Hello
I'd like to build myself an Asterisk server for SOHO use. Intel's
D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very
good deal, but I'm concerned about two things:
1. Will an A400P (from OpenVox, but supposed to be Digium-compatible
http://tinyurl.com/ck6nfu) fit with a
Paulo Santos I did some tests on it, not many. Without going higher
than 2.0 load average I managed to do 10 calls per second, lasting 5
seconds each. During those 5 seconds, 2 sound files were played
(sln). MySQL CDR was enabled, so that's also 10 DB writes/second. I
don't know exactly what
On Mon, 31 May 2010 13:12:25 +0200, lesouvage i...@meetmecall.nl
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
On Sun, 30 May 2010 02:45:51 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com 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
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
On Thu, 3 Jun 2010 08:24:11 -0500, Danny Nicholas
da...@debsinc.com 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
On Fri, 04 Jun 2010 11:20:49 +0200, Gilles codecompl...@free.fr
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
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
On Mon, 21 Jun 2010 14:06:22 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net 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 ;-)
--
On Mon, 21 Jun 2010 17:25:09 +0300, Motiejus JakÂtys
desired@gmail.com 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
On Mon, 21 Jun 2010 16:47:08 -0700, CunningPike
cunningp...@gmail.com 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
On Mon, 21 Jun 2010 16:10:12 +, Edwin Quijada
listas_quij...@hotmail.com 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
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
On Fri, 25 Jun 2010 09:53:34 +0200, Randy R randulo2...@gmail.com
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
On Fri, 25 Jun 2010 11:43:04 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com 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
On Fri, 25 Jun 2010 12:09:09 +0200, Randy R randulo2...@gmail.com
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.
--
On Fri, 25 Jun 2010 11:19:25 +0100, Gareth Blades
list-aster...@skycomuk.com 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
On Fri, 25 Jun 2010 13:25:18 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com 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.
taking 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 access the mailing list through Gmane
On Fri, 25 Jun 2010 12:32:03 -0400, Barry Miller
asterisk-us...@notanet.net 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
On Fri, 25 Jun 2010 08:59:32 +0200, Gilles codecompl...@free.fr
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
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
On Sat, 26 Jun 2010 21:09:54 +0300, Eyal Goltzman
egoltz...@gmail.com 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
On Sat, 26 Jun 2010 13:35:12 -0400, Paul Belanger
paul.belan...@polybeacon.com 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
On Sat, 26 Jun 2010 17:53:27 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net 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
On Wed, 30 Jun 2010 17:01:50 -0700 (PDT), Steve Edwards
asterisk@sedwards.com 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,
On Thu, 01 Jul 2010 15:22:33 +0500, Faisal Hanif fai...@vopium.com
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.
On Thu, 01 Jul 2010 01:32:08 +0200, Gilles codecompl...@free.fr
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
tzafrir.co...@xorcom.com 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
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
On Mon, 05 Jul 2010 12:45:34 +0200, Gilles codecompl...@free.fr
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 about
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
On Fri, 9 Jul 2010 08:06:04 -0400, Ryan Wagoner rswago...@gmail.com
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
On Fri, 09 Jul 2010 13:45:18 -0500, Shaun Ruffell
sruff...@digium.com wrote:
# lsmod | grep -i wc
wctc4xxp 32414 0
dahdi_transcode 5751 1 wctc4xxp
wcb4xxp33905 0
wcfxo 8968 0
wctdm24xxp116684 0
wcte11xp
On Sat, 3 Jul 2010 13:47:23 -0500, Tilghman Lesher
tles...@digium.com 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
da...@debsinc.com 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
da...@debsinc.com 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
da...@debsinc.com 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
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.
--
On Tue, 10 Aug 2010 10:45:59 -0600, Dave d...@mynatt.biz 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.
--
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
On Mon, 18 Oct 2010 17:36:30 +0530, Jigar Joshi jiga...@gmail.com
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
On Mon, 18 Oct 2010 17:54:27 +0530, Jigar Joshi jiga...@gmail.com
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
On Mon, 18 Oct 2010 19:12:48 +0530, Jigar Joshi jiga...@gmail.com
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
On Mon, 18 Oct 2010 13:09:50 +0200, Gilles codecompl...@free.fr
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
On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneen mden...@gmail.com
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,
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:
On Fri, 19 Nov 2010 10:15:40 -0500, jon pounder j...@inline.net
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
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.
--
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http://www.asterisk.org/hello
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
On Mon, 6 Dec 2010 10:15:34 -0600, Danny Nicholas
da...@debsinc.com 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 =
On Mon, 06 Dec 2010 20:03:03 +0100, Gilles codecompl...@free.fr
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
ip04
On Mon, 6 Dec 2010 13:39:33 -0600, Danny Nicholas
da...@debsinc.com 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
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
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
On Tue, 07 Dec 2010 10:39:44 -0800, Dave Platt dpl...@radagast.org
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
On Wed, 8 Dec 2010 17:56:51 +0300, Sevana Oy sa...@sevana.fi
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:
On Wed, 8 Dec 2010 09:33:22 -0500, David Backeberg
dbackeb...@gmail.com 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
On Wed, 8 Dec 2010 14:46:59 -0500, David Backeberg
dbackeb...@gmail.com 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 dpl...@radagast.org
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 dpl...@radagast.org
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
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 h...@a-domani.nl
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
On Sun, 12 Dec 2010 23:49:50 +0100, Hans Witvliet h...@a-domani.nl
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
On Mon, 13 Dec 2010 12:06:56 +0100, Administrator TOOTAI
ad...@tootai.net 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
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 example
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
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
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 more
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
On Tue, 14 Dec 2010 16:56:14 +0100, Gilles codecompl...@free.fr
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 l...@lcrcomputer.net
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
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:
On Thu, 16 Dec 2010 10:06:35 +0100, Administrator TOOTAI
ad...@tootai.net 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
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton
jstaple...@computer-business.com 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
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net 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
On Thu, 16 Dec 2010 11:54:31 +0100, Gilles codecompl...@free.fr
wrote:
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:
I mean: Do I really have to first create a section in sip.conf each
time a user needs to call a number on a new
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic
On Fri, 17 Dec 2010 09:00:39 -0500, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
You have to tell it the host to request the extension from. All you're doing
is
dialing SIP/*031600, which with that format, is going to try and call
[*031600]
as defined in sip.conf.
