configure the asterisk
ser.cfg
if(uri =~"sip:1024#"){
~ log(1,"Forwarding to Asterisk\n");
~ setflag(1);
~ rewritehostport("192.168.1.3:5061");
~ t_relay();
}
asterisk
thanks hans
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using
.
e. Setup voice mail and make sure PBX transfers calls to it when
appropriate.
Can this be done like this ?
Regards,
Hans Vledder
The Netherlands
P.S. He who comes up with clean internal ISDN bus (point to multi-point)
support for Asterisk, based on CologneChip based equipment receives an 18"
monitoring your site on a 24hrs/day basis from now on,
please keep me posted on both the driver and the 4 port BRI interface going
GA !
Regards,
Hans
P.S. This is excellent !
-Original Message-
From: Klaus-Peter Junghanns [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 4:35 PM
w how
it will operate with * yet but it seems rather transparant.
Regards,
Hans
-Original Message-
From: Andrew Nelson [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 18, 2003 9:35 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] PBX (Asterisk) <-> Cellular Phone Network
Maybe s
Hi Brian,
Excellent job, but how about calling the application 'cdr_odbc' instead of
'cdr_unixodbc', because up to now 'unix' is obvious/trivial when it comes to
* isn't it? Besides, I think 'cdr_odbc' is more in line with cdr_mysql and
cdr
here and live by -- please just dress local and act local, so we can
finally stop smoking pot just to keep up foreign misconception ...
:-)
Regards,
Hans Vledder
The Netherlands
-Original Message-
From: Roy Sigurd Karlsbakk [mailto:[EMAIL PROTECTED]
Sent: Monday, December 01, 2003 3:44 PM
T
All,
I would like to play an announcement to the user on what external line a
call came in, right before this call get bridged to this user. How would I
go about implementing this in * ?
Regards,
Hans
--
The contents of this e-mail are intended for the named addressee only. It
contains
nce I am only
planning an Asterisk PBX currently (awaiting the 4BRI + drivers of Kapejod),
I am not able to actually test it yet.
Cheers,
Hans
-Original Message-
From: Lenny Tropiano [mailto:[EMAIL PROTECTED]
Sent: Monday, December 01, 2003 6:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-
ncorporates the A(x),
but in turn is still somewhat buggy ?
Hans
-Original Message-
From: Lenny Tropiano [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 1:19 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Call Announcement - How To ...
> Thanks for your response.
Hi Dan,
I've been investigating the same thing. Try to Google for Asterisk+Soekris,
Soekris is the company (http://www.soekris.com) that makes cute little 586
class fan-less single board computers that run both Linux and FreeBSD ...
Good luck,
Hans
-Original Message-
From: [
Hi Kristian,
Anywhere I can read about this Soekris/AstLinux project? ...
Regards,
Hans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kristian
Kielhofner
Sent: Thursday, February 24, 2005 6:02 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non
USB.
My intention is to allow incoming and outgoing calls from SIP to ISDN. Is
this setup in any way supported by *?
Regards,
Hans Vledder
The Netherlands
--
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are
;s supported by Linux, but
there's no info on access to B and D channels.
Regards,
Hans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 10, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussi
On Fri, 2007-12-14 at 20:38 +0100, Vincent wrote:
> On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins <[EMAIL PROTECTED]>
> wrote:
> >I have to reboot my desktop xp box daily for it to run well.
>
> I haven't rebooted my XPSP2 in months,
I also have no problem with the stability of Windows XP
On Thu, 2007-12-13 at 22:21 -0600, Tilghman Lesher wrote:
> On Thursday 13 December 2007 19:55:39 Vincent wrote:
> > I was wondering why there doesn't seem to a Windows version of Zaptel,
> > making the Digium and its clones unavailable for a Windows PBX.
>
> Because nobody has done it yet. The r
That analyses all of the config files, and gves an error with file-name
& line number of the offendig config
(perhaps with a suggestion of what it MIGHT be...
