On Thu, 2011-04-28 at 11:25 -0400, Bruce B wrote:
Hi everyone,
How can I introduce some distortion, echo, chopping sound and all
other bad quality things that can happen to a SIP trunk? I have plenty
of bandwidth and crisp clear lines so the only thing that I can think
of is to limit
On Sat, 2011-04-23 at 10:52 -0700, Jason Rogers wrote:
Where would one find, or better yet determine from code, all of the table
definitions for ARA dynamic families?
There seems to be some bits and pieces in various places around the internet,
ie. voip-info, the definitive guide, ect. but
Hi,
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other pitfalls?
hw
--
_
-- Bandwidth and
On Wed, 2011-04-06 at 13:57 -0700, Jonathan Thurman wrote:
On 11-04-06 03:53 PM, Hans Witvliet wrote:
I'm going to have a go with realtime mysql.
Just wondering, most examples i came across while googling, was with 1.6
systems.
So any drastic changes with 1.8.3, table-layout? other
On Fri, 2011-04-01 at 13:27 +0100, Roger Burton West wrote:
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,
On Thu, 2011-03-31 at 21:40 +0200, Michelle Konzack wrote:
Hello *,
I have an All-In-One intranet server samba3 and usualy a seperated
router as default gateway, which connect me using HSPA to the internet
Now I have installed an Huawai K3765 on my samba3, installed pppd +
On Mon, 2011-03-21 at 21:45 -0300, Juan wrote:
damn, advertisements everywhere, also in non commercial mailing lists...
ITSPTEC.COM seems don't understand what a NON-COMMERCIAL DISCUSSION is
about
I will never buy anything from people like you who don't seems to
understand so basic
On Fri, 2011-02-18 at 00:51 +0100, Albert wrote:
On 18.02.2011 00:30, Andrew Joakimsen wrote:
On Sat, Feb 12, 2011 at 07:31, ast guy ast...@gmail.com wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone
On Tue, 2011-02-15 at 05:57 +, A J Stiles wrote:
On Tuesday 15 Feb 2011, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
On Tue, 2011-02-15 at 09:01 +, Steve Howes wrote:
On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote:
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
What's the highest current 'genuine' one
On Tue, 2011-02-15 at 07:18 -0500, Richard Kenner wrote:
Anyway, the answer is: No, it's mathematically impossible to do
that. Even if the passwords were stored encrypted, Asterisk itself
has to be able to get the plaintext passwords to send to the remote
server; so the code to decrypt
kept on reading the thread...
Wouldn't it be better, for asterisk at least, to get rid of all this
identification / authentication stuff?
Keeping config files holding pain passwords or simple md5 isn't the way
to solve this...
Within the unix world those issues have been solved over and over
On Fri, 2011-02-04 at 15:43 +1000, Ryan Tucker wrote:
Hey Guys,
I'm after a way to monitor our sip trunks (peers) and send an email if they
go down.
I know I could use 'asterisk -rx sip show peers' in a shell script but that
seems messy,
especially since I'd like to monitor it fairly
Hi,
Anybody seen this before?
(using a pre-compiled asterisk from the OBS on a sles11sp1)
(I mean, i did the same with a 1.6 without any problem, but i need 1.8)
after starting:
kc3004:~ # /usr/sbin/safe_asterisk: line 145: 16133 Segmentation fault
(core dumped) nice -n $PRIORITY
On Wed, 2011-01-12 at 14:18 -0500, Mark Deneen wrote:
Static Key disadvantages
* Limited scalability -- one client, one server
* Lack of perfect forward secrecy -- key compromise results in total
disclosure of previous sessions
* Secret key must exist in plaintext form on each VPN peer
*
On Sun, 2011-01-02 at 00:10 +, Sebastian wrote:
Hi,
One possibility that you might want to explore is OpenVPN. If your VoIP
clients support OpenVPN (either through a local Openvpn client on the
clients network being used as an OpenVPN gateway, or through individual
clients supporting
On Sun, 2010-12-12 at 15:24 +0100, Gilles wrote:
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?