You're missing the host
On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil
d.nik...@cem-solutions.net wrote:
Does anyone ported Asterisk to Android OS .please give details
www.servalproject.org
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On Fri, 17 Dec 2010 17:54:00 +, Roger Burton West
ro...@firedrake.org wrote:
How would you _expect_ to be able to specify a destination server from a
telephone keypad?
Thanks guys for the infos. My goal was to learn how to configure
Asterisk so it could call SIP URI (u...@domain) using XLite,
Hello
I've done some googling, but still puzzled at my working
configuration.
Apparently, a user can only receive calls through Asterisk, a peer
can only make calls, and a friend can do both.
If that's correct, I don't understand why my VOSP requires the
following settings in sip.conf
On Mon, 20 Dec 2010 12:39:44 -0600, Kevin P. Fleming
kpflem...@digium.com wrote:
You've missed a very important point here: you are using a *SIP*
endpoint to call a *SIP* URI. The endpoint can do that directly, and
doesn't need any help from Asterisk to do it. If you wanted to be able
to
On Tue, 21 Dec 2010 14:20:55 +, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
The same way Ubuntu, Slackware, CentOS c. differ from each other. They are
all using the Linux kernel and the X Window System under the bonnet. Well,
every Free and Open Source telephony system is using
Hello
I have an Asterisk 1.4 server and two XLite softphones, where
Asterisk and the local XLite phone are located in a LAN behind a NAT
router, and the remote XLite phone is located elsewhere on the Net
behind its own NAT router:
http://img252.imageshack.us/img252/3749/asterisknat.png
On Wed, 22 Dec 2010 14:31:32 +0100, Stefan Schmidt s...@sil.at wrote:
you have a typicall nat issue. Asterisk receives messages from the phone
but cannot send any messages back (thats why it tries to resend the 200
ok message 6 times).
try setting qualify=yes to your sip peers config to keep the
On Wed, 22 Dec 2010 13:18:38 +, Steve Davies davies...@gmail.com
wrote:
Look in the XLite advanced network settings and disable the 2 timeout
settings (RTP and RTCP?). This is not always necessary, but there are
sufficient cases where the packets XLite expects appear early on, but
do not
On Wed, 22 Dec 2010 13:22:47 -0500, Bruce B bruceb...@gmail.com
wrote:
This is a NAT issue like noted before.
Try:
localnet=192.168.0.0/ http://192.168.0.0/24255.255.255.0
instead of:
localnet=192.168.0.0/24
http://192.168.0.0/24Also, make sure you have all your VPN connections as
localnet and
On Thu, 23 Dec 2010 15:54:59 +0100, Jeroen Eeuwes
jeroeneeu...@gmail.com wrote:
In sip.conf I have added for all the remote users the setting
canreinvite=no. The downside to that setting is that Asterisk is
always in the audio path. For my situation that does not really
matter.
Thanks Jeroen.
On Sat, 25 Dec 2010 09:49:29 -0500, John Ervin jer...@jervin.com
wrote:
So, assuming your Asterisk box is behind one firewall (Linksys/Tomato
Software) and your Wireless SIP phone is behind another firewall
(SonicWall 1260 Enhanced). Is there anything special that I have to do
to the
On Mon, 27 Dec 2010 09:14:22 + (GMT), Gordon Henderson
gordon+aster...@drogon.net wrote:
I've used OpenVox analogue cards. They seem to just work without having
to do anything special.
+1. I have an OpenVox with a single FXO module, and it's been working
for 4 years now. I don't know the
Hello
I don't have a landine and use a VOSP to provide access to the
telephone network.
In case a call comes in and I'm not home, I'd like Asterisk to log the
call, and then send an SIP message to my VOSP so the call is forwarded
to my cellphone and is thus charged to the caller, without
On Thu, 30 Dec 2010 23:53:04 -0500, Jeremy Kister
asterisk...@jeremykister.com wrote:
I've got just about everything turned on via menuselect, but then i have
a bunch of modules turned off via modules.conf
Incidently, is there a sure-fire way (eg. checking error messages in
Asterisk's log file)
On Fri, 31 Dec 2010 08:11:18 -0600, Danny Nicholas
da...@debsinc.com wrote:
Incidently, is there a sure-fire way (eg. checking error messages in
Asterisk's log file) to know which modules a given Asterisk setup
needs, so we can safely not load unneeded modules?
Check /var/log/asterisk/full from
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