Would be worthwhile for everybody writing configfiles manually,
Not only for migrating purposes...
Hans
__
On Sun, 2007-12-23 at 09:06 +0100, Vincent wrote:
> On Sun, 23 Dec 2007 06:15:22 +0200, Tzafrir Cohen
> <[EMAIL PROTECTED]> wrote:
> >Yes. I have a version of our CD that boots from PXE. It took minor
> >changes and rebuilding as a "PXE image", as Debian Live has basic
> >support of that already.
On Mon, 2007-12-24 at 04:11 -0700, Anthony Francis wrote:
> Axel Thimm wrote:
> > On Mon, Dec 17, 2007 at 10:40:32PM +0100, Benny Amorsen wrote:
> >
> >> Olle E Johansson <[EMAIL PROTECTED]> writes:
> >>
> >>
> >>> But on the other hand, if people rely on third-party distributions
> >>> we
et up a connection from a different IAX2 device (zoiper) with a
different
configuration behind the same public ip-address.
Thanks,
Hans Feringa
___
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asterisk-users mailing list
To UNSU
On Sat, 2008-01-05 at 13:23 -0800, Steve Edwards wrote:
> On Sat, 5 Jan 2008, Gres + wrote:
>
> > please can anyone help me knowing if i can install Linux and Asterisk on
> > HP servers
>
Yes, have asterisk with sles / centos of dl360 servers
hw
___
or the later, there are prebuild packages and regular updates, see:
http://ftp5.gwdg.de/pub/opensuse/repositories/network:/telephony/
An rpm -ivh on the commandline, or a search on yast will install what
ever you need
Hans
___
--Bandwidth
On Mon, 2008-01-07 at 01:50 +0200, Tzafrir Cohen wrote:
> On Mon, Jan 07, 2008 at 12:41:11AM +0100, Hans Witvliet wrote:
> > On Sat, 2008-01-05 at 13:36 -0500, Shane D wrote:
> > > Hello List,
> > >
> > > I am getting Asterisk set up. I am going to be insta
On Thu, 2008-01-10 at 08:35 +0100, IT-Connect wrote:
> stoffell schrieb:
> > > Has anyone been able to get ISDN-BRI support to work reliably on
> > > Asterisk 1.4? If so, I'd love to know how you did it (hardware,
> > > distro, kernel, modules, versions, config files).
> > >
> >
> >
> I'
On Wed, 2008-01-16 at 15:52 -0800, Steven wrote:
> I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri
> 1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a
> TE220B. The system load is below 0.10.
>
> I moved the server into production, with one PRI, on Fri
rained considering
ESD and their manufacturing department has to follow ever ESD rule.
(Otherwise governemental departmens won't do business with them)
Even medium-sized companies have people who are completeli ESD-ignorant.
So for serious installations, either commerical-quality, or diy, if you
On Sun, 2008-01-27 at 20:19 +0100, Benny Amorsen wrote:
> John Millican <[EMAIL PROTECTED]> writes:
>
> > I am trying to avoid loading a soft phone since I don't want to have
> > to instruct the users on how to use one (mostly NON-technical
> > types).
>
> You can't have a USB handset without a s
eased.
>
Any progress on IPv6 ?
Still completely seperate code, or is it already being merged into the
tree...
Perhaps i overlooked it, but i couldn't find any reference in:
http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co
HtH, Hans
__
the otherway around.
Leaving removed key=value from the configuration file unchanged in the
channel configuration after a reload.
Is this normal behaviour? I could not find any information for it. It
sent me in the wrong directions before I realized what was happening.
rgds,
Hans Feringa
On Mon, 2008-03-03 at 10:14 +0800, NOC Ph wrote:
> Hi Guys,
>
>
>
> I’m new in VoIP, I heard from a friend that asterisk is good in VoIP
> service especially on SIP. I’m planning to replace our old PBX system
> (legacy of Panasonic) to VoIP so that even out of the country we can
> still communi
On Wed, 2008-03-05 at 14:46 -0800, bilal ghayyad wrote:
> Hi All;
>
> Anyone tried to install Asterisk based on UNIX (not
> linux)? Which UNIX was good to work with Asterisk?