On Sun, 2010-12-12 at 21:35 +, Gordon Henderson wrote:
On Sun, 12 Dec 2010, 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
On Sun, 2010-12-12 at 23:07 +0100, Gilles wrote:
The problem with VoIP, is that this side of the pond/channel, there
aren't that many ADSL providers that also support VoIP and have
competitive offers for small businesses, and thus, I wouldn't
recommend ADSL for professional telephony.
On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know if this is in 1.4/1.6 as well, but
should be easy enough to test.
Here is an example of what I see on the manager interface during a
On Wed, 2010-11-24 at 15:47 -0600, Sherwood McGowan wrote:
On Wed, Nov 24, 2010 at 3:08 PM, Hans Witvliet h...@a-domani.nl wrote:
On Wed, 2010-11-24 at 08:29 -0500, Ryan Bullock wrote:
On Asterisk 1.8 when a SIP peer resgisters or unregisters it generates
a PeerStatus event. I don't know
Hi all,
Perhaps someone has dealt with it before.
I want to activate a bunch of my own scripts after someone has registred
om my asterisk, or when his cient has de-registerded.
have been skimming through AGI and AMI, and seen a lot of nice features,
but not the (de-)registering events.
Kind
On Wed, 2010-11-10 at 08:38 +0100, Olle E. Johansson wrote:
6 nov 2010 kl. 15.30 skrev Hans Witvliet:
Hi all,
As stated in the subject, slightly off-topic, as it is not directly a
Asterisk issue, but more SIP in general
Because security in general, and specifically identification
Hi all,
As stated in the subject, slightly off-topic, as it is not directly a
Asterisk issue, but more SIP in general
Because security in general, and specifically identification becomes
more and more a subject for more concern, and Asterisk is capable of
doing sip/TLS, i was wondering what more
On Sun, 2010-10-31 at 11:39 -0600, Joel Maslak wrote:
To guess an 8 character (which is short) password that consists of random
upper case, lower case, numbers, and 10 symbols (there are more you can use
if you want), the average number of passwords that you would have to try to
get in is:
On Sat, 2010-10-30 at 14:28 -0400, Zeeshan Zakaria wrote:
My main asterisk server is under unusual heavy attack, and so far
Fail2Ban has blocked about 30 IPs, from various different countries.
At this time it is blocking about 1 IP address every few minutes.
Just wondering if anybody else is
While on the subject,
what is digium doing on my port 113?
just from my logfile:
Oct 31 01:11:07 fw2 kernel: EXT; INC, INTRUDER IN=eth0 OUT=
MAC=08:00:20:da:3b:4a:00:90:1a:42:70:d3:08:00
SRC=216.207.245.17 LEN=40 TOS=0x00 PREC=0x00 TTL=247 ID=15394 PROTO=TCP
SPT=56211 DPT=113 WINDOW=0
On Fri, 2010-10-22 at 11:16 +0200, Dave Cotton wrote:
On 22/10/10 11:05, Hans Witvliet wrote:
On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote:
On 21/10/10 22:04, Hans Witvliet wrote:
For suse there is a precompiled version on the OBS (vitsoft)
Package search on the OBS shows
For those who might be interested...
If possible i rather use maintream prebuild packages.
As from now, they (asterisk180) are available for openSUSE_11.1,
openSUSE_11.2, openSUSE_11.3, SLES10, SLES11, SLES11SP1 via:
http://software.opensuse.org/search?q=asterisk180baseproject=openSUSE%
On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote:
On 21/10/10 22:04, Hans Witvliet wrote:
For suse there is a precompiled version on the OBS (vitsoft)
Package search on the OBS shows nothing for 1.8.0 at all.
Perhaps you know where it is hidden.
