s...@sil.at
Sent:Mon 14-11-2011 15:42
Subject:Re: [asterisk-users] unavailable state not reported to Cisco SPA50X
phone
Am 14.11.11 06:54, schrieb Linux:
I tried to understand the rfc4235 which states the following:
However, using this package to model state for non-
session dialog
Hello,
(using trixbox with Asterisk 1.6.0.26)
I am looking for information about how Asterisk should notify the unavailable
(SIP) state of a SIP device.
I found out that the phone (SPA504G with attendant console) sends a SUBSCRIBE
request with an Accept: application/dialog-info+xml.
On Mon, Oct 10, 2011 at 8:08 PM, Andres and...@telesip.net wrote:
I would recommend Acrobits. Not free but only a few bucks. It works fine
with ATT 3G.
This begs the question... which is more expensive (and where)...
making a regular cell call or making a SIP call over 3G ?
--
I'm about to start building my asterisk server and I can't seem to find
anything that discusses the pros and cons of installing the OS (Fedora 15)
as console only or GUI, ie install KDE as well.
So, other than a bit of disk space, is there any reason why I shouldn't
install KDE when I set it up ?
On Mon, Sep 12, 2011 at 12:24 PM, Robert Huddleston
rhuddles...@gmail.comwrote:
I personally would never install a GUI o/s… By doing so you always open
yourself up to more security concerns.. Packages / ports / etc.
** **
Course one might argue – “it’s behind a firewall”….
** **
Great discussion, people.
I'm ordering hardware today.
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I'm looking for an FXO device to connect to a POTS line that communicates
via USB or Ethernet.
Is there such a device ?
Thanks !
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Great discussion, all of it. Thanks, people.
How much power does the home asterisk box need ?
I'm using Asus Eee Box (1012Ps) as Myth front ends in another project.
About $280 with 320 Gb hard drive and 2 GB RAM. Atom 510 processor. Built
in Wifi. Nearly silent. Runs F15 nicely. Would one
I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk
system.
Any ideas ?
Thanks !
LG
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I was thinking of using a PAP2T-NA for the ATA to handle the fax. It
appears to have a large number of fax specific settings. Can anyone comment
on using this device with a fax ?
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In our house, we need wireless. I have a Grandstream already.
I am looking for something with a form factor more conventional than a
cellphone. Maybe that is silly ? I see various unlocked large screen
Android devices for ~$150.
I was hoping to spend on the order of $50 per handset.
I don't
Do any of the DECT systems handle multiple incoming phone lines ?
How do the DECT systems integrate with the voice mail services on an
Asterisk system ?
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I like the idea of running multiple ATAs with a single base or handset on
each line.
Something like the Panasonic KX-TG4111B which sells for about $40 for a
handset and base. PAP2s sell for about $50 or $25 per line. Total cost of
$65 per handset.
Comments on this approach ?
--
On Fri, Aug 26, 2011 at 12:55 PM, Ian Pilcher arequip...@gmail.com wrote:
On 08/26/2011 01:02 PM, linux guy wrote:
I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home
asterisk system.
I don't have any experience with them, but the Siemens Gigaset A580 IP
seems to be about
Lee Wilson wrote:
--- On Wed, 14/1/09, Ex Vito ex.vitor...@gmail.com wrote:
While I don't know the OpenVOX B200P specifics, some
interface cards
need you to change physical jumpers in order to acheive
NT vs TE, mode.
Could that be the case ?
--
exvito
I've just checked the card
Lee Wilson wrote:
Also, I guess at this point it doesn't matter for
L1, but should I be using Point-To-Point or
Point-To-Multipoint?
Thanks
Yes, you would still need to configure mISDN correctly as
well! And
AFAIK you will need to use PTMP, as that is what the router
Erik de Wild: Tripple-o wrote:
What is the most reliable method for Asterisk
to detect the Called ID for incoming calls on
an analog line in the Netherlands?
In Holland you have to pay to receive cid info on the incoming line.
If you don't pay at the moment you can
Drew Gibson wrote:
SNIP
I suspect that this is due to the call
billing structure in Europe. They make the North American telcos look
positively philanthropic.
Yes indeed!
Flat rate calling plans? What are those?
Flat rate Mobile Internet? non-existant!
We pay per minute/SMS/MB on every
Problem:
When I have more than one IAX2 connection (on server zuiderven), I have
problems in receiving calls from IAX peers except for the first in the
list as seen by the iax2 show peers command.
In my tests it showed that by removing one by one the entries from the
iax.conf file in the order as
Have a look at this:
http://bugs.digium.com/view.php?id=7261nbn=6
If you are running on linux (with a recent kernel), add the iptables
firewall rule to drop the bad packets:
# drop Keep Alive packets from Cirpack SIP proxy xs4all
/sbin/iptables -A INPUT -p udp -m udp --dport 5060 -m string
I have been trying for some time now to make the hook flash work on the
FXS port.