>
> Regards
> Bilal
On what ground would you refrain from using any linux based distro?
___
On Thu, 2008-03-06 at 08:21 +0100, randulo wrote:
> On Thu, Mar 6, 2008 at 5:32 AM, Carole Migden <[EMAIL PROTECTED]> wrote:
> > Generally what you know is best
>
> This is close to the best advice I've seen on this list in the last 5
> years! The rest is a question of religion ;)
>
Should have
On Sat, 2008-03-08 at 23:24 +0100, Grygoriy Dobrovolskyy wrote:
> I had same problem in france, not much choice, i have ordered from
> germany http://www.voipango.de
>
> This is not an advertisement i am not working for them.
>
> Are you sure about PRI ? in europe it's bri as i heard, PRI is in
>
On Mon, 2008-03-10 at 16:05 +1300, Matt Riddell wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Has anyone done any integration with this?
>
> All I know so far is that it appears to use some non standard form of SIP.
>
> Any pointers?
afaik, one of my colleages has done something
On Mon, 2008-03-24 at 14:29 -0400, Alex Balashov wrote:
> mark morreny wrote:
> > Hi
> >
> > I am working on deploying voip for my company and would like to seek
> > some advice on the number of E1 lines we need to rent. Our telco told
> > us that there can be at most 30 concurrent channels on
On Thu, 2008-04-10 at 16:05 -0400, Al Baker wrote:
> Remember - True TELCO grade systems simply cannot be compared to
> anything else.
> You want the reliability , uptime, and all the bells and whistles of
> true Carrier Grade Hardware/Software
> then you pay for it. If you want something "you" c
itch
is hard to do. And with enough bandwith on GB switches nowadays, i fail
to see the strict requirent to isolate voip from data-traffic
Perhaps you could clarify, as i am not AVAYA-certified.
Hans
--
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C
On Tue, 2007-05-29 at 14:51 +1200, Carlos Hernandez wrote:
>
> Please get in touch off list.. We're wanting to hire a professional
> subcontractor, developer or company to get around some issues like these:
>
> Please let me know if you have done this type of work before. We are not
> wanting
ed, use dialplan "K"
if DiD "Q" is called, use dialplan "L"
3)
Filtering on
- specific CLID
- anonymous calls (no clid)
All help is welcome
Hans
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
as
pendant.
How much of these improvements have already made it into the 1.4
branche?
Hans
--
pgp-id: 926EBB12
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Registered linux user: 75761 (http://counter.li.org)
___
--Bandwidth and Coloc
R and ES on the digium site)
Any pointers for german, dutch, greek, italian, prompts?
Hans
--
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12
Registered linux user: 75761 (http://counter.li.org)
___
--Bandwidth
On Wed, 2007-07-04 at 09:57 -0500, John Faubion wrote:
> >Is it just me? After the mail list server upgrade, the average delivery
> >time for messages to the users list is between 4 and 5 days. The Dev
>
> I've seen several people mention it taking a few days to send messages. I've
> usually see
On Sun, 2007-07-01 at 18:27 -0500, Russell Bryant wrote:
> Hans Witvliet wrote:
> > Before taking a plunch into the code
> > Marc Blanchet wrote that he's making code ip-version independant.
> > How much of these improvements have already made it into the 1.4
> &
thy tutorials to to install firmware,
what all the new features and bugfixes are, but no point to howto obtain
the latest version
HtH, Hans
--
pgp-id: 926EBB12
pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12
Registered linux user: 75761 (http://counter.li.
angup("SIP/kimura1-08236550", "")
in new stack
== Spawn extension (macro-fastbusy, s, 5) exited non-zero on
'SIP/kimura1-08236550' in macro 'fastbusy'
== Spawn extension (macro-fastbusy, s, 5) exited non-zero on
'SIP/kimura1-08236550'
Any h
I dialed it, but I am still thirsty. ;-)
> On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
>> On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
>> > Gordon Henderson wrote:
>> > > ; *99:
>> > > ; 99 bottles of beer on the wall.
