On Sat, 26 Mar 2011 10:50:19 +0100, wrote:
>I am looking for a way to check the status of a cell phone. Found one way that
>worked for me and would like to have some feedback or suggestion of
>improvments.
I'd like to check I understood: Your Asterisk server is connected to a
landline and can c
On Sat, 26 Mar 2011 14:58:30 +0100, wrote:
>Celluar Network - E1 - Avaya - OOH323 - Asterisk
Thanks for the tip.
So here's how it works:
1. The web app calls a script that uses AMI + Originate to send a call
to the Avaya PBX
2. Avaya is able to check that a number (cellphone in this case) is
bus
On Mon, 28 Mar 2011 14:12:09 +0200, wrote:
>Its not the Avaya that makes the call back, it is mobile.
I thought the way you handled things, is that Asterisk would call your
cellphone through the Avaya PBX just to check whether the cellphone is
in_use/busy. At what point does the cellphone call Av
On Tue, 29 Mar 2011 07:48:08 +0200, wrote:
>I was a little unclear, it is not the cell phone that does the call-back, it
>is the cell-phone-network.
Makes more sense :-) Thank you.
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On Mon, 28 Mar 2011 08:20:23 -0400, vip killa
wrote:
>Is anyone using asterisk with fail2ban?
Sorry for hi-jacking the thread, but I was wondering if there were a
lighter alternative that I could run on appliances?
Python uses too much RAM, but I need to find a way to ban hackers from
trying to
On Tue, 29 Mar 2011 07:31:18 -0500 (CDT), Joe Greco
wrote:
>sshguard is *extremely* lightweight compared to most things; it's a very
>efficient compiled C application that doesn't have (m?)any dependencies.
Thanks much for the tip. I'll study how to install/configure iptable
and sshguard.
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On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
wrote:
>First thing I'd do is restrict the ip blocks your sip endpoints can
>register/call from in sip.conf (or your database's table for sip endpoints)
Thanks for the idea, but it's not possible, as the Asterisk must be
accessible for road war
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
wrote:
>Remember guys, there's a LOT of IP blocks out there that are almost
>definitely not going to be somewhere you expect to receive SIP traffic
>from.
I agree. Is there a list I could use to check which blocks have been
allocated to which c
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote:
>On 03-29-2011 19:25, Steve Edwards wrote:
>> Really? How many callers are you expecting from North Korea, Libya, China,
>> Iran, etc?
>after reviewing last week's log i'd say around 25-28k/min :)
So it looks like I should check out sshguard i
On Wed, 30 Mar 2011 01:45:20 +0300, Ioan Indreias
wrote:
>Just to provide an alternative to sshguard: you could use BFD[1]
Thanks Ioan. I'll give it a shot.
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On Wed, 30 Mar 2011 16:54:51 -0500, Darrick Hartman
wrote:
>One of our developers on the AstLinux team worked out a plugin for
>Arno's firewall (iptables based) which performs similar to fail2ban, but
>uses bash. He called it adaptive-ban. You might be able to adapt it
>for your use, but as i
Hello
I'm no expert of iptables, and it seems like it can handle banning
IP's that are trying to register and fail too many times.
I'd like to use this feature instead of having to install a second
tool such as SSHGuard or BFS that parses the logs and reconfigure
iptables on the fly.
Is
On Wed, 6 Apr 2011 09:46:12 +0100 (BST), Gordon Henderson
wrote:
>Have a look at these:
Thanks much Gordon. I'll study the scripts you mentionned. It looks
like iptables is good enough and I won't have to install a second tool
to watch the logs and reconfigure iptables on the fly.
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On Tue, 5 Apr 2011 17:38:15 -0400, Paul Dugas
wrote:
>First, this appears to be working for me though I'm not 100% sure of
>that and cannot guarantee it will for you in any way, shape or form.
>With the lawyering out of the way...
Thanks a lot, Paul.
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On Wed, 27 Apr 2011 11:55:14 +0300, Ashik Ali
wrote:
>The problem here is that as soon as asterisk dialing on fxo lines it
>sets channel status as "answered" although the chennel is getting
>ring back tone from
>other party.