Dave Cotton
http
On Thu, 2010-10-21 at 01:15 -0400, Zeeshan Zakaria wrote:
Yes, one server will do it all. It will not be in a data center but at
customer premisis, so doesn't have to be 1U.
In that case, how about a dell-server?
And if it is not in a data center, take care of an UPS for both the
server and
On Thu, 2010-10-21 at 17:12 +0200, Dave Cotton wrote:
On 21/10/10 17:05, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make
On Tue, 2010-09-28 at 09:06 -0300, Daviramos Roussenq Fortunato wrote:
Hi List.
It is possible to travel over the X.25 protocol on Asterisk SIP?
--
Hi Daviramos,
You gotta be joking!
X.25 can only be found in telecom-musea's nowadays.
Latest development was the X.31 and X.32 interface
On Sun, 2010-09-26 at 23:16 +0200, Dmitry Nedospasov wrote:
Asterisk TFOT actually has a chapter on this, though it might be a bit
outdated [1]. It's Chapter 2, Preparing a System for Asterisk.
afaicr, i though that the magnificant book would get un update for the
1.8 release... (or are
On Sun, 2010-09-12 at 15:32 -0700, Kevin Keane wrote:
In terms of telephony, a T-1 can make a huge difference over DSL. DSL
gives you a lot of raw bandwidth, true, but for voice that really
doesn’t matter all that much. Voice calls only take a relatively small
amount of bandwidth anyway; you
On Mon, 2010-09-13 at 00:32 -0700, Kevin Keane wrote:
Latency also is the reason VoIP does not work at all over satellite
connections even though they tend to have plenty of bandwidth.
Please define does not work at all over satellite ???
Sure, it is not studio HIFI quality, but is th
On Wed, 2010-08-25 at 14:04 -0400, Matt wrote:
Has anyone successfully compiled the AMR codec into an Asterisk
install, and if so, what steps did you take?
--
_
Just noticed that packman has prebuild packages:
(with
On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote:
Might be worth your time to check out: http://www.humbuglabs.org/
Though they write:
...
insight into the enterprise’s telephony infrastructure. Utilizing a set
of none-intrusive analytical technologies, Humbug is capable of
On Thu, 2010-07-22 at 17:41 -0500, Karl Fife wrote:
enough amps to power the full load at the end.
You could do someting with passive POE--in other words not 802.2af POE, but
rather the 'dumb' kind of POE which just injects power on the unused pairs.
Passive POE (being passive) does
On Wed, 2010-07-14 at 23:52 -0400, C F wrote:
On Wed, Jul 14, 2010 at 5:03 AM, liuxin nyliuxin...@gmail.com wrote:
Hi,
probably a misconfiguration or you havent plugged the cable in yet.
OMG you are right, I forgot to plug in the cable. Hey but wait which
cable you talking about?
On Tue, 2010-07-13 at 06:53 -0400, cov...@ccs.covici.com wrote:
What you can do -- I don't know about nomad, but can you make them use
authentication?
Randy R randulo2...@gmail.com wrote:
On Tue, Jul 13, 2010 at 12:29 PM, cov...@ccs.covici.com wrote:
What I do, is only open port 25 to
On Wed, 2010-07-07 at 12:12 +0600, ABBAS SHAKEEL wrote:
Thanks to Gordon and Paul for kind help.
Actually we have a limitation to place the Asterisk server in client
premises if the server is in there premises then this means they have
full control over it.
harddisk encryption seems
On Wed, 2010-07-07 at 09:06 -0700, Steve Edwards wrote:
On Wed, 7 Jul 2010, Faisal Hanif wrote:
2nd option is by enabling execincludes=yes in asterisk.conf you can use
#exec in any of asterisk conf file to call any external application and
asterisk will use configuration returned by
On Fri, 2010-06-18 at 17:19 -0500, Cary Fitch wrote:
If domain name, what are they using for DNS?
Indeed, interesting question.