>> > >
>> > > exten => *99,1,Noop(99 Bottles of be
Thanks for your response. My answers below.
> On Fri, 17 Aug 2007, Hans Feringa wrote:
>
>> I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
>> Band Analoog FXO) working with Asterisk.
>> I had a working FXO configuration to a analog port of a sm
On Wed, 2007-09-12 at 09:19 -0400, Jon Pounder wrote:
> there is tons of information about linux and flash drives on the
> nslu2-linux.org and the openwrt sites.
>
> main points :
>
> - disable swap
> - disable atime
> - disable most logging
>
> once the drive is not being written to then it w
Hi all,
Probably this is the wrong place to ask,
but is there an estimated time of arrival of the future?
i.e. TFOT--next generation dealing with * -1.4
I attended a workshop some time ago, and the book was part of the
package
HtH, Hans
On Tue, 2007-10-09 at 10:14 -0400, Matt wrote:
> Before you put any work into this... ask yourself... what exactly are
> you hoping to accomplish? There is no way one system can handle a
> DS3s worth of traffic... therefore, what good would this do?
>
I presume you can compare it with an ETSI C3
On Tue, 2007-10-09 at 15:29 -0500, Erik Anderson wrote:
> On 10/9/07, Hans Witvliet <[EMAIL PROTECTED]> wrote:
> > Hi all,
> >
> > Probably this is the wrong place to ask,
> > but is there an estimated time of arrival of the future?
> > i.e. TFOT--next g
is 2.6.22-14-386
Can I ignore this message, and is trunking working despite this warning?
The ztdummy module is not part of the zaptel ubuntu package, so it cannot
be loaded. I wanted to install from ubuntu packages for a change and not
compile it from source.
rgds,
Ha
When SOME (not all) people phone me (isdn-incoming) DTMF is not
recognized.
How come?
Either it works for a particular configuration, or it doesn't.
It doesn't make sense to me that it works sometimes...
Hans
___
--Bandwidth and Colocation P
That was my (mis)understanding as well. It seems that it is currently not
possible to compile the zaptel modules for a 2.6.22 linux kernel. For now
I will not use the trunking option.
Thanks,
Hans Feringa
> On Tue, 2007-11-06 at 18:30 +0100, Hans Feringa wrote:
>> I understood that
eshooting
>
> -jr
>
> On Nov 6, 2007 2:12 PM, Hans Witvliet < [EMAIL PROTECTED]> wrote:
> Hi all,
>
> Perhaps someone can give me a hint i the right direction...
>
> Sometimes dtmf is recognized, sometimes not.
>
On Wed, 2007-12-05 at 12:18 -0500, Alex Balashov wrote:
> Sam,
>
> Thank you for the suggestion. That is pretty much what I ended up doing
> for myself anyway; the real issue is standardising it and doing it on a
> mass scale for all users of a platform.
>
> -- Alex
>
> On Wed, 5 Dec 2007, Lut
a
separate machine(s). never got the pci forwarding from a hypervisor to a
dom-U properly working
Hans
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AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.a
On Mon, 2008-10-13 at 17:37 -0400, Steve Totaro wrote:
> I have done this. Why BRIs exist in the US is beyond me.
Much of the idea's behind ISDN are hopelesly outdated, except for one:
With POTS, the analogue/Digital conversion is done some miles away, in
the your local number exchange, and the d
On Wed, 2008-10-15 at 21:19 -0200, Rafael Puga wrote:
> Hi friends,
>
> I need a help to configure the TLS certificate chains to use with
> Asterisk for SIP TLS, can anyone help me with this, sending a link, a
> tutorial or something like that which explains how to generate and use
> CAs???