>
>Anyone can suggest me to solve this issue ?
The only solution I know
On Wed, 27 Apr 2011 14:15:10 +0300, Ashik Ali
wrote:
>Anybody can explain me why asterisk is unable to detect ringback tone
>from PSTN telco ? .
I guess it was a lot of work, and nobody bothered adding this to the
Zaptel driver.
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On Tue, 3 May 2011 13:48:34 -0500, Shaun Ruffell
wrote:
>I know this thread is dead but: I do not believe this should go into the DAHDI
>kernel modules.
I agree. It's just too bad Dahdi is unable to report how an outgoing
call is doing: Still ringing, busy, answered.
--
On Wed, 11 May 2011 01:09:16 +0800, Scott Zhang
wrote:
>So does this mean no solution when used ZAP/DAHDI with PSTN line?
>
>If I installed an E1, will that work?
Before getting an E1, maybe ISDN provides call supervision?
--
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Hello
I just read this article about a kid in England who built a box with a
3G SIM card:
www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html
When someone rings your intercom, the box will call your cellphone so
yo
On Tue, 7 Jun 2011 13:06:23 +0100 (BST), Gordon Henderson
wrote:
>Why bother when you can buy off the shelf stuff to do it for you.
The trick is that this connector must work with existing interphones,
such as this one at home:
http://img220.imageshack.us/img220/8334/intercomhome.jpg
So after I
On Mon, 31 May 2010 13:12:25 +0200, lesouvage
wrote:
>If you are interested in really integrating GSM phones into an
>Asterisk based system without any telco involved check the OpenBTS
>project. I have done a research and trial project and this combination
>of open hardware (USRP), the OpenB
On Sun, 30 May 2010 02:45:51 +0300, Tzafrir Cohen
wrote:
>> This is a bug of the netjet module. It should not try to handle those
>> devices. While they use the netjet chipset, they are not the ISDN BRI
>> devices drivven by it.
>
>[Snip details]
>
>> If nobody beats me to it, I'll try submitting
Hello
I just read this article and would like some feedback from
experienced Asterisk users:
===
"Failed open source VoIP deployment leads to hosted VoIP strategy" By
Jessica Scarpati
"When budgets are crimped, open source voice over IP (VoIP) solutions
look attractive -- a l
On Thu, 3 Jun 2010 08:24:11 -0500, "Danny Nicholas"
wrote:
>Txgain/rxgain in dahdi.conf control this - you will have to restart asterisk
>on each change to test the values to set to your liking - my settings are
>rxgain=8.0
>txgain=4.0
Out of curiosity...
I noticed that when playing back a messa
On Fri, 04 Jun 2010 11:20:49 +0200, Gilles
wrote:
>I noticed that when playing back a message I recorded, volume* is
>lower on an Atcom IP01 appliance running Asterisk/Zaptel, than it its
>when using an Asterisk/Dahdi on a PC, where the client is the same
>(XLite on a PC with head
Hello
Out of curiosity, are those weaknesses still there in Asterisk 1.6, or
have they been fixed?
"How does FreeSWITCH compare to Asterisk?"
http://www.freeswitch.org/node/117
Thank you.
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Hello
I'm learning how to work with Asterisk on an embedded system (MMU-less
Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what
people use as scripting language to handle calls through the dialplan
and AGI, considering the hardware limitations?
Ideally, I'd rather use a rich lan
On Mon, 21 Jun 2010 14:06:22 +0100 (BST), Gordon Henderson
wrote:
>You could always type
>
> asterisk blackfin
>
>into google and see what it suggests.
