On Tue, 2006-07-18 at 06:02 +1200, Matt Riddell (NZ) wrote:
:)
Which applications exist that have been disclaimed, well coded, are
patent unencumbered and are not accepted?
res_js for example, which in my experience on a more or less fair
comparison (the javascript dialplan has more error
On Tue, 2006-07-18 at 13:10 -0500, Brent Torrenga wrote:
Anyone notice that tf.voipmich.com (ENUM for US toll free service) will
connect you successfully, but then disconnect after what seems like 30
seconds or so? Anyone know what might be going on here? I googled the hell
out of voipmich and
On Sun, 2006-07-16 at 23:57 -0700, Martin Joseph wrote:
I think if you keep the older source in a separate directory, you can
always cd back to it and do a make clean, make, make install.
or if you are lazy, make takes multiple targets so you could do:
make clean all install
all on one like
On Mon, 2006-07-17 at 19:21 +1200, Matt Riddell (NZ) wrote:
It will sometimes tell you that there are modules inside
/var/lib/asterisk/modules which were not compiled for the version you
are compiling. If these are not asterisk-addons modules you will likely
need to remove them.
or modules
On Mon, 2006-07-17 at 15:17 +0200, olivier.taylor wrote:
email message attachment (where is the error?)
SELECT\ left(Customer.balance,instr(Customer.balance,'.')-1)\ FROM\
Customer\ Inner\ Join\ subscriber\ ON\ subscriber.customer_id\ =\
Customer.id\ WHERE\ subscriber.username\ =\
On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote:
Hello,
I'm recently having the problem where Asterisk just stops working.
The console gets disconnected and the process appears to die. I am
using Asterisk version 1.2.9.1. Anyone have any ideas on where I
should be looking for the cause
On Tue, 2006-07-11 at 09:08 -0700, Ira wrote:
At 04:59 AM 7/11/2006, you wrote:
I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec
apart most of the time and then sometimes for about 45 - 74 minutes
I have tried a reload and sip reload but neither bring the provider
back
On Tue, 2006-07-11 at 14:10 -0400, Rick Smith wrote:
teliax had a 2.5 hour outage today. I wouldn't call that short.
its all relative, nufone had a 30 day outage :P
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31
There is also the issue of origin of the caller. Not just
geographically, but which network provider they use, and to some degree
when. Most ISPs see higher traffic volumes when school gets out for the
day (abnout 3pm) continuing for a few hours, then gradually declining
until its later (11pm or
On Wed, 2006-07-12 at 06:40 +0800, Ronald Wiplinger wrote:
Dear NuFone,
Without misunderstanding I ask you again, please send the log file and
pay back my money!
Not following this request results in the assumption that NuFone is
cheating and I will post this info every hour on more
On Tue, 2006-07-11 at 20:16 -0400, C F wrote:
I find this hard to believe, half a truth is a whole lie. First you
just say the screwed you out $3k, not saying how, letting everyone
assume thru phone service, then you change the story, you lied before,
how do we know you are saying the truth
On Tue, 2006-07-11 at 20:51 -0400, C F wrote:
While I don't disagree with you, look at what my point was, just
accusing them for such without any documentation doesn't make sens.
He only brought that up after people started questioning it. So I
dunno. And lets face it, this is the internet
On Tue, 2006-07-11 at 21:44 -0400, Andrew D Kirch wrote:
Michael Workman wrote:
Very Simple.
I hired JerJer to Have a SER and Asterisk setup with Acounting...
JerJer told me to Talk to Shido6 and he would do it... He told me it
Would cost me $3000 and he do it.
He demanded the $
On Wed, 2006-07-12 at 00:41 -0400, Alexander Lopez wrote:
taken off line. Please respect the wishes of those that fund the list.
___
--Bandwidth and Colocation provided by Easynews.com --
easynews?
--
Trixter http://www.0xdecafbad.com Bret
On Wed, 2006-07-12 at 01:00 -0400, Alexander Lopez wrote:
Lists.digium.com
yeah easynews provides that. Thanks for being clear.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306 553058 US WA +1 360 207 0479
US
On Mon, 2006-07-10 at 07:34 -0500, Mike Bates wrote:
Are you talking about ZiPhone a USB device ?
