On Thu, Oct 25, 2012 at 11:09:01AM -0700, Matthew Hixson wrote:
- Is the Linksys SPA3102 a good piece of hardware for this type of setup or
is there something cheaper? Perhaps a card that can go right into the Linux
box?
I'm using an OpenVox A400 (with an FXO module), which Asterisk can
On Thu, Oct 25, 2012 at 11:33:06AM -0700, Matthew Hixson wrote:
Is there any reason a regular old voicemodem wouldn't work?
IME the voice quality and reliability are pretty grotty. If you find
one that works, great!
R
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On Tue, May 22, 2012 at 11:32:19PM -0400, ft...@mindspring.com wrote:
The calls are routed just fine, but when a call is answered at one of
the extensions or externally (by a home telephone) the asterisk
extensions continue to ring one more time. Is there a way to have
Asterisk drop an incoming
On Mon, May 07, 2012 at 12:03:17PM +0200, Bart Coninckx wrote:
What about phones like the Unidata WPU-7800 (
http://www.udcsystems.com/product/wpu7800.php) ? Does anyone have
experience with those? Would these also suffer from connection
losses?
I've been using a UTStarcom GF-210 for the last
On Mon, May 07, 2012 at 09:14:36PM +0200, Hans Witvliet wrote:
Hope that these are better that the utstar F1000:
Keep on re-chargibg as battery is empty in no-time, and security is
lousy; just wep, no wpa.
WPA and WPA2. Battery lasts about a day in dual mode, much longer in
2G-only of course.
On Thu, Apr 12, 2012 at 01:14:25PM -0700, motty.cruz wrote:
Can this be acomplish? I hope I explained better.
Yes, no problem.
First, get the two servers talking to each other (I like IAX for this,
but SIP also works). If NAT is a concern, there are various ways round
it (I like VPN tunnels).
As I've occasionally posted here before, I have user terminals which can
accept SIP text messages to an SMS-like interface.
After upgrading to Asterisk 10, I do indeed have external processes
generating these messages. But it's a bit ugly. What I'd _like_ to do is
simply generate a callfile, and
On Tue, Apr 10, 2012 at 11:50:40AM -0500, Danny Nicholas wrote:
This is what core show applications in 10.1.3 shows
SendDTMF: Sends arbitrary DTMF digits
SendFAX: Sends a specified TIFF/F file as a FAX.
SendImage: Sends an image file.
SendText:
On Tue, Nov 15, 2011 at 04:42:05PM +, Tony Mountifield wrote:
But it sounds like it is distro-specific.
No, it's system-specific. Debian for example will assign UIDs out of the
relevant range based on the order in which packages are installed.
Just use the textual UID/GID values, not the
On Tue, Oct 11, 2011 at 02:53:26PM +0100, Jonathan Archer wrote:
How can I get the 5 to stay where it is so that lookups work correctly?
is it part of the outbound CID?
My trunking (prefix 9 to get trunk access from either side of the link)
includes things like:
exten =
On Mon, Sep 12, 2011 at 12:21:06PM -0600, linux guy wrote:
FWIW, I'm OK doing things via the CLI, but sometimes its really nice to have
graphical tools.
To add to what everyone else has said: if you _really_ need to run a
graphical tool on the server, you can always ssh -X into it without
having
On Fri, Aug 12, 2011 at 12:23:22PM -0300, equis software wrote:
shutil.move('/var/tmp/1.call','/var/spool/asterisk/outgoing/1.call')
Are both /var/tmp and /var/spool/asterisk/outgoing on the same
filesystem?
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On Fri, Aug 12, 2011 at 12:27:45PM -0300, equis software wrote:
Yes, same server, same filesystem...
I don't do Python, but a web search for shutil.move suggests that it
doesn't reliably use the rename syscall. Might be worth shelling out
to your system's mv command.
R
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On Fri, Aug 05, 2011 at 10:59:03AM +0200, Jorge Barreiro wrote:
What I try to do is that, when there is an incoming call from the ouside, if
someone answers on a phone, then the PBX won't answer.
I have a couple of VoIP phones fed through Asterisk, as well as analogue
phones linked directly to
On Sun, Jun 19, 2011 at 01:40:31PM +0100, Sagbo Romaric wrote:
No, I can't, because, it's a different NAT. I try to simulate P2P with
asterisk.