>
>Here, I'll save you the effort:
Thanks but I already know this (uCasterisk is deprecated). And can't
stand Perl ;-)
--
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On Mon, 21 Jun 2010 17:25:09 +0300, Motiejus JakÂtys
wrote:
>If you can install python or PHP in that machine (in means of
>storage), you are free to run it there. 64 RAM is really enough to run
>python, so you have to just try if it suits in the application. If it
>takes too slow to initialize -
On Mon, 21 Jun 2010 16:47:08 -0700, CunningPike
wrote:
>Not in our experience as a 500-phone, 20-site install for a municipal
>government. We are just migrating from our first generation install to
>replacement hardware (to new blades from servers that are now 5 years
>old) and are still committed
On Mon, 21 Jun 2010 16:10:12 +, Edwin Quijada
wrote:
>Uhmmm.. remember for each channel you run perl or php interpreter so with that
>amount of memory maybe this can be a problem.
> For that kind of project I'd use C or java as fastagi protocol
Thanks Edwin. In my case, the hardware will onl
Hello
About every three months, my dad's little Asterisk server that handles
his business phone line with an OpenVox PCI card stops taking calls.
To check if it's the cause, I'd like to run a CRON job every night to
restart Zaptel and Asterisk.
Before I go ahead, I'd like to know if I can just s
On Fri, 25 Jun 2010 09:53:34 +0200, Randy R
wrote:
>IMO, if it's a business phone, you'd do well to just reboot it at 3AM
>once a week or once a month or some interval that you're comfortable
>with. We used to do this for a similar reason.
Right, but he won't remember to do this, and I don't want
On Fri, 25 Jun 2010 11:43:04 +0300, Tzafrir Cohen
wrote:
>That does not really check if that is the problem. Mind giving more
>information as for the nature of the problem?
I don't have more information. Could just be the Zaptel driver working
with the OpenVoice card, in which case unloading/relo
On Fri, 25 Jun 2010 12:09:09 +0200, Randy R
wrote:
>No I meant as a CRON job! No one will be calling at 3AM, there's ample
>time to reboot once a month for example.
Sorry for the misunderstanding. So I can just run "reboot" from a CRON
job then.
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On Fri, 25 Jun 2010 11:19:25 +0100, Gareth Blades
wrote:
>If you are going to reboot the server regularly then make sure and
>system updates are set to not automatically install new kernel versions.
>Otherwise if you get a kernel update and reboot zaptel/dahdi wont load
>until you recompile it.
On Fri, 25 Jun 2010 13:25:18 +0300, Tzafrir Cohen
wrote:
>Asterisk 1.4.x works with Zaptel as well.
>
>But yes, this means an upgrade of Asterisk, and maybe you'd like to
>avoid that.
Yup. I'll just stop/start Zaptel every night and see if that fixes the
problem. Thank you.
--
ng call after X weeks.
>Hey Gilles, any chance of you fixing whatever it is that you are doing
>that causes you to double-post EVERYTHING?
Sorry about that. I think I fixed it. It looks like a wrong setting in
the NTTP reader I'm using to acces
On Fri, 25 Jun 2010 12:32:03 -0400, Barry Miller
wrote:
>Hi Gilles. You appear to be both posting to newsgroup
>gmane.comp.telephony.pbx.asterisk.user AND sending the same message
>directly to the asterisk-users list. This means that we list subscribers
>see two copies of all your m
On Fri, 25 Jun 2010 08:59:32 +0200, Gilles
wrote:
>Before I go ahead, I'd like to know if I can just send the following
>commands, or if there are issues I should know about:
To avoid issues about the host hanging after a reboot due to
upgrades... I think I'll just run a CRON j
Hello
Googling for this type of non-PC hardware returns products that could
be missing in action for years.
www.google.com/search?q=asterisk+appliance
Is there an up-to-date list of Asterisk appliances, ideally broken
down by price (ie. not just entreprise stuff, but also SOHO)?
Thank you.
--
Hello
To run Asterisk on an embedded appliance, ie. where RAM and
non-volatile memory is an issue (respectively 64MB and 256MB), I need
to check how much space voice messages take to save and play back.
The appliance is connected to a landline in Europe (in case that makes
a difference as
On Sat, 26 Jun 2010 21:09:54 +0300, "Eyal Goltzman"
wrote:
>After installing and learning Asterisk I found myself with a need for a
>minimal set of empty configuration files with only the "must have" stuff in
>order to setup a SIP only machine, is there a place to find it?