Mike
zphone is phil zimmermans (creator of pgp) encrypted rtp system. Unlike
SRTP this does not rely on the server itself to provide the encryption.
It also lets you be reasonably assured
On Sun, 2006-07-09 at 10:22 +0200, Vincent Delporte wrote:
Still, considering the number of people having similar problems with those
cards, I was wondering what the problem is. Is it because the hardware, no
matter what is advertised, is actually not identical from card to card so
the
On Wed, 2006-07-05 at 12:18 +0200, Patrick wrote:
I read the page about the Skype API at
https://developer.skype.com/Docs/ApiDoc/Using_the_Skype_API_on_Linux
Not being a programmer, I wonder if it's possible to use the API and the
examples at the end of the page to come up with some way to
On Wed, 2006-07-05 at 16:01 -0400, Tigran Kocharyan wrote:
Patrick,
I think this is your answer:
https://www.nch.com.au/skypetosip/index.html
That oinly runs on windows, which isnt acceptable for some people.
Further I have yet to hear from anyone that has used it with more than 1
channel, has
it appears that scott has an autoresponder on the list saying he wil be
gone for the next 2 weeks, it also appears that he is responding to
himself, which is going to cause an exponential growth in the list
volume.
Infact it also appears that he will respond to this message as well,
which will
On Tue, 2006-07-04 at 23:10 -0400, David Beckerdite wrote:
Is there an archive for this list that can be searched? If so, could someone
tell me where it's located?
google works...
site:lists.digium.com asterisk-users your search query here
--
Trixter http://www.0xdecafbad.com Bret
On Tue, 2006-07-04 at 23:49 -0500, voiplist wrote:
Any way to monitor this? Send an email to admins? Something?
On 7/4/06, Thomas Kenyon [EMAIL PROTECTED] wrote:
voiplist wrote:
What happens when/if your Asterisk server is asked to handled more
g729 calls than it has licenses?
On Wed, 2006-07-05 at 08:32 +0300, Tzafrir Cohen wrote:
ln -s /dev/pts/20 /dev/tty9
I got a terminal for asterisk *g* and now I have
a colored CLI running :))
But how are you going to guarantee /dev/pts/26 will exist?
Specifically: what happens when you end your current ssh session?
On Sat, 2006-07-01 at 11:33 -0400, John Kington wrote:
I tried to get an update from NuFone but
Has anyone gotten their tollfree number ported
to another provider by NuFone? Should I just
forget it and move on?
Regards,
John
I have heard a lot of people have gotten em ported to
On Fri, 2006-06-30 at 13:31 +0400, Jean-Michel Hiver wrote:
Hi List
I have 10 separate SIP phones, and I wish to limit the simultaneous
available channels to 5 maximum for these. How would you go about it
without setting up a separate * box?
Cheers,
Jean-Michel.
you can limit it to
On Fri, 2006-06-30 at 10:44 +0100, Steve Langstaff wrote:
trixter aka Bret McDanel wrote:
Lastly, and probably the least effective, is you can watch channel usage
and when someone exceeds 5 run over to their desk and smack them with a
rotten fish.
http://www.voip-info.org/wiki
How many channels have you guys been able to get with this?
The only problem I have with this is that it takes skype and a soundcard
(virtual or otherwise) and the API is really executing commands on a
running skype process. In my opinion its not worth it for 1 concurrent
call per account.