What you suggest to me ?
I like VPN tunnels. They give you a flat network topology and decent
security.
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On Fri, Jun 17, 2011 at 05:20:39PM +, salaheddine elharit wrote:
i want to use sip 223 in order to call phone number
Is that meant to be the originator or the destination?
Channel: gets the originator; Extension: gets the destination.
Roger
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On Fri, May 27, 2011 at 10:31:57AM +0200, Mark Scholten wrote:
What could the reason be audio in 1 direction is dropping? (Normally from
the Asterisk server to the mentioned SIP clients.) No clear information is
in the logs (it is like the call ended normally) and not all calls are
having problem
On Tue, May 17, 2011 at 01:30:33PM -0400, Mike wrote:
Is there any softphone or TAPI plug-in that allows one to dial from a web
page? As you may know, Skype has a mechanism that converts phone numbers on
a web page to a click-to-dial application. I'd like to use this but on a
normal softphone
On Mon, May 09, 2011 at 03:00:19PM -0600, John Marvin wrote:
However, I want to record what is said during that time and send it
to a third voicemail box once the caller hangs up without having
pressed 1 or 2.
You could use Monitor to record the whole call, then use an AGI to do
something with it
On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote:
Can anyone suggest which is the best scripting language for Asterisk or any
telecom device?
Depends on the other parameters. Perl is great for rapid development,
but I wouldn't run it per-call on a box taking hundreds of calls
On Mon, Mar 28, 2011 at 05:14:50PM +0530, Raj Mathur wrote:
Is there a better way of handling the post-hangup
processing?
Callfiles?
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On Sat, Feb 26, 2011 at 03:08:02AM -0600, Dan Saul wrote:
I am attempting to create a intercom buzzer system using asterisk as a
back end. Most is figured out except the actual action of buzzing the
door. I need to detect whether a DTMF key was pressed by the the
called party (the resident). Is
On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote:
Still last night there was a call to a customer. Plz help me figure out the
solution for this problem.
Can you be sure that the call _is_ coming through your Asterisk server,
rather than being the result of random scanning for your
The relevant part of my setup is something like:
SIP phones - local server - remote server - SIP-to-PSTN provider
I want _some_ of the SIP phones on the local server to be able to get
access to SIP-to-PSTN, but not all of them. The local-to-remote
connection is IAX2 over VPN.
Do I need to set
On Fri, Feb 11, 2011 at 04:37:49PM -0600, Danny Nicholas wrote:
In 500 words or less (if possible), please explain what is a
legal music-on-hold file?
One source of explicitly royalty-free music is the podcasting community:
http://uhort.no/ and http://www.podsafeaudio.com/ both have
On Tue, Feb 15, 2011 at 08:39:57AM -0600, Danny Nicholas wrote:
Good suggestion, Roger, but this seems like a slippery slope path.
Today's podcaster could be tomorrows ASCAP/BMI member coming back for you?
Doesn't matter if you use music that has been explicitly released as
royalty-free (usually
On Tue, Feb 15, 2011 at 09:01:16AM -0600, Danny Nicholas wrote:
Thanks for the tip - got a Norwegian translator for uhort.no?
Anything wrong with
http://translate.google.com/translate?js=nprev=_thl=enie=UTF-8layout=2eotf=1sl=notl=enu=uhort.no
?
R
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On Mon, Feb 14, 2011 at 04:06:10AM +, Edwin Quijada wrote:
How would be the dialplan for this context from-lan ???
This list is for non-commercial support. If you want someone to do the
work for you, I suggest you go elsewhere and offer money.
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On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote:
I have a webpage with information about a customer so in this page the agent
click a phone number and asterisk do the call and transfer the call to agent
if this call is answered.
Usually it's the other way round: the agent's phone
On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote:
This works for me.! but the agent has to dial the number ?
How could be the context for do this ? U can give an example ?
I'm using this to place calls from local IP-phones over the PSTN. So my
script will generate, say:
Channel:
On Thu, Feb 10, 2011 at 04:58:05PM -0800, John Jolly wrote:
i am trying to configure the meetme conference (asterisk 1.8) to play a *
random* sound file from a specific directory prior to it dropping the caller
into the conference itself.