On a quiet moment i would suggest you do:
tcpdump -vvX -i eth0 port 53 on the machine that is your gateway to
internet, and make an internal call (or use wireshark)
On Sat, 2010-06-19 at 01:39 +0200, Philipp von Klitzing wrote:
Hi!
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip.
No iax. Why does the asterisk machine have to resolve any address?
Probably because you have one or more register = statements in your
sip.conf
On Fri, 2010-06-18 at 21:04 -0400, Andres wrote:
Our company has set up hundreds of asterisk boxes over the years. One
thing we learned early on was to avoid any type of DNS resolution by
asterisk. Asterisk gets hung when it can't access your DNS server and
all things grind to a
On Fri, 2010-06-18 at 13:59 -0400, sean darcy wrote:
If the internet server is down, there can't be a valid DNS server
accessible to Asterisk. The asterisk server is a caching name server,
but obviously won't be able to resolve addresses not in its cache.
Asterisk clearly doesn't need
Why no flash?
* Small pre-built PC (not buying board, case, all parts separately)
* Low power consumption
* No fan or very small fan
* Hard drive (not flash memory)
An ssd uses less power, so generates less warmth, hence less need for
fan in the drive area. Also less noise..
On Wed, 2010-06-09 at 19:43 +, Edwin Quijada wrote:
Just is PRI line you can do it..
No, not so.
I both have some PRI and BRI lines.
All of them have a main-number, and some additional numbers
Depending on what contract you have with your ISDN-provider the amount
of those number can vary.
Just curious,
Any chance of using amr for asterisk?
http://en.wikipedia.org/wiki/Adaptive_Multi-Rate_audio_codec
The codecs (both wb and nb) seems to be available at packman:
http://ftp5.gwdg.de/pub/linux/misc/packman/suse/11.2/src/amrnb-7.0.0.2-0.pm.5.1.src.rpm
On Mon, 2010-05-31 at 13:12 +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
On Fri, 2010-05-21 at 09:04 +0100, Gordon Henderson wrote:
On Fri, 21 May 2010, Steve Totaro wrote:
On Thu, May 20, 2010 at 7:09 PM, Leif Madsen
I'm still deploying 1.2 - Got one next week with an ISDN-30 connection and
40 seats. It just works and ticks all the boxes I need to tick for
On Sun, 2010-05-09 at 13:34 +0300, Tzafrir Cohen wrote:
On Tue, May 04, 2010 at 06:46:59PM +0200, isca...@free.fr wrote:
- Create a SSH tunnel from the Windows client to the Asterisk server using
putty
(redirecting ports used for VoIP)
= it doesn't work because either SIP/RTP or
On Sat, 2010-04-24 at 10:56 -0500, Michael Graves wrote:
On Fri, 23 Apr 2010 23:11:06 +0200, ad...@3a.hu wrote:
Hi Guys,
On 04-23-2010 21:40, Nathan Clemons wrote:
SIP is just the control protocol, and can be negotiated over TCP or UDP.
The
actual payload is done over RTP, which is a
On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote:
On Tue, 13 Apr 2010, Alyed wrote:
Think we need some solution WITHIN the Asterisk core. Roderick A. suggested
something that looks nice using iptables, some others have pointed out using
RBL or fail2ban, but the best would be to
On Tue, 2010-04-13 at 15:49 +0200, Philipp von Klitzing wrote:
Hi!
Any aditional security within * is fine, but if someone is simply
drowning your bandwith, action must be taken at a lower level.
Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip,
mail, ssh, ldap,
Just noticed that packman has precompiled versions of amr codec.
Both wideband and narrowband. Can these be used for asterisk?
Heard some nice about AMR (in general)
If so, any one around with experience with either??
hw
--
_
On Sun, 2010-03-21 at 19:00 -0400, Zeeshan Zakaria wrote:
Good to know. I'll try that. I needed such solution for a client few
months ago.