I
On Wed, 2006-06-28 at 08:14 +, [EMAIL PROTECTED] wrote:
Well, look at it this way: if you get the working, you can buy one of those
tiny form-factor 386 boards with the 2 pcmcia slots and get a pcmcia soundcard
and a ethernet port. Run Linux off a CF card and have it setup to *only*
On Wed, 2006-06-28 at 13:15 +0300, Tzafrir Cohen wrote:
Since you can make a Skype account for free and
can (for right now) make US and Canada LD calls for free, I think the cost
and time to make them would be worth it. :) And if you figure out a good
price for them, people might even
On Wed, 2006-06-28 at 19:10 -0700, Mike Fedyk wrote:
Hi,
We have a page on the wiki just for phone reviews, but I think it needs
a bit of format change. Instead of individual reviews for each phone, I
think each person should review all phones they have worked with and
list the phones
On Tue, 2006-06-27 at 15:00 +0200, Morten Isaksen wrote:
Hi!
I have this setup:
PABX --ISDN30-- Asterisk 1 --SIP-- Asterisk 2 --ISDN30-- TELCO
Digium TE410P is used in both Asterisk 1 and 2.
When I set the CLIR bit on the PABX the Callerid / ANI is removed
somewhere between the
On Mon, 2006-06-26 at 13:16 -0400, Brian Capouch wrote:
It will be interesting to see how many standards get broken, and how
many proprietary hooks get thrown into the pot. The bean counters smell
some money, and their OS franchise is waning:
On Thu, 2006-06-22 at 13:16 +0200, olivier.taylor wrote:
sipsak is ok for that
Olivier
sipp.sf.net is also not a bad product. They both work slightly
differently so it depends on what exactly you need.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE
On Tue, 2006-06-20 at 10:49 -0400, William Piper wrote:
List,
Does anyone know how to add the dst Country to the CDR's via Macro
(preferably).
cdr(userinfo)?
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht NL +31 306
On Tue, 2006-06-20 at 11:14 -0400, William Piper wrote:
Thanks Bret, but how about an example or webpage?
I'm not finding anything on google about this command for asterisk.
What about AppendCDRUserField()... would this work?
that seems to be the same thing. the userfield lets you stick
On Sat, 2006-06-17 at 10:16 +0200, Florian Overkamp wrote:
There are ways to guesstimate MOS scores on a call by continuously
getting some decent statistics from the jitterbuffer. We've had an
intern do some work on this using IAXclient.
http://www.speakup.nl/en/opensource/jitterbuffer/
On Sat, 2006-06-17 at 12:52 +0200, Florian Overkamp wrote:
The work that you have done so far is a great step towards a product
that many people might find useful. In a nutshell the concept I am
thinking about is a tool that you drop onto your network and it will
monitor the data
On Sat, 2006-06-17 at 23:25 +0800, Steve Underwood wrote:
Calling MOS totally subjective is rather strange. Telephony only has to
meet subjective goals. In reality, MOS is pretty objective, as it is a
carefully controlled experiment across enough subjective individuals to
filter out a
On Sat, 2006-06-17 at 01:26 -0400, Daniel Salama wrote:
Is there any tool that can do LCR for Asterisk but also take into
account MOS scores?
Is it possible to automatically generate MOS scores on random calls
so as to keep an updated database on a per provider, per destination,
per
On Wed, 2006-06-14 at 19:58 -0400, Daniel Salama wrote:
Mainly GXP-2000 (with silence suppression off) and Eyebeam (with
Enable microphone noise reduction off)
its safe to ignore that too, it just means that asterisk doesnt support
a sip feature that your phone does and its telling you hey I
TRX Teleocmmunications the VoIP provider that pays you would like to
assist those that make asterisk better. To that end we are setting up a
program where the community itself can help double the bounty for all of
the outstanding code that is wanted but not yet present.
TRX will match any
On Wed, 2006-06-14 at 00:50 +0400, Jean-Michel Hiver wrote:
Actually i've done 50,000+ line dialplans using my Asterisk::LCR
dialplan generator, and asterisk has been just fine with it.
I have you beat, I have done over 500k when loading my country list that
I no longer maintain which is now
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
Ideally I would have liked the pap2 to have done the same as 'immediate'
when talking about fxo, capi, misdn, etc, but I couldn't get it to
automatically dial nothing. A '0' was the best I could do. If anyone
knows how to put it into
On Thu, 2006-06-08 at 16:49 -0400, Steven wrote:
My legacy PBX accepts CID number, but not name.