The experts might chime
On Sat, 26 Jun 2010 13:35:12 -0400, Paul Belanger
wrote:
>Might get better results on asterisk-biz, and posting your budget price range.
I'll check it out. Thanks Paul
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Hello
I'm taking a look at how to write scripts to be called from the
dialplan, and saw pbx_lua mentioned.
I'd like to know more about this feature, such as what the difference
is with just calling the Lua interpreter through AGI (same difference
as between php-cgi and mod_php?), whether it's pro
On Sat, 26 Jun 2010 17:53:27 +0100 (BST), Gordon Henderson
wrote:
>Dial an extension that answers and stores to voicemail, say blah blah into
>it for one minute and check the resulting file size. divide it by 60 and
>you'll get a good estimate of the number of bytes per second of recording
>for
On Wed, 30 Jun 2010 17:01:50 -0700 (PDT), Steve Edwards
wrote:
>I've never used it (I'm a 1.2 Luddite), but I would be very interested in
>anything that looks like a "real" language for writing dialplans.
That's why I'm interested in using Lua to write dialplan scripts,
besides the fact that due
On Thu, 01 Jul 2010 15:22:33 +0500, Faisal Hanif
wrote:
>I am in process of merging all my AGIs+Dialplan to a single LUA
>dialplan. It seems much interesting to me spacial LUA tables which allow
>me to support a complete object like programming. Yet I did not
>completed / tested.
Thanks for th
On Thu, 01 Jul 2010 01:32:08 +0200, Gilles
wrote:
>I'm taking a look at how to write scripts to be called from the
>dialplan, and saw pbx_lua mentioned.
I'm not having much luck adding the pbx_lua module to Asterisk (on a
Ubuntu 10.04) :-/
# apt-get install lua5.1 liblua5.1-0
On Thu, 1 Jul 2010 15:26:27 +0300, Tzafrir Cohen
wrote:
>Re-run ./configure
Ah, hadn't thought of this :-/
>The Debian asterisk package depends on liblua5.1-0-dev and builds
>pbx_lua just fine.
Yes, it did compile after re-running ./configure, make menuconfig,
make.
I'll check how to use exten
Hello
I have a couple of questions about using modules in Asterisk (1.4 or
1.6):
1. I'd like to experiment with extensions.lua: What happens if...
- I leave extensions.conf enabled by not using "noload=pbx_config.so"
in /etc/asterisk/modules.conf? Will the two dialplans get mixed
together, with
Hello
In case Asterisk is used in a private LAN behind a firewall while
allowing remote SIP clients to connect from the Net, we must open
UDP5060 for SIP and a range of UDP ports (as set in rtp.conf) so let
incoming voice packets.
Provided the user doesn't have access to the firewall (eg.
On Mon, 05 Jul 2010 12:45:34 +0200, Gilles
wrote:
>Provided the user doesn't have access to the firewall (eg. corporate
>or hotel), and the firewall doesn't allow dynamic port opening through
>UPnP or NAT-PMP...
For those interested, I was tipped through private e-mail abo
Hello
To use Dahdi + Asterisk with a PCI card with a single FXO port, I
just...
1. compiled and installed Dahdi
2. edited /etc/modprobe.d/dahdi.blacklist.conf to blacklist "netjet"
and unblacklist "wctdm":
==
# cat /etc/modprobe.d/dahdi.blacklist.conf
blacklist wct4xxp
blacklist wct
On Fri, 9 Jul 2010 08:06:04 -0400, Ryan Wagoner
wrote:
>I have around 50 Snom 370s configured this way. They work great for
>remote workers. However the Snom speakerphone is terrible compared to
>Aastra and Polycom. If there is any background noise it will cut in
>and out the other party.