On 2010-03-21 6:06 PM, Gordon Henderson gordon
+aster...@drogon.net wrote:
On Sun, 21 Mar 2010, Zeeshan Zakaria wrote:
Virtual machine
On Thu, 2010-03-04 at 23:27 +, Steve Howes wrote:
On 4 Mar 2010, at 23:11, Steve Edwards wrote:
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create
On Wed, 2010-01-20 at 17:06 -0800, Jim Dickenson wrote:
My development system for asterisk is a virtual CentOS 5.4 world running
under Fusion on my MacBook. I am usually only doing a few calls at a time. I
have an IAX trunk to our office Asterisk PBX so I can access the PRI line
there. I do
On Fri, 2010-01-15 at 12:17 -0500, John Todd wrote:
I don't know how many of you are going to be at ITEXPO/Digium Asterisk
World in Miami next week - I hope to see as many of you as possible,
though.
A bit too far, i'll be at fosdem, Brussels
--
On Fri, 2010-01-15 at 19:10 -0700, Andrew Hakman wrote:
2 cables is definitely the best, followed by a cheap gig switch at each desk.
GB lan for interconnecting computers would be optimal.
Seperate cabling for voip: ok, but i wouldn't put a switch at every
desktop: A 100MB switch with
On Fri, 2010-01-15 at 14:11 +0100, randall wrote:
its not the network switch that i'm worried about, its the build in
switch of the phones with the double network card
--
Hi,
Don't think you'll find phone's with an internally gbit switch.
As for voip it is not needed.
If you connect
Perhaps slightly O.T.
Does anybody know of (or even better, has experience with)
softphone clients on a blackberry?
Some friends of mine have those devices, but they can only use it for
data, voice has been disabled in their abo, so much of them carry they
business-BB, and their private GSM.
On Thu, 2009-11-12 at 20:18 -0700, Joseph wrote:
Digium has discontinued their ATA iaxy adapter; don't blame them, too
expensive so they can not compete.
Compete, With which iax-ata ???
___
-- Bandwidth and Colocation Provided by
On Thu, 2009-11-12 at 18:59 -0500, Martin wrote:
Grandstream HT503. For me works just fine. 1xFXO 1xFXS port. Each port
has its own sip account.
Martin
- Original Message -
From: jonas kellens
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Mon, 2009-11-02 at 09:37 +, aster...@opensourcesolution.in wrote:
hello friends
friend i had just finished my chapters of asterisk. ill be
configuring asterisk in for home for r/d purpose. i am having p4
machine with 1 GB RAM, ill be configuring asterisk on centos 5.3, the
only doubt
On Wed, 2009-10-28 at 14:59 +, Ott Rose wrote:
I am sure many of you have seen my post asking question that I cannot
seem to resolve. While the responses i have been getting have been
helpful i still cannot seem to get this working 100%.
So I have waving the white flag here. I give
On Tue, 2009-10-27 at 09:34 +, aster...@opensourcesolution.in wrote:
hi ,
i have started reading asterisk book need your guidance.friend as i
am newbie in asterisk so plz plz forgive me if i ask stupid questions.
Installing Asterisk
- on which linux flavour i should start the
On Tue, 2009-10-13 at 14:42 -0500, Karl Fife wrote:
I think one of the very best options is pfSense. Free Open-source,
but it's BSD based, rather than LINUX based. As such it has a lower
risk of external exploits. The user-interface makes it incredibly
simple to set up and maintain. There
On Tue, 2009-10-06 at 22:11 -0700, Kirill 'Big K' Katsnelson wrote:
On 091001 0406, Mindaugas Kezys wrote:
We had many problems with IAX2, changing to SIP solved them all.
Let me paste link to wise-words which clearly illustrates our experience:
On Tue, 2009-10-06 at 17:03 +0100, Gordon Henderson wrote:
On Tue, 6 Oct 2009, Faraz Khan wrote:
In pakistan they have protocol scanners mounted on all the 4 fibers that
enter and leave pakistan. They can detect VOIP usage by upload/download
patterns / etc. They have invaded and arrested
Just detecting this tread...
