Is there some way my uplink can tell my * the price of a call, either per
timeunit in the conversation at start of the call, or the total cost at the
end of the call?
I'd like to pass the bill on to the extensions.
Leif
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I search a application pop-up for the call. When I receve a call, I want a pop-up menu. Where I can found this application open source? Thanks Fabio
Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___
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Hi,
I used Aterisk 1.0.7 Debian package and it is now running. I
noticed that when I used X-lite and GSM codec is enabled Aterisk
stops and says Bus Error. Here is the message i got:
*CLI> -- Executing VoiceMailMain("SIP/mydomain.com-0011a700",
"810020") in new stack
-- Playi
A stupid question, but is it possible to use the PauseQueueMember
function with AgendLogin?
Whenever I use AgendLogin, I have a connection and the agend cannot dial
another extension to pause.
Creating a new line does not help, because the server returns 403
(forbidden).
If I use AgendCallbackLo
On Wed, 2005-11-16 at 19:53 +1300, Matt Riddell wrote:
> Damn you're fast LOL!
>
> :D
>
Its make up for earlier when it was taking > 1 hour to see my posts :P
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516
On 11/16/05, Rafael R. GV <[EMAIL PROTECTED]> wrote:
> issue related to cdr´s: I can see cdr records in 'call' table from mysql
> console but they don't appear in a2bill web interface, I only can see
> records of current date ...not sure if its a mysql 4.x incompatibility> so now I´ve installed po
I got TDM card with 4 FXO ports. But I need to dial out? How I can do
this? Is it possible? Please help
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mail
trixter aka Bret McDanel wrote:
> He was making a comment in reference to what I said. The blurb that I
> said included information on a specific project implementation of this,
> which the current licensing structure wont allow for, as such I legally
> cant license it unless I change my plans. T
Are you doing agent recording, or recording through the dialplan ?
Julian
Kelvin Williams wrote:
I have an ongoing problem and do not know where to begin troubleshooting it.
We run a helpdesk, and call recording is extremely important. But we have
found that calls are recorded at random. We rece
> issue related to cdr´s: I can see cdr records in 'call' table from mysql> console but they don't appear in a2bill web interface, I only can see
> records of current date ...not sure if its a mysql 4.x incompatibility> so now I´ve installed postgres to verify, let me now if someone has same> prob
Amaury BOSSE wrote:
> Hi all,
>
>
>
> I am trying to install an AVM Fritz card USB v2.1 on my Asterisk Box.
Did you remember to remove HiSAX?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoi
Chuck Bunn wrote:
> Hi,
>
> I did something stupid - I deleted the Asterisk log files instead of
> copying empty information into the file (I know dumb) to clear out the
> data. I recreated the files with the same rights as before (644) using
> the asterisk user and group but asterisk does not app
On Wed, 2005-11-16 at 19:32 +1300, Matt Riddell wrote:
> > http://www.sipro.com/news.php
> > MONTREAL, CANADA, July 6, 2005. The G.729 Consortium today announced
> > that it has changed its licensing policy and will now offer licenses to
> > use the G.729 patented technology to end-product manufact
Allison Smith wrote:
> Asterisk Community Members:
>
> What a day we've all had! OK -- deep breath -- relax, and let's take a
> look at the situation calmly and rationally.
Beautiful!
We were under that assumption, but your mail really leaves no room for
speculation. Thanks for your ongoing su
trixter aka Bret McDanel wrote:
> On Tue, 2005-11-15 at 23:27 -0600, Kevin P. Fleming wrote:
>
>>The ITU is not involved in licensing or patent indemnification; they are
>>non-profit standards body. The G.729 patent holders have given Sipro the
>>task of managing their patent portfolio licensing
Sergey Okhapkin wrote:
> Already supported (simple patch exists).
> http://bugs.digium.com/view.php?id=5374
Which? It's already supported in Asterisk or you can patch Asterisk to add it?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (
Will ask them to check the speaker volumes.
Not sure if you meant outside of my case, but in my case it's less than 15
ms.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: <[EMAIL PROTECTED]>
To:
Sent: Monday, November 14, 2005 9:
Jeremiah,
I’m glad to see that someone have
working this schema, really I followed the steps mentioned in the voip-info wiki, but without luck.
I see that the TNT is registered by
Asterisk
*CLI> sip show peers
Name/username Host
Dyn Nat ACL Mask
Port
On Tue, 2005-11-15 at 23:27 -0600, Kevin P. Fleming wrote:
> The ITU is not involved in licensing or patent indemnification; they are
> non-profit standards body. The G.729 patent holders have given Sipro the
> task of managing their patent portfolio licensing, so that is who you
> would need to
Asterisk Community Members:
What a day we've all had! OK -- deep breath --
relax, and let's take a look at the situation calmly and
rationally.