Thanks
On Fri, 09 Jul 2010 13:45:18 -0500, Shaun Ruffell
wrote:
>> # lsmod | grep -i wc
>> wctc4xxp 32414 0
>> dahdi_transcode 5751 1 wctc4xxp
>> wcb4xxp33905 0
>> wcfxo 8968 0
>> wctdm24xxp116684 0
>> wcte11xp 229
On Sat, 3 Jul 2010 13:47:23 -0500, Tilghman Lesher
wrote:
(snip)
Thanks much for the education.
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Hello
I'd like to write a script that would make it easier for people to
call in, listen to the IVR, and make an appointment (eg. "When? ASAP?
A given day?" -> "Morning? Afternon", etc.)
I assume I'm not the first one to try and write this type of IVR, so
would appreciate any feedback on writing
On Thu, 15 Jul 2010 12:39:51 -0500, "Danny Nicholas"
wrote:
>This how I would do it
Thanks a lot Danny. I'll study this and see how it goes.
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On Thu, 15 Jul 2010 12:39:51 -0500, "Danny Nicholas"
wrote:
>This how I would do it
BTW, is it possible to trigger an AGI script right from the first step
and handle the whole IVR logic in an higher-level script language than
what's available in extensions.conf?
--
On Fri, 16 Jul 2010 09:36:04 -0500, "Danny Nicholas"
wrote:
>Also, in my experience, you will live a happier life depending on the
>dialplan to handle DTMF processing than an AGI.
Thanks for the input. Writing logic in extensions.conf is such a pain
that I was looking for a higher-level solution,
Hello
I just read an article on the tiny Ben NanoNote:
http://en.qi-hardware.com/wiki/Ben_NanoNote
As CPU, it uses a "JZ4720 366 MHz MIPS compatible processor from
Ingenic Semiconductor Co", and it runs Linux.
Does someone know if Asterisk has been ported to that platform?
Thank you.
--
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On Tue, 10 Aug 2010 10:45:59 -0600, "Dave" wrote:
>Hu.. $99 each sounds good. Specs are interesting and it'll boot from a
>USB port. So, Asterisk sounds like it'll work.
Thanks guys for the feedback. I'll check it out.
--
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Hello
I'm sure someone has already tried this: I use a couple of electric
heaters to heat my office.
I'd like to somehow connect them to Asterisk so that I could switch
them on remotely by either calling the IVR or sending an e-mail to the
Asterisk host, so that the room is warm when I get to the
On Mon, 18 Oct 2010 17:36:30 +0530, Jigar Joshi
wrote:
>I need a international number all network should be able to connect to it.
>After ringing a ring call should be picked up. and should ask for a code.
>code should come from mysql or any other DB
>depending upon the code it should route the ca
On Mon, 18 Oct 2010 17:54:27 +0530, Jigar Joshi
wrote:
>Gillies, Can't I configure Asterisk for the same on my live IP system. ?
I don't understand what you mean.
To let Asterisk get calls from the phone network (POTS" a.k.a. PSTN),
you either need a phone line + PCI card or ATA to connect the P
On Mon, 18 Oct 2010 19:12:48 +0530, Jigar Joshi
wrote:
>1. An international number , [That you told ,we 'll get it from VIOP
>providers] ,I will work on it
VoIP provider.
>2 Configuration that will stream all call ,[all incoming calls with any
>extension to a application running on machine, with
On Mon, 18 Oct 2010 13:09:50 +0200, Gilles
wrote:
>I'm sure someone has already tried this: I use a couple of electric
>heaters to heat my office.
Thanks everyone for the great feedback. Following Steve Edward's
advice, I won't automate the process and will only switch the h
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen
wrote:
>Are you saying ADSL as in a generic term for broadband router or do
>you really mean that the router also acts as a DSL transceiver?
Sorry about that. Ideally, the unit should be both an ADSL modem +
router, but apparently, most of them are
Hello
For users who 1) don't have a QoS-capable ADSL router and 2) would
like to run Asterisk with a couple of SIP trunks, I was wondering what
hardware is recommend to run any of the main open-source *WRT projects
to which Asterisk has been ported:
(http://en.wikipedia.org/wiki/List_of_wireless_
On Fri, 19 Nov 2010 10:15:40 -0500, jon pounder
wrote:
>What is nice is when the $50 hardware and the $1000 hardware run exactly
>the same software so other than the drivers for the hardware itself,
>everything else behaves the same way and its easy to move around
>configurations to grow. (I am
Hello
Some SOHO prospects only have a cellphone and I was wondering if
someone had investigate running Asterisk on a smartphone, to perform
tasks such as IVR, CID rewriting, voice-mail, notifications through
e-mails, etc.?
