any suggestions on how to setup a network to run VPN
over wireless.
I currently have:
Wireless Laptop ----> Router with VPN pass through ---> DSL modem.
There is also a wired desktop running Win '98se connected to the router.
The Wireless Laptop is running XP Pro sp1.
I am open for suggestions on how to run VPN over the wireless. I just want to
protect
2007 May 13
0
Asterisknow b5 - trouble registering at voip provider
Hi, there.
I have asterisknow beta 5 with the following data:
Ip 192.168.0.60
mask 255.255.255.0
gw 192.168.0.1
the router (a linksys) has port forwarded the port udp 5060 and from
16384 to 16482 udp-tcp from the internet to the asterisk machine.
the only protocol allowed is g729. Which work fine for the ip phones I
already have setup in the LAN.
My problem is trying to register to a voip
2004 Jan 23
1
ipfw + named problem
Ok, I am really stumped on this one. I setup ipfw with all my rules.
Everything works great except for dns. If I do nslookup I get
-su-2.05b# nslookup yahoo.com
Server: localhost.webspacesolutions.com
Address: 127.0.0.1
*** localhost.webspacesolutions.com can't find yahoo.com: Non-existent
host/domain
This is what I have in my ipfw.rules
add 00310 allow tcp from any to any 53 out via