>
Yo
On Sat, 2008-10-25 at 11:54 -0600, Joseph L. Casale wrote:
> I need to increase reliability at an office as SIP/Internet provider outages
> are causing some issues.
> What would be the least expensive analogue card that people are using
> reliably?
>
If its for reliability, i wouldn't recommend
On Fri, 2008-10-31 at 10:54 +0100, randulo wrote:
> Morning!
>
> This may be the "day of the dead" in some regions, but we expect the
> usual lively discussion today at 9AM PDT, 11 Central, 12 Noon EDT, 4PM
> UK and Portugal, 5PM Paris, $deity-forsaken hour down under. This
> Sunday, I believe the
On Sun, 2008-12-21 at 21:00 +1100, Mikel Lindsaar wrote:
> Hello list,
>
>
> I am doing some work for a non profit group.
>
>
> As part of this, I am going to be putting in a 30 handset Asterisk
> solution. We are trying to keep the costs down as much as possible,
> as this job includes cabli
On Sat, 2008-12-20 at 19:27 +0200, Elliot Murdock wrote:
> Hello!
>
>
> What kind of sms text messaging capabilities does Asterisk have?
>
> I do not know very much about about SMS technology, but I am looking
> for the following features:
>
> 1. mobile SIP devices can send and receive SMS mess
On Sat, 2008-04-26 at 15:11 +0300, Kashif Naeem wrote:
> Hello All,
>
> A company has two requirements:
> 1) They are looking to develop its own CRM
> 2) Second thing is that they want to develop enhancements / new
> features in Asterisk like Thirdlane.
>
> What are your comments about technolo
On Wed, 2008-04-30 at 13:11 +0100, Dee Lowndes wrote:
> Hi All,
>
> I am trying to decide weather to move my asterisk setup on to a Xen
> setup or not. I do use transcoding, meetme and music on hold although in a
> purely sip scenario real lines are handled via cisco kit. Currently its a
> ded
On Thu, 2008-05-01 at 17:00 -0500, Tilghman Lesher wrote:
> We're about to do another batch of sounds, and I see by my word count that we
> have some extra time left over. So, suggestions will be entertained for
> additional prompts in English, Spanish, or French. The only rules are: 1) the
> pro
On Wed, 2008-05-07 at 11:44 +1000, Paul Hales wrote:
>
> Tomorrow night is the monthly Asterisk night...in melbourne
> (australia)...
>
> The usual stuff - get together, eat, show off tech toys.
>
> At the Pint on Punt, from 7pm.
>
> later,
>
> PaulH
>
Love to come, but as my b
On Wed, 2008-06-11 at 16:30 +0100, Ade Vickers wrote:
> Hi,
>
> I've been asked to spec up a small Asterisk system, which needs to:
> - Connect to ISDN2e (I'm thinking of using a B100P card here)
> - Connect to the POTS (A400P with 1 FXO)
> - Allow remote phones (thinking of an ETC 6050 utilis
On Thu, 2008-06-19 at 15:50 -0400, Paul Belanger wrote:
> List,
>
> Could anybody speak to the status of development in 1.6 branch? I
> know support for SIP over TCP is pretty new / experimental but it
> seems active development of it has slowed or stopped in recent months.
> Is that a correct s
On Fri, 2008-07-11 at 11:22 -0500, Tilghman Lesher wrote:
> On Friday 11 July 2008 09:17:37 Artie Gold wrote:
> > In updating to 1.4.21 recently, we've encountered a problem, when running
> > over a satellite connection (where the latency is considerable; a "regular"
> > internet connection did not
On Fri, 2008-07-11 at 18:37 +0200, Dave Cotton wrote:
> SIP wrote:
> > Joseph wrote:
> >> I need another Sipura 3K and the replacement I think is Linksys SPA3102.
> >> Any input on how reliable is it?
> >>
> >>
> > We have a few dozen subscribers using them at any given point in time. I
> > and
Before trying something impossible, and making a fool of my self, i
rather ask the list...
At my work i've got a single E1-test line, and all the project-leaders
are constantly fighting over the use of the line.