Moving to Debian is quite a big step.
How about updating to openSUSE_11.1 and use the prebuild asterisk
packages (either zaptel or dahdi) .
On the OBS they are available for 1.4.x, 1.6.0, 1.6.1
hw
___
-- Bandwidth and
On Sat, 2009-09-26 at 22:47 -0700, Dave Platt wrote:
Isn't an SSL based tunnel all TCP?
There seems to be a good deal of feeling (and evidence) that
trying to use TCP as the container for a tunnel is likely
to cause more trouble than it solves. Yes, the TCP layer
will make the tunnel
On Sat, 2009-09-26 at 21:54 +0500, ABBAS SHAKEEL wrote:
Thanks Alex
By just avoiding this will solve this problem?
No,
Just moving the asterisk-server before the firewall won;t do any good.
because in that situation the firewall is in between asterisk and your
LOCAL sip-clients: you
On Sat, 2009-09-26 at 20:07 +0100, Alan Lord (News) wrote:
On 26/09/09 19:42, Hans Witvliet wrote:
snip /
What you can do (perhaps not the best solution...) is having one
asterisk server behind your firewall, serving all your local
sip-clients. And another at the other side
On Sat, 2009-09-26 at 19:32 +, Jeff LaCoursiere wrote:
On Sat, 26 Sep 2009, Alan Lord (News) wrote:
Hmmm, has anyone tried SIP over a VPN?
We are thinking of testing this but haven't yet...
Al
I have a client with Sonicwall VPNs. Asterisk is at head office on
internal
On Sat, 2009-09-26 at 22:09 +, Jeff LaCoursiere wrote:
On Sat, 26 Sep 2009, John A. Sullivan III wrote:
snip
We are using SIP over both IPSec and SSL VPNs very successfully with
access controls in the tunnel ingress via the ISCS network security
management project
On Thu, 2009-09-24 at 16:20 +0300, Tzafrir Cohen wrote:
On Thu, Sep 24, 2009 at 02:47:18PM +0200, Vincent wrote:
Hello
I assume I'm not the first one to think about this: Is it possible to
connect an intercom and/or door bell to Asterisk, so that I can get an
e-mail that someone rang
On Thu, 2009-09-24 at 09:56 -0400, jon pounder wrote:
Dean Collins wrote:
Earlier in the thread someone made a comment about using gsm since
everyone had gsm handsets already.
Can you explain in detail please ? (what hardware specifically, and how
does this actually work ?) My ignorant
On Wed, 2009-09-23 at 09:39 -0700, mgra...@mstvp.com wrote:
I had a good experience with that Polycom/Spectralink phone. Very rugged
as you say. The experience did highlight the weaknesses in consumer
Wifi AP, which reinforced my commitment to continue using DECT around my
office.
Michael
On Wed, 2009-08-26 at 06:18 +1000, Alex Samad wrote:
any thoughts of different media like 10G ethernet or infiniband ?
For another project i had a look at 10G.
Prices of the nic's were reasonable, but even an 5-ports nice were mind
blowing (35,000 Euro)
So i opted for multiple nic's and
On Tue, 2009-05-26 at 10:26 -0400, John Novack wrote:
That is a pretty long run.
The type of analog phone can be an issue. How LITTLE loop current will
it operate on? Most need more than 20 Ma to signal properly, and the
voltage output of the ATA needs to be known
Type of signaling? DTMF?
On Tue, 2009-05-26 at 21:29 +, asterisk-us...@rogg.is wrote:
Appreciate all your input folks. Much of it very helpful in the greater
context of the initial question.