Absent an Asterisk-specific solution, how about a
I have a mobile phone (UTStarCom GF-210) that uses SIP MESSAGE to send
SMS messages over VoIP. My Asterisk 1.4 installation drops these
messages and returns a failure condition to the phone:
[Feb 9 10:17:22] WARNING[11960]: chan_sip.c:9859 receive_message: Received
message to
On Mon, Jan 24, 2011 at 02:58:45AM -0500, RR wrote:
In the meantime, does anyone have a nice way to update a stable/stock lenny
installation with the updated glibc as well as the latest kernel
At this point the easiest option will be to upgrade to squeeze.
R
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On Wed, Jan 12, 2011 at 10:13:22AM -0600, Kevin P. Fleming wrote:
His point is valid though... A's registration should not have been
overwritten until B *successfully* registered. A failed attempt to
register should have no effect on the existing registration.
Indeed, the avenue for a
On Sun, Jan 02, 2011 at 12:04:07AM +, JP CR wrote:
What I want is when a potential client submits his number... the PBX dials the
number makes an announcement and dials an extension (which is actually a
cellhopne dahdi member) and makes the connection.
You might try something based on
On Mon, Jan 03, 2011 at 02:41:36AM +0400, Thomas Perron wrote:
Cool. So, one Asterisk machine handling up to 100 DID numbers, correct?
As many as you like, modulo memory and CPU requirements.
I assume that the DID mumbers dialed would be the exaxt match needed
to start the respective context.
On Mon, Nov 29, 2010 at 01:36:17PM -0600, Chris Gentle wrote:
This is click-to-call. It can be done with the Asterisk Manager Interface
(AMI). See this site:
Thanks to you and Tilghman for this, though as it turned out it was much
simpler to avoid AMI completely and use the Extension:
On Mon, Dec 20, 2010 at 02:34:23PM +, salaheddine elharit wrote:
the 0,1, and 6 are OFF just the 2,3,4,5 are ON ,
and when i reboot the server i found that the service httpd is off with
command service httpd status and service asterisk status
please advice
This is just one of many problems
On Sun, Dec 19, 2010 at 12:14:11AM -0500, Stephen Reese wrote:
Thanks for the heads up, I have been setting the caller-ID but the
trouble I'm running into is specifying the which number to call out
as. How can an extension specify a different number? See below for my
current extension.conf,
On Fri, Dec 17, 2010 at 04:52:32PM +0100, Gilles wrote:
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.
How would you _expect_ to be able to specify a destination server from a
telephone keypad?
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On Mon, Dec 13, 2010 at 07:28:58PM +0100, Gilles wrote:
This is a newbie question : With a simple Asterisk server on a private
LAN, an FXO port to handle the PSTN, and an ADSL connection to the
Net, ie. with no VOSP in the mix... how should I configure Asterisk so
that SIP clients can dial SIP
On Mon, Dec 06, 2010 at 03:23:42PM +0100, Jonas Kellens wrote:
I'm trying to set up a video call from my Ekiga client to a
Grandstream GXV3140 IP-phone. The call succeeds but there is no
video.
Try restricting video codec to H.261.
R
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The desired result is that user A's phone rings; when he picks it up,
user B is dialled, and user A's channel is connected to that. (This is
to be a back-end for a web-based address book.)
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On Fri, Nov 05, 2010 at 12:49:45PM -0400, John Regal wrote:
Anyway, I think the idea of replicating the function into an extension will
work. Any pointers on the best way to accomplish this? I created a new
extension but am unsure what to do next. I thought about the FollowMe
feature but I would
On Wed, Nov 03, 2010 at 12:05:51PM +, Ronny Adsetts wrote:
What hardware would I need in the Asterisk so I could hook up some analogue
extensions? Am I right in thinking I need something like an FXO/FXS card?
Yes, this ought to work. If you're plugging phones into the Samsung it's
On Tue, Nov 02, 2010 at 04:13:01PM +, Ronny Adsetts wrote:
3. Other ways?
It all rather depends on what your proprietary system has been set up to do.
(If you didn't already have the Samsung box, you wouldn't need to buy one.)
Dedicated telephony hardware tends to be restricted in all
On Tue, Nov 02, 2010 at 05:54:27PM +, Ronny Adsetts wrote:
Would it be possible do you know to use the Samsung handsets with an Asterisk
system? Is it worth even trying to save money here? (I've no idea of the cost
of VoIP handsets for use with Asterisk).