Thank you.
--
Thanks everyone for the feedback.
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ast
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays wha
On Mon, 6 Dec 2010 10:15:34 -0600, "Danny Nicholas"
wrote:
>Here how I changed my information calling an xlite client from a polycom
>501.
>Sipuser = xlite
>144 = polycom
>Exten => 145,1,set(CALLERID(num)=5551212)
>Exten => 145,n,set(CALLERID(name)=JOES POOL HALL)
>Exten => 145,n,Dial(SIP/sipuser,
On Mon, 06 Dec 2010 20:03:03 +0100, Gilles
wrote:
>Any idea why Asterisk shows nothing, and how to retrieve the original
>CID information?
Sorry about that, I forgot that the console had to be started in
verbose mode for NoOp() to display data:
> asterisk -r
On Mon, 6 Dec 2010 13:39:33 -0600, "Danny Nicholas"
wrote:
>#2 you might want to save the original ID to a variable, the reset
>CALLERID(num) to that variable. (if #2 is corrected, this one probably won't
>matter).
Thanks Danny, and sorry for the trouble: I was paying so much
attention to the wea
Hello,
I'm having the following problem when using a headset on XP
connected to an on-board Realtek soundcard on an AsusTek M2N68-AM Plus
motherboard:
- Using any sound recorder (Windows', Audacity, XLite), the level is
just too low when speaking at a conversational level, even with the
m
Hello
I need to find a recent and neutral comparison of the major products
available to connect an Asterisk server to the telephone network,
whether ISDN (BRI) or PSTN, and through a PCI card or some external
box. I'm told there are less issues (echo, stability) with external
boxes compare
On Tue, 07 Dec 2010 10:39:44 -0800, Dave Platt
wrote:
>Same headset model, or different headset model?
Different brand/model, but similar as they are both el cheapo,
entry-level headsets. I tried using them on a laptop, and I get
marginally better microphone output, even with its volume cranked a
On Wed, 8 Dec 2010 17:56:51 +0300, "Sevana Oy"
wrote:
>We would be happy to offer you Asterisk VQM for voice quality assessment,
>however, it's Asterisk based and works with every hardware that works with
>Asterisk:
>http://www.sevana.fi/aqua-powered-asterisk-voice-quality-monitoring-solution.p
On Wed, 8 Dec 2010 09:33:22 -0500, David Backeberg
wrote:
>* pay somebody else to do it in the form of appliance and lose most
>control versus do it yourself and have total control but also the
>chance to screw up.
Thanks for the input. Has someone in this ng tried a PCI card and then
an applianc
On Wed, 8 Dec 2010 14:46:59 -0500, David Backeberg
wrote:
>Both the cards and the appliances have had 'issues'.
Thanks guys for the input.
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On Wed, 08 Dec 2010 10:48:06 -0800, Dave Platt
wrote:
(snip)
I'll read up more about sound quality and Asterisk and see if
something can be done about this.
Thanks again for the help.
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On Wed, 08 Dec 2010 10:48:06 -0800, Dave Platt
wrote:
>It does sound as if the mic-input gain is too low for those
>headsets.
Disabling the on-board soundcard and using even an entr-level PCI
soundcard solved the issue. If some customers complain about low sound
when using the on-board soundcard,
Hello
For customers who need a small IP PBX to handle up to four ISDN lines
(in France, so I guess that means EuroISDN) instead of a PC + Asterisk
and an ISDN gateway box, has someone already played with the Atcom
IP-4B?
www.atcom.cn/IP-BRIM.html
Any feedback appreciated.