My old PRI vendor never sent the name, so there was never an issue.
I have wedged asterisk between the Legacy PBX and PSTN. PSTN - PRI -
asterisk - PRI - Legacy.
Any calls from asterisk
On Fri, 2006-06-09 at 01:52 +0100, Chris Bagnall wrote:
I think the issue for many people here is not the cost of the licence
itself, but the very frustrating lockdown to specific pieces of hardware
without any real reason.
I say without any real reason because anyone who doesn't care about
On Fri, 2006-06-09 at 04:49 +0300, Tzafrir Cohen wrote:
On Fri, 2006-06-09 at 01:52 +0100, Chris Bagnall wrote:
I say without any real reason because anyone who doesn't care about the
licencing of g729 has an easy alternative in the form of the downloadable
g729 binaries. They aren't
On Wed, 2006-06-07 at 07:55 -0700, [EMAIL PROTECTED] wrote:
For all the noise about this noone has mentioned one important thing.
We should be gratefull that we have access to G.729a in Asterisk,
whatever the mechanics of the licensing. It's obvious that its a pain
in the [EMAIL PROTECTED] for
On Wed, 2006-06-07 at 11:17 -0400, Ben Klang wrote:
On Monday 05 June 2006 15:41, Andrew Kohlsmith wrote:
The (current) problem is that the registration program does not ask which
ethernet card you wish to bind to, nor does it look at the Asterisk config
and use the MAC address of the
On Tue, 2006-06-06 at 11:37 +0600, [EMAIL PROTECTED] wrote:
try asterisk -rx 'show channels'
that is what I did try, yes I ommited the quotes in the email guess it
wasnt understood that it returns only the header and not any information
on what channels are in use nor any information on how
On Tue, 2006-06-06 at 15:17 +0800, Chen Fan wrote:
Hi,all.
There have any STUN spport for asterisk?
thanks,,,
where asterisk queries a stun server or where asterisk acts like a stun
server?
Becuase stun is totally self contained it would be silly (in my opinion
anyway) to have a stun
On Tue, 2006-06-06 at 16:12 +0800, Chen Fan wrote:
hi,
We need STUN client support for asterisk...
becasue the service provider only offer STUN interface,, so i can not
connect asterisk to their server
all stun does is resolve your external IP by sending data to a foreign
server
On Tue, 2006-06-06 at 15:41 -0400, Russell Handorf wrote:
Hello all,
I'm playing with app_flite, as I'm sure you've guessed. I have the
sources compiled and running, headers and libraries in their respective
places. I then compiled app_flite without any problems or errors. When I
try to
On Tue, 2006-06-06 at 13:09 -0700, John Todd wrote:
http://www.boingboing.net/2006/06/05/play_zork_by_phone.html
Let me preface this idea with one comment: I don't have the time to
do this - I don't even have time to eat these days. But someone out
there has the cycles to do this... and
On Tue, 2006-06-06 at 13:47 -0700, John Todd wrote:
While I love voice synthesis, I think that you'd gain legend status
only if you had Allison (or, I hesitate to say, some other voice
talent) do the dramatic readings. People respond so much better to
real recordings - I tend to use
On Mon, 2006-06-05 at 09:49 -0400, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 16:57, trixter aka Bret McDanel wrote:
but $10 only gets you one license, what if you are vonage sized and need
to support a million customers? What if you accept that you can settle
If you are Vonage
On Mon, 2006-06-05 at 10:00 -0400, Andrew Kohlsmith wrote:
On Saturday 03 June 2006 04:05, Sahil Gupta wrote:
We recently had around 60-80 licenses become useless because Digium
refused to renew the keys on that. That was a bit of money kissed
goodbye.
Ok, that's a great fairy tale.
On Mon, 2006-06-05 at 10:46 -0400, Paul wrote:
I really doubt that Digium would insist on the $10 fee for a quantity buyer.
no they do give some discount for quantity, people have mentioned that
when they bought a bunch. However I think they said it was close to
$8/license for 672 channels.