Thank you for the suggestion of using various wireless devices, but I'm
stuck with fixed wiring since this is a
On Sun, 2009-05-24 at 16:21 +0400, Manoj Panicker - FOES wrote:
Hi ,
Any idea as how to divert the Incoming PSTN calls on the FritzBox
to one of the Numbers in the Asterisk domian? and vice versa.
I want ot use the FritzBox as the bridge between the PSTN and Astrisk
Thanks
Manoj
On Mon, 2009-05-25 at 17:07 +0200, Ngo-Vi Hoai-Anh wrote:
is installing asterisk directly on FritzBox an option for you? If yes
I 'v found an interesting link
http://www.ip-phone-forum.de/showthread.php?t=146132
Manoj Panicker - FOES schrieb:
Hi ,
Any idea as how to divert
On Mon, 2009-05-25 at 22:19 +0200, Philipp von Klitzing wrote:
Hi!
looks interesting, indeed, but as the O.P. wanted to divert PSTN call,
one would need chan_dahdi.so or chan_misdn.so/chan_capi.so (If the
hardware of Fritz is capable of it)
Divert-ing is a misleading term in this
On Fri, 2009-05-08 at 20:30 -0700, Eric Fort wrote:
Could those on the list who have used or tried to use VoIP over a
satellite internet connection comment on how well it works or if it
even works at all in a reliable way. What is the effect of latency on
the VoIP path and how much is
Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and IPv6
Date: Mon, 28 Jan 2008 18:45:42 -0600
Hans Witvliet wrote:
Any progress on IPv6 ?
Still completely seperate
On Tue, 2009-04-14 at 09:45 -0700, awerf...@hotmail.com wrote:
Hi Friend,
How are you doing recently?
I'm getting bored.
___
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On Fri, 2009-04-10 at 23:55 -0300, David fire wrote:
hi
how i can give the control of a digium card to the virtual machine? i
am using XEN
do you recomendo other virtual machine? VMWare openVZ etc...?
with lspci you can determine the hardware adres of your boards, and pass
it through from
On Wed, 2009-04-01 at 09:18 +0200, Olle E. Johansson wrote:
snip
For more information, please do not contact Digium sales.
To be released: 2009-04-01
Should say enough...
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On Wed, 2009-04-01 at 11:41 -0500, Brent Davidson wrote:
Cary Fitch wrote:
It uses proprietary EDC. (Extreme Data Compression) The 140 bytes at 8
bits each, and that is 2^140^8, a nearly inexhaustible key number which is
related to audio and video data simultaneously stored on a Google
On Mon, 2009-03-23 at 23:15 +, Steve Howes wrote:
On 23 Mar 2009, at 22:44, Hans Witvliet wrote:
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
concrete shielding wireless phones.
So i was thinking
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
concrete shielding wireless phones.
So i was thinking to use one of those usb-phones, and plug it into one
of my servers there.
But what i read from the thread, i seems
For those who are using SuSE:
At last they've managed to create ready-to-run packages for
openSUSE_11.1. They are there since a couple of hours...
(For other versions it was allready available for some time on the OBS)
On Wed, 2009-02-25 at 19:14 -0200, David fire wrote:
please keep us informed about it.
David
2009/2/25 Kristian Kielhofner kristian.kielhof...@gmail.com
Hello everyone,
I just ordered one of these:
On Thu, 2009-02-19 at 15:22 +1300, Michael wrote:
This is everything that is wrong with Open Source - no body wants to pay for
anything
Your statement is not correct!
Well atleast half of it. You should have said:
-no body wants to pay for anything-
This has absolutely nothing to do
On Thu, 2009-02-05 at 22:01 +0100, Philipp Kempgen wrote:
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
snip
How about .. dialplan.conf .;-)
On Mon, 2009-02-02 at 22:25 -0200, Alejandro Cabrera wrote:
Dear all, I've implemented an Asterisk 1.4 with SIP service for voip and
video. So I can establish a voip + video connection *one-to-one*
onlyit works OK.
But I'd like to implement a videoconference *one-to-many* in order to
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