I am using Asterisk 1.4.10.1 with zaptel 1.4.4.
When I manually flash the hook I can manage to find the duration to put
a call on hold. However when pushing the flash button it never works. The
phone's flashtime
PROTECTED]
wrote:
On 8/1/07, Linux Lover [EMAIL PROTECTED]
wrote:
This SOHO PBX box won't interop with Asterisk
because it doesn't speak any
of the protocols that Asterisk does. This box
I tend agree with your evaluation. Still, I was
thinking that since all these el-cheapo SOHO
Hello,
I am a small business owner in need for a solution
that automatically answers an incoming call, prompts
the caller via touch-tone menu (press 1 to leave a
message, press 0 to speak to a representative) and
will ring my (real) phone ONLY if requested by caller.
I know that Asterisk is
Yes, you understood correctly. Thank you - and all
others who replied so quickly - for your precise and
guiding answers.
The Digium TDM11B looks looks like the perfect match
for me:
http://www.telephonyware.com/telephonyware/tw00068.html
But one thing that I forgot to mention is that my
:
Linux Lover wrote:
But one thing that I forgot to mention is that my
business is only in its beginning stage and I need
to
be as thrifty as possible. While $216 is a
reasonable
price, I was wondering whether my (currently very
modest) goal can be achieved by spending much less
James, thank you for your educating answer.
--- James FitzGibbon [EMAIL PROTECTED]
wrote:
This SOHO PBX box won't interop with Asterisk
because it doesn't speak any
of the protocols that Asterisk does. This box
appears to be a solid-state
(and I'd assume very feature restricted)
This arrived today: July 16th. (but I also receive older messages;
oldest is from july 4th)
Looking at the headers of the messages it always appears to be the last step
that is delayed. (that is the delivery to my local mail server).
Example below:
Received: from viadoos.rzuiderven.nl ([unix
You are definitely not the only one. For me it is the same. I did sent a
message about this to the list owner.
grz,
Hans Feringa
Hello list,
I am getting the list with days of delay, take for example this message:
Received: from unknown (HELO lists.digium.com) (216.207.245.17) by
I already noticed the hisax problem, so I removed the module from the
modules directory so it cannot be loaded anymore. Are you referring to this
driver in specific, or other misdn specific driver.
BTW it seems that messages from the list have about 2 days delay, that is
why I did not see the
in linux.
Hans,
Have a look at the man page for modprobe.conf, specifically the
install directive. There is an example of how to force the order.
But it is already heavily abused.
You may actually want to load one and not the other, and with that
directive you can't .
One alternative
I experienced the same problem. The only way I could get both
ISDN and analog working was unloading kernel modules for zaptel
and mISDN after boot and then load them in the order:
zaptel first and then mISDN. Still need to find out how to configure
load order in linux.
grz,
Hans Feringa
I am new to Asterisk (1.4.5), and I am trying to get chan_mobile working.
My intention is to use it as a cheap GSM gateway.
In the dialplan I configured that all mobile numbers should go thru the
mobile channel. The current situation is that I can setup the call via the
mobile channel
Hi all,
I am preparing the new asterisk system for 60 concurrent calls with 2 E1.
I have to use server HP DL380 G5.
Anybody get TE212P card work on this server using asterisk?
Thanks,
M
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Hi all,
I got this error message below after running the two
commands below: -
insmod zaptel
insmod wct4xxp t1e1override=0xFF
ztcfg -vvv
The error message is as below: -
Jul 25 13:23:00 test kernel: Zapata Telephony
Interface Registered on major 196
Jul 25 13:23:00 test kernel: Zaptel Version:
Hi all,
I am using CentOS 4.3, asterisk 1.2.9.1 with rx_fax
and tx_fax
I am having problem sending out fax from fax using an
ATA connected to the asterisk.
Below is the message pop up in asterisk -vvvgc: -
-- Executing Set(SIP/60005-1c18,
FAXFILE=/var/spool/asterisk/fax/mydocument.tif) in
I am not using any Zaptel card... I am doing a
back-to-back to Verso C5CM via Internet
--- Avi Miller [EMAIL PROTECTED] wrote:
root linux wrote:
I am having problem sending out fax from fax using
an
ATA connected to the asterisk.
Your system is detecting the fax and trying
Here is my E1 setting in zaptel.conf: -
span=1,1,0,ccs,hdb3
bchan=1-15 # set this to 1-15,17-31 for E1
bchan=17-31 # set this to 1-15,17-31 for E1
dchan=16 # set this to 16 for E1
loadzone = us
defaultzone=us
Here is my E1 setting in zapata.conf: -
switchtype=euroisdn
signalling=pri_net
group=1
Hi all,
I am running asterisk 1.2.9 + digium te110p
Does my setup about support outbound fax?