On Mon, 2006-06-05 at 12:01 -0400, Andrew Kohlsmith wrote:
On Monday 05 June 2006 11:03, trixter aka Bret McDanel wrote:
Again, 10k channels you'll have a half dozen MaxTNT boxes terminating
DS3s. Your fixed costs will already be significantly higher and that
little $10 license fee
On Mon, 2006-06-05 at 11:24 -0500, Moises Silva wrote:
Asterisk support the concept of configuration engine, this means
that you can write a configuration engine to get the data from
anywhere. The default configuration engine is text_file_engine, that
reads the configuration from text files.
On Mon, 2006-06-05 at 12:05 -0500, Kevin P. Fleming wrote:
I have proposed that a number of times internally, only to be told
(vehemently) that customers would never go for it. That includes responses
from our distributors and channel partners, among others. It would also
dramatically
On Mon, 2006-06-05 at 13:47 -0400, Matt Florell wrote:
What are the reasons that people/companies/manufacturers use G729
instead of comperable codecs like GSM or Speex?
Microsoft and Apple both support GSM in their software, and Speex is
the same compression ratio as G729 yet is BSD-like
On Tue, 2006-06-06 at 00:36 +0300, Tzafrir Cohen wrote:
On Mon, Jun 05, 2006 at 04:32:29PM -0400, Cory Andrews wrote:
Voiceage in Montreal is supposed to be working on an open source G.729A
codec, although it mentions only that they allow developers to freely use
their G.729(A) codec object
On Tue, 2006-06-06 at 08:41 +0800, Steve Underwood wrote:
Cory Andrews wrote:
Voiceage in Montreal is supposed to be working on an open source G.729A
codec, although it mentions only that they allow developers to freely use
their G.729(A) codec object code for non-commercial purposes.
has anyone else noticed what appears to be a threading issue in asterisk
1.2.9 (it broke sometime between 1.2.4 and 1.2.9) where if you have
about 50 calls and do
asterisk -rx show channels
it will display the header but nothing about channels, total calls,
active calls, etc.
--
Trixter
On Sat, 2006-06-03 at 04:01 -0400, Chris Mason (Lists) wrote:
I have no problem with paying Digium the $10 for G729 licenses, everyone
has to make money. It's the administration of the licenses that sucks. I
experiment with different hardware a lot, and make up demo machines to
install for
On Fri, 2006-06-02 at 12:12 -0400, Andrew Kohlsmith wrote:
The Intel g729 code is licensed for educational use ONLY. Commercial use is
forbidden without paying the patent holder. $10 a port won't break the bank
of any business with a shred of a hope of a chance of surviving, and you stay
On Sat, 2006-06-03 at 16:03 -0500, voiplist wrote:
Can someone tell me the size (or any other) limitations for the
extensions.conf?
We have managed to keep our file pretty small thanks to AGI but we are
about to setup a bunch of call restrictions based on area and country
code.
One line
On Sat, 2006-06-03 at 23:20 -0500, Erick Perez wrote:
As stated here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe
A Meetme room uses Ulaw as the audio codec, so if the other channels
use different codecs, then * will transcode.
Does the app_conference application
On Thu, 2006-06-01 at 21:16 -0700, Mike Fedyk wrote:
The way asterisk works is it abstracts concepts from protocol details.
sorta, it would be better if it actually abstracted everything so that
applications (modules prefixed with app_ generally) dont have to know
much, if anything, about a
On Fri, 2006-06-02 at 14:56 -0500, Erick Perez wrote:
If i get a 8XX number, my provider told me that they will send all the
calls he gets. But due to bandwidth and asterisk capacitiy in this
particular installation, the system is only able to handle 27 incoming
calls.