As it is mere the fact that they just need the E1 as a line, but not the
amount of tr
On Mon, 2007-03-05 at 12:54 -0500, James FitzGibbon wrote:
> The HP DL380 G5 (like many rack servers) has no AMP Mate-n-Lok
> connector available to attach to a card that needs more power than the
> PCI bus can provide, like the TDM400P when FXS modules are used. HP
> has confirmed that there is n
On Mon, 2007-04-02 at 22:12 -0400, Matthew Rubenstein wrote:
> What it means is that Flash memory cells wear out after a large number
> of read/write cycles, but not nearly as large as hard drives:
> http://en.wikipedia.org/wiki/Flash_rom#Limitations . So using Flash in
> place of RAM, even
On Tue, 2007-04-03 at 05:30 -0700, Jason Kim wrote:
> Is it exists?
>
If not, how could they have done this:
http://opensourcepbx.tmcnet.com/topics/applications/articles/5450-industry-forum-hails-successful-voip-over-asterisk-ipv6.htm
(But i'll guess it's not mainstream code yet...)
hw
--
pgp
On Wed, 2007-04-18 at 17:11 -0400, Dean Collins wrote:
> Hi guys,
>
> I know it’s a little off topic but……Wondering if you can help.
> My wife has been asked to find a writer to produce a story on “The
> dramatic ramifications of IPV6 on commercial businesses and how it
> will change the product d
On Sat, 2007-04-21 at 11:30 -0700, Ira wrote:
> At 02:31 PM 4/19/2007, you wrote:
> >Fridge (sending snmp traps if a dork leaves the door open ;)
>
> But will it tell you when the person who put in the box that's
> holding the door open now slammed the door on that box instead of
> putting it in
On Fri, 2007-04-20 at 00:43 +0200, Remco Post wrote:
> Hans Witvliet wrote:
>
> > The only obstacles currently, are the ISP's.
>
> Any decent ISP (eg. XS4All.nl) will give you an ipv6 address as well as
> an ipv4 address.
>
> > afaik, all dsl-mod
prise version? (just in case of, if one needs it)
Hans
--
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Registered linux user: 75761 (http://counter.li.org)
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On Mon, 2007-04-23 at 04:46 +0300, Tzafrir Cohen wrote:
> On Mon, Apr 23, 2007 at 01:49:12AM +0200, Hans Witvliet wrote:
> > Hi all,
> >
> > Just curious,
> >
> > Quite a while a go, i was checking for supported SW-platform.
> > AFAIR, it was RHES and SLES
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
> M
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
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 te
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 securi
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 an
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.
>
> Mic
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 some
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 ign
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:
>
>
> > What you can do (perhaps not the best solution...) is having one
> > asterisk server behind your firewall, serving all your local
> > sip-clients. And anoth
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 Sat, 2009-09-26 at 22:09 +, Jeff LaCoursiere wrote:
> On Sat, 26 Sep 2009, John A. Sullivan III wrote:
>
>
>
> > 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 (http://iscs
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 tu
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 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 ar
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:
> > http://wiki.kolmisoft.com/index.ph
On Fri, 2009-01-16 at 15:52 +, Julian Lyndon-Smith wrote:
> Can anyone who has used both comment on the pros and cons ? Need to buy
> about 30 of these, for a small company with limited IT support.
>
For evaluation a got a couple of different phones.
gxp2000, cheap, but it works, some peopl
O.P. said he's using SLE. You're talking about openSUSE.
Are you using the rpm's from the OBS?
The zaptel-rpm for 10.3 were containing the proper startup scripts.
I've got some suse machines running asterisk, but as soon as hw get's
involved, i'm
e
1.6.1-rc version, as video was/is one of the points on the 1.6.1-list.
However, skimming through the release-notes i fear that much is left on
the "todo-list".
Major point is still the video-multiplexing of all the individual
streams... (afaibt)
Hans
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. ;-)
How about .. dialplan.conf .;-)
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