I've never heard of Samsung
On Tue, Nov 02, 2010 at 03:20:48PM -0400, Silver Thorne wrote:
I am not looking for someone to do this for me, I am just not really
sure how to get started. Perhaps some suggested reading, examples,
etc?
The simplest approach would be to skip the answering and just dial
through immediately,
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.
I think what you need to do here is check/set a variable in the astdb.
(If the variable
On Fri, Oct 15, 2010 at 06:22:07PM +0200, Zarko Zivanovic wrote:
We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this
natted network.
The simplest solution will be to stick another Asterisk box inside the
NAT and tunnel IAX or SIP over a VPN.
R
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You have two separate problems here:
(1)
dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 8 bit,
mono 8000 Hz
You should have generated this with 16-bit resolution, like all the
others.
(2)
Not sure about the cents - sure it's coming out as 16-bit? Is the file
in the right
I now have an OpenVox A400P and it is working well. Thanks to Ade
Vickers for the recommendation, which I second.
However, I need to make a slow transition between a conventional
multiple-extension setup and a full VoIP network on these premises. So
at the moment the Asterisk box shares the PSTN
On Sat, Oct 02, 2010 at 06:24:24PM +0200, mancyb...@gmail.com wrote:
for a vicidial server which uses only voip,
which is the minimum telephony card which would provide the required clock
timing source for conferences to work properly ?
Can't speak for vicidial, but MeetMe() works fine for me
On Sat, Oct 02, 2010 at 04:09:33PM -0400, bruce bruce wrote:
Can't I in my ip tables just accept the pap2t.dyndns.org if that is bind to
the PAP2T? do you think the devices comes in with it's external IP rather
than the dyndns domain?
Yes. An IP datagram carries only the source and destination IP
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote:
What should i do?
aptitude install module-assistant
m-a a-i dahdi
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On Wed, Sep 22, 2010 at 01:27:07AM -0400, bruce bruce wrote:
I have setup an OpenVPN tunnel between Server A (running Asterisk) and
Server B suppling it's SIP Phones with DHCP pool of IPs.
Have you considered running Asterisk on Server B as well, and using IAX
to trunk between them? This is
On Wed, Sep 22, 2010 at 12:32:21PM +0200, Jonas Kellens wrote:
[Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full:
File vm-INBOXs does not exist in any format
[Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable
to open vm-INBOXs (format 0x8 (alaw)): No such file or
On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote:
Can we not do pastebin any more?
No, it's just one user with an excessively paranoid and chatty
mailfilter.
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I am working with a simple follow-me-style service: rather than have
something that rings several phones in turn, the user dials a number (in
the present implementation, unique to that user) to register his
presence at a particular extension.
What's the standard way to protect this from
On Mon, Sep 13, 2010 at 11:22:33AM -0400, Bryant Zimmerman wrote:
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
Consider writing a filter for fail2ban [http://www.fail2ban.org/] that
works on the Asterisk logs?
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On Mon, Sep 13, 2010 at 08:15:34PM +0200, Jonas Kellens wrote:
[Sep 13 20:14:59] -- Launched AGI Script
/var/lib/asterisk/agi-bin/cleanpickup.agi
[Sep 13 20:14:59] opruimenpickup.agi: Failed to execute
'/var/lib/asterisk/agi-bin/cleanpickup.agi': Permission denied
So check that
On Fri, Sep 10, 2010 at 03:51:01PM -0500, Hose wrote:
Does anyone have any suggestions as to
how to approach that, or if they have a entirely different way in mind?
AGI script that can look directly at your master list of numbers/routes?
R
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On Tue, Sep 07, 2010 at 10:58:18AM -0700, Dave Platt wrote:
Note that ifconfig will not necessarily show all of your
interfaces (hard- or soft-) - only the active, configured ones.
ifconfig -a would help here. Kernel upgrades often seem to bring in new
default interfaces.
If this turns out to
I have a pair of Asterisk servers which are happily routeing VoIP calls.
I want to hook one of them to the PSTN. Given that I am in the UK, what
is a reasonably easily-available device to provide an FXO interface from
a Linux box, with a minimum of faffing around with drivers? Just one
line is
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