How in my dialplan do
On Fri, 2006-06-02 at 22:49 +0200, Alejandro Vargas wrote:
2006/6/2, Leon Sun [EMAIL PROTECTED]:
10$/channel
If you are connecting a device that uses g729 with another that don't
support it... let's say it uses gsm. Then you will use 2 channels, one
for encoding and one for decoding. Is
On Fri, 2006-06-02 at 14:18 -0700, Lee Howard wrote:
Andrew took issue with my initial sarcastic comment because this thread
involves the G.729 codec - but remember that if someone does ultimately
choose to obtain a license illegally that they're not cheating *Digium*
- rather, they're
On Fri, 2006-06-02 at 14:57 -0700, Lee Howard wrote:
As an example: Company X sells PCs with pirated copies of Windows that,
following proper and normal channels, they should have purchased from
Distributor Y. Microsoft sues Company X and wins a court judgment
against Company X.
On Fri, 2006-06-02 at 18:37 -0500, voiplist wrote:
Is there a list somewhere or a way to find the following:
1- All non US 48 area codes which can be dialed as 1+10
2- All strange area codes which are used for premium services such as
900-XXX-
3- Anything else that should be restricted
On Thu, 2006-06-01 at 11:18 +0200, Benjamin Stocker wrote:
At least you know to break this down into different parts, it still
amazes me how many people look at something as one big thing instead of
several smaller things that interrelate :)
you should have example config files that came with
On Wed, 2006-05-31 at 02:01 -0700, Michael Collins wrote:
Regarding my earlier post about labels and the 'n' priority:
The TFOT book covers the use of these. See the box on page 81 entitled
Unnumbered Priorities.
http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
And one of
On Wed, 2006-05-31 at 19:31 -0600, Joseph wrote:
Thanks for suggestion, but I'm not looking for software (spyware) type
service.
in that case how about www.dizzytel.com ?
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast IE +44 28 9099 6461DE +49 801 777 555 3402
Utrecht
On Tue, 2006-05-30 at 08:07 +0200, Attilla de Groot wrote:
It may have been 2 years since I worked with Debian on production
systems, but in my experience there are alot of unstable packages in
unstable. So it's a bad advice to run unstable on production systems.
the debian stable, testing,
On Tue, 2006-05-30 at 16:03 -0400, Matt Roth wrote:
mpg123 has the same problem with zombie processes as you were
experiencing with MadPlay. For a scalable system, native MOH is the way
to go. As per Kevin Fleming, it only introduces a slight memory
overhead. mpg123 consumes CPU cycles
On Wed, 2006-05-31 at 12:09 +1000, Peter J Dean wrote:
( cd asterisk; make clean ; make )This didn't compile ok,
and outputs the following error.
just a guess but did the headers change for libpri and not get installed
to the same location as before, as a result you are using
On Sun, 2006-05-28 at 23:41 -0400, Steve Totaro wrote:
Henry J. Cobb wrote:
to increase the security for remote extensions I would like to limit a
sip-peer to a specific MAC address. Is it possible to hook into the
authentication mechanism in asterisk and allow/deny incoming
On Sat, 2006-05-27 at 12:48 -0300, Hermann Wecke wrote:
Carlos Chavez wrote:
Now that Nufone is dead, what are other providers of 800 numbers that
work with Asterisk?
You may also consider Asterlink. I'm a new client there, their support
is a little slow, sometimes irresponsive (you
I was wondering if anyone was going to cluecon http://www.cluecon.com
I would like to start by saying I am not affiliated with cluecon, or anyone
speaking there. I just think this conference sounds good.
For those that dont know its a telephony conference in chicago august 1-3. It
will talk
On 3/29/06, Matt [EMAIL PROTECTED] wrote:
Hi,
Did anyone hear about a recent ruling which makes it illegal to have
caller-id set to anything except what is on the account of the user?
IE... If your name is Joe Smith you can't have Mary Smith set as
the caller-id name, unless mary
On Mon, 2006-03-20 at 09:32 +, David Waugh wrote:
NOTE: This is my first shell script so I'm sure it can be improved!
noted, in that spirit see notes below ...
***
[EMAIL PROTECTED] INBOX]# more /etc/asterisk/voicemail-clean
cd
On Thu, 2006-03-02 at 17:51 +0100, Francesco Peeters (Asterisk) wrote:
Just in my Inbox:
From the makers of Voipbuster: http://www.internetcalls.com
Over 40 FREE destinations, PLUS free VoipIn number AND Call Forwarding!