--
_
On Sun, 12 Dec 2010 20:02:00 +0100, Hans Witvliet
wrote:
>But as BRI / (aso known as ISDN2) is more a thing of the past, i mean
>pre-adsl, for the general public, the number of people with bri and
>hence their potential market is (too) small, i fear.
The problem with VoIP, is that this side of th
On Sun, 12 Dec 2010 23:49:50 +0100, Hans Witvliet
wrote:
>I don't know what their price-range is, (just going through their site)
>
>Other alternative i heard about, is the DSL-modems from AVM.
>What i heard, is that you can use the 7170 and 7270 (perhaps their
>latest models also) as an BRI-gatew
On Mon, 13 Dec 2010 12:06:56 +0100, Administrator TOOTAI
wrote:
>We are selling our own xDSL but a France Telecom Pro can do the job.
>Always dedicate the ADSL line to VoIP, use the right codec and you will
>have the quality you need. In big towns, some of our cutomers uses ADSL
>from Free Tele
On Mon, 13 Dec 2010 12:14:26 +0100,
klitz...@pool.informatik.rwth-aachen.de wrote:
>The built-in SIP proxy is made for "inside LAN" usage, although there are ways
>to make the
>box also accept SIP UAs on the Internet as local "phones". Do not expect too
>many features
>for these IP phones, for
Hello
I was wondering if someone knew of an application that could check
that the user has a firewall and a broadband connection that will work
OK with Asterisk and VoIP.
The app would first perform some bandwith + jitter tests, and will
then call a STUN server to check that the firewall isn't sym
Hello
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 numbers on the Net, such as those below
to perform an
On Mon, 13 Dec 2010 18:35:03 +0100,
klitz...@pool.informatik.rwth-aachen.de wrote:
>Until Asterisk 1.8 STUN support was faulty, and in 1.8 it has been corrected
>(?) and strongly
>limited. Search the asterisk-dev mailing list archive for STUN and do the same
>in the Asterisk
>bug tracker for mo
Hello
I'm having a difficult time finding precisely what to put in
sip.conf and extensions.conf (and possibly other files) to get a
working configuration to connect an Asterisk (1.4) server to a VoIP
provider with the Asterisk server + SIP clients located in a private
LAN behind a NAT rout
On Tue, 14 Dec 2010 16:56:14 +0100, Gilles
wrote:
>PS: Here's what I'm thinking of using:
At this point, Asterisk seems to register OK with my VOSP, but when I
call the number from my cellphone, I get this error:
"NOTICE[88]: chan_sip.c:14033 handle_request_invite: Call from
On Tue, 14 Dec 2010 11:19:48 -0600, Lyle Giese
wrote:
>You are setting up a SIP trunk from your VOSP provider(whatever VOSP
>is). It dials your phone number. So whatever you dial from your cell
>phone is the extension that this trunk should land at.
>
>'s' is not an extension. It's a placeholder f
Hello
At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT
set up with a VOSP trunk that I can use to make/receive calls to/from
the PSTN.
Now, I'd like to be able to call any number on the Net that is
advertised as "sip:u...@domain.com", such as those:
www.voip-info.org/wiki/vi
On Thu, 16 Dec 2010 10:06:35 +0100, Administrator TOOTAI
wrote:
>Why 2 context? Todays Asterisk versions only needs one peer context for
>incoming/outgoing. Something like
I tried combining the two sections in sip.conf, but get a BUSY signal
for incoming calls from the PSTN. Could it be because
On Thu, 16 Dec 2010 17:05:35 -0500, "Jamie A. Stapleton"
wrote:
>Just add something like this to your dialplan:
>
>exten=>1234,1,Dial(SIP/u...@domain.com)
>
>Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.
Thanks Jamie, but isn't there a universal way to solve this
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
wrote:
>Then create a prefix for SIP calls
>
>exten=>_9.,1,Dial(SIP/${EXTEN:1})
>
>and you dial 9u...@domain.com from XLite
>
>Remember that calling sip URL is not as easy with a phone. Imagine you have an
>ATA with DECT or POTS
>phone conn
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