Regards,
rootlinux
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When I fax, I got the following error msg: -
Jul 4 16:26:14 WARNING[17522] chan_sip.c: Unknown SDP
media type in offer: image 7002 udptl t38
--- root linux [EMAIL PROTECTED] wrote:
Hi all,
I am running asterisk 1.2.9 + digium te110p
Does my setup about support outbound fax?
Regards
I want send from asterisk to fxo line (tdm400p) a number (232) and Flash Key
For example 232R (R in italy is flash key)
I wrote this configuration in extension.conf:
exten = 377,1,Dial(Zap/7/232)
exten = 377,2,Flash()
exten = 377,3,Hangup()
But this config don't work
Help me
HiI am facing very strange problem when i try to use asterisk in media proxy mode by using canreinvite=no i receive no voice at both ends. and when i use canreinvite=yes voice is OK at both endpoints. i tried to use play back application to check if asterisk is communicating well with UA and
Hi
I need send a codenumber + key R (flash) from isdn telephone to a interface
on pstn.
isdn telephone -asterisk - (fxo)-- interface
Help me!!!
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Asterisk-Users mailing list
To UNSUBSCRIBE or
- Original Message -
From: root linux [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 23, 2005 4:48 AM
Subject: Re: [Asterisk-Users] Unable to create
channel of type 'Zap'
My context in zapata.conf
Yep, I am connecting to some other equipemnt...its a
Clarent gateway equipped with a National Microsystems
(Quad Port)
--- El Flynn [EMAIL PROTECTED] wrote:
Hi there,
Are you getting the E1 span in from Telekom, or are
you connecting to some other
equipment?
root linux wrote:
My
providing the
clock? The Asterisk side
or the other side (telco?) If it is the telco the
clock needs to be set to
'0'.
Span=1,0,0,cas,hdb3
-Nate
-Original Message-
From: root linux [mailto:[EMAIL PROTECTED]
Sent: Sunday, August 21, 2005 10:30 PM
To: asterisk-users
') to see
what messages are being passed between your system
and the provider.
Regards,
Derek
- Original Message -
From: root linux [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, August 23, 2005 4:48
[EMAIL PROTECTED] wrote:
On Monday 22 Aug 2005 04:30, root linux wrote:
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
You do not appear to be in the US but Malaysia. Not
sure what
- Original Message -
From: root linux [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 22, 2005 11:30 AM
Subject: [Asterisk-Users] Unable to create channel
of type 'Zap'
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
My zaptel.conf config: -
# Below setting is for E1
span=1,1,0,cas,hdb3
bchan=1-15
bchan=17-31
dchan=16
loadzone = us
defaultzone=us
My zapata.conf config: -
# Below setting is for E1
switchtype = national
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
My extension.conf
http://www.wifi-cell.com/On 7/1/05, Huddleston, Robert [EMAIL PROTECTED]
wrote:Do you know where to get one of these? -Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of chawki hammoud Sent: Thursday, June 30, 2005 4:35 PM To:
Hi
I found the site, I will call tomorrow to find out prices
http://www.wifi-cell.com/
Good luck! to everybody!
fachtopiaOn 7/1/05, Richard Malcolm-Smith [EMAIL PROTECTED] wrote:
If it does materialize, im up for 3 or 4 of them at that price.Huddleston, Robert wrote: Well poo - if I can use
Hello people
This is the site: http://www.wifi-cell.com/
Make a call and find out, I am interest it into hear opinions or Beta Testers
See ya
FachtopiaOn 7/1/05, Cory Andrews [EMAIL PROTECTED] wrote:
Robert - I suspect what they are doing is just trying to build buzzand simply not mentioning
Hi Matt,
Can I know your setting in asterisk to allow
connection from SER?
Below is my configuration in sip.conf : -
; incoming calls from ser
[ser-in]
type=friend
host=1.1.1.2
And, can I have your SER configuration file ser.cfg?
Below is my ser.cfg config: -
if (uri=~1.1.1.4) {
Hi all,
I am working on a similar project for my college like
below: -
SIP Phone -- SER -- Asterisk -- Clarent C5CM
My SIP Phone can registered to SER. A SIP Phone X
(with public IP) can call a SIP Phone Y (with public
IP ).
These two X and Y SIP Phones can call to Asterisk
1000.
Does this
. Also openvpn runs on linux, *bsd, solaris, windows,
and maybe in
other OS.