Finerea has sipdiscount.com which also is offering the same deal. it
On Thu, 2006-02-23 at 08:09 +, Lee Archer wrote:
I spent a days or two on this and in the end did
Musiconhold.conf
[livestream1]
mode=custom
application=/usr/local/bin/mpg123 -s --mono -y -f 8192 -r 8000 -@
/etc/asterisk/stream.playlist
Then in stream.playlist I just put the links
On Thu, 2006-02-23 at 16:46 +0800, Nathan Alberti wrote:
Is there a way to have extensions automatically created for
registered sip users ?
in sip.conf
regcontext=sipregistrations
that adds them to sipregistrations, you can make that anything you want
however I am willing to bet there might
On Sun, 2006-02-19 at 20:30 -0500, Doug Lytle wrote:
I have the following in my dialplan, counts the number of loops and when
it hits greater then 5, exit. It works, but errors initially with,
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or
tolken; Input: +1.
Could
On Sun, 2006-02-19 at 21:05 -0500, Doug Lytle wrote:
trixter aka Bret McDanel wrote:
Could somebody tell me why?
is count defined before it tries to do count + 1?
No it isn't, thank you for the clue. I'll define it.
since you have had a little time to play
http://www.trxtel.com/index.php?page=Tollfree_Termination
This is a free service, I am not selling anything with this service. I
just thought that individuals that read this list may enjoy getting
tollfree access free this way (yet another way) given that it lets you
send your caller id and some
On Fri, 2006-02-17 at 14:32 +0100, Alejandro Vargas wrote:
2006/2/17, adibar [EMAIL PROTECTED]:
I would sugest, that you just register without balancing your
account. Than use the supplied username/password and it will
work. I doubt that the test/test works.
Thanks. This worked. I
I am curious if anyone has had problems trunking iax2 with 100+
concurrent calls. I am planning on testing this out tomorrow, however I
wanted to know if anyone else has had a problem with this prior to my
test so I know what to look for if anything is known and what
resolutions have been found
On Thu, 2006-02-16 at 13:38 +0200, Zoa wrote:
A long time ago i tried to make one big iax2 trunk for one of my
customers, i soon changed this to several small trunks. (bandwith doesnt
rise all that much if you use 2 trunks instead of 1.) Asterisk didnt
seem to like my big trunk very much (i
On Thu, 2006-02-16 at 14:04 +0200, Zoa wrote:
I think, but am not sure, that with a lot of calls inside the trunk,
some calls seemed to go suddenly go outside of the trunk in one or more
directions, bursts of error messages appeared on the cli etc.
i didnt investigate it a lot more, my
On Thu, 2006-02-16 at 14:29 +0200, Zoa wrote:
The trunks were made to be maximum 60 simultaneous channels iirc.
I doubt seriously you will be able to do 600 simultaneous on any system.
(with or without trunking). (at least out of the box).
Zoa
At 100 with g.729 its running 95% idle, in
On Thu, 2006-02-16 at 14:54 +0200, Zoa wrote:
When you have a lot of calls, try doing a show channels and iax2 trunk
debug. (they are killers)
Zoa
not having trunks set up yet, I dont do the latter but I do the former
all the time. Mostly becuase this is a new server and I wanted to make
On Thu, 2006-02-16 at 14:58 +0200, yusuf wrote:
also doing IAX2 trunking. What do yuo mean you dont run asterisk out of
the box. Also want to know what is you bandwith usage for 100 calls and
g729
I run a modified version of asterisk. There are a few things that I
felt needed to be added,
On Thu, 2006-02-16 at 05:46 -0800, jonny hashem wrote:
HI:
what is command on console to know how many calls are
sending in the same time?
I will guess 'show channels'
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479
has anyone experienced a problem where RTP audio cuts out when doing
30-40 concurrent channels via sip?
The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel -
not even a timing source)
The box has plenty of bandwidth, when a call to the same box is iax2 it
works, but when its
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