Miguel
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appreciated
Bye
Fach
On Sat, 20 Nov 2004 00:39:28 -0500, Gregory Junker
[EMAIL PROTECTED] wrote:
Linux 2.6 kernel includes IPSec directly, and ipsec-tools can be used to
create a secure point-to-point link. OpenSWAN makes use of the kernel
IPSec in 2.6, and makes it available in 2.2 and 2.4
Hello everybody
A given scenario:
A client does want to have his own VoIP PBX with Asterisk running, but
he ask me. How secure can be the communication among all subscribers?
If there're sniffers on the middle or any other listening device on a
given netowork.
The client is not fictitial, but
Hello everybody
I wonder if I can get any good references of web-based management
interfaces of Asterisk?
I would rather prefer more on detailed functionalities that a great
look, I am willling to work on provide a UI support in case there's
one in progress
Also, is there's some for a price I
Hi
I am getting more hands on about Asterisk issues but I got a question to ask.
What is the common factor, to put all configurations bind to MySQL or
have them as they are originally on text configuration files.
Maybe this questions can be out of focus, but it will clear up some
ideas in the
Maybe is not right to ask, but those changes are GPL to the community?
If yes, any README to integrate them, or detailed explanation
If not, no problem, is nice to know that those changes exist
Regards
John Fach
On Sun, 19 Sep 2004 18:06:56 -0500, Matthew Boehm [EMAIL PROTECTED] wrote:
hi...
what features are you looking for to be included in such application?
SJ
Do you have any? Is web based?
Please send me the one you have with features, price and demo (if any)
on private email
Bye
John Fach
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Hello everybody
How can I get * timming in a server without usb and not rtc ?
Note: the server don't have USB port too
The server is remote and plans to be and serve as hosted *
for services
Regards
JMFA
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Daniel
Well, that seems like a good joke, but forget it!. Honestly, and don't
take me wrong. I am cuban and it is one of risky and hard to maintain
deals in the whole worldwide telco business.
Is better to get termination in the Polinesian Island than Cuba
But, if you get it let me know I will
, Asterisk - linux - JVB wrote:
Yes I linked all the mp3 and mpg extensions with the mpg123 program
(/usr/local/bin) ... but still not able to get the music on hold playing
Getting curious now what I am doing wrong ...
Andrew Joakimsen wrote:
Did you remove the symlink
Conference call problem - do not have any special hardware added to the
system yet.
Did the following:
* Uncommented the ztdummy.c in the Makefile (/zaptel) - recompiled all
* Extensions.conf
exten = 2675,1,meetme,2675
* meetme.conf
conf = 2675
When I dial 2675 I get the
-2-know?
(I am not a linux expert but is the module loaded automatically at the
next startup?)
[[EMAIL PROTECTED] root]# depmod -a
[EMAIL PROTECTED] root]# modprobe ztdummy
[EMAIL PROTECTED] root]# lsmod
Module Size Used by Not tainted
ztdummy 2548 0 (unused)
zaptel 180032 0 [ztdummy]
ppp_generic
Does anybody know why I can NOT hear the MusicOnHold - using SJphone on
another PC in our network (normal playback is not a problem) .
See the * output and the line configured in extension.conf below (also
mp3player does not function)
Any suggestions?
Asterisk output:
*CLI -- Executing
to schedule in the past?!?!
Any suggestions are welcome
Andrew Joakimsen wrote:
http://www.marko.net/asterisk/archives/0207/0097.html
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Asterisk - linux -
JVB
Sent: Tuesday
:[EMAIL PROTECTED]] On Behalf Of Asterisk - linux -
JVB
Sent: Tuesday, August 19, 2003 3:13 PM
To: [EMAIL PROTECTED]
Subject: Re:
[Asterisk-Users] MusicOnHold
Andrew, thanks I
already have got mpg123 installed and working. However still got the
MOH stuff up and running.
Got a feeling it has
playbacks or on voice calls too ?
Make sure the video card is in a text mode. What spec is the machine ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk -
linux - JVB
Sent: Thursday, August 14, 2003 2:22 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk
Installed Asterisk on Redhat 9.0 - and not channeled to PSTN/PLMN
networks (no XP100 or special hardware) yet
When I use * with a softphone (SIP) - Asterisk answers the call but
voicemail or other playbacks are STOTTERING for the first 30 secs
(approx.)It happens more often when I start
- 64 MB
Jeroen
Richard Alexander wrote:
Only audio playbacks or on voice calls too ?
Make sure the video card is in a text mode. What spec is the machine ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk -
linux - JVB
Sent: Thursday, August 14
All,
Could one use the standard analoge modem to test Asterisk functionality.
I mean just the phone and line jack OR do I need specific hardware?
Thanks in advance
Jeroen
-
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