I -- as the voice of Asterisk -- am really out of
the loop gossip-wise. I'm not even officially on the Asterisk-Users mailing
list. Who said wha
Were you able to compile txfax and fxfax? It compiles but at asterisk
loading time I get errors about not find resources.
Did yours load ok?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|George Vagenas
|Sent: Tuesday, November 15, 2005 8:23 PM
|T
You can pause a queue member using the PauseQueueMember function.
On 11/16/05, Marcus Deluigi (intern) <[EMAIL PROTECTED]> wrote:
>
> Hi!
>
> Is it possible for an agent (member of a queue) to set its status to
> "not ready", e.g. if he has to do some work after a call? And is it
> possible to re
Hi!
Is it possible for an agent (member of a queue) to set its status to
"not ready", e.g. if he has to do some work after a call? And is it
possible to record that time?
I mean except of hanging up the phone and logging in to the queue again
...
Greetings,
Marcus
__
Hi friends,
I want to change the standard 5060 sip port to our any
defined port. i made some change in sip.conf but it is
not working, I have 2 softphone which are able to
register with 81 port but the any kind of hardphone is
not able to register using 81 port.
here is my sip.conf configuration
trixter aka Bret McDanel wrote:
Now for the ITUs site, I found the data set that has the '3
diskettes' (their words) in the zip file (pdf or word) however that is
82 swiss francs, does anyone have a copy that doesnt require such
payment (legally of course)?
The license under which you obtain t
trixter aka Bret McDanel wrote:
If you know where to look there is another option out there that doesnt
use either method, but I have doubts about how legal that one is, so I
will not comment on that.
Can someone give me some pointers for this?
smime.p7s
Description: S/MIME Cryptographic S
Hi ! :)
I just installed Asterisk, Zaptel and mISDN on a linux box based on a
Fedora Core 4.
When I start asterisk, it says :
Nov 16 06:10:02 ERROR[27460]: chan_misdn.c:3388 load_module: Unable to
initialize mISDN
Nov 16 06:10:02 WARNING[27460]: loader.c:345 ast_load_resource:
chan_misdn.so: load
[EMAIL PROTECTED] wrote on 11/15/2005
06:42:40 PM:
> That being said, and as I mentioned earlier, your cheapest choice
is
> to go to eBay and search for X100P.
Here's a question: why are you building your
hobby box? To gain practical experience? Then forget the X100:
it's like learning Win
On Wed, 2005-11-16 at 15:15 +1100, David Uzzell wrote:
> Ok I will give you that.
>
> If it is not enforcable in NZ thats great. Get a copy of the code and
> build your own codec. If you want to use digium's codec then you have to
> pay the lic fee even if the patent is not enforcable for you, tha
Makes sense. I will be reworking the network cabling tomorrow and see
how that behaves. Today I changed 2 workstations to a dedicated
wireless network to see if they also experience call drops as
frequent as the wired stations.
Thanks,
Waldo
On Nov 15, 2005, at 10:12 PM, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
> Millions of dollars in hardware sales? I admit I don't know what there
> sales level is, but I doubt that it's 1 mil at this stage in the game.
> And, if it were, that has nothing to do with profit. You can make 1 mil
> in sales in a year but still walk away with a net
I wouldn't believe 10mil/yr. Maybe if they had other non-asterisk
products, but that just does not seem reasonable if you look at asterisk
at this stage. Besides, are they a public company? Are they required to
report to the SEC? That makes a big difference. I mean realistically,
a 900 dollar
trixter aka Bret McDanel wrote:
> On Wed, 2005-11-16 at 13:21 +1100, David Uzzell wrote:
>
>>trixter aka Bret McDanel wrote:
>>
>>>On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote:
>>>
>>>
As far as I was aware a license was only required in contries that had
software
p
On Wed, 2005-11-16 at 13:21 +1100, David Uzzell wrote:
> trixter aka Bret McDanel wrote:
> > On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote:
> >
> >>As far as I was aware a license was only required in contries that had
> >>software
> >>patents, I know that there arnt here so I a
Maybe this will open your eyes :)
This article says Digium has sales of 10 million per year.
http://news.com.com/Is+the+telephone+industry+ready+for+open+source/2008-108
2_3-5737703.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED
Michael Welter wrote:
Nov 15 20:09:15 NOTICE[27290]: chan_zap.c:7395 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
I ran top during that time, and there was no significant cpu usage.
Probably interrupt starvation... are there any interrupts being shared,
or does
Their tdm24 prices are lower, but many others like channel banks and
te405 are higher than voipsupply.
Just shop around... Sorry though, I have not dealt with them before.
Voipsupply.com has always been good with me...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTEC
> Philip Edelbrock wrote:
>
> >
> > On Nov 15, 2005, at 5:40 PM, Logan wrote:
> >
> >>>
> >> As stupid as this may seem ::cough::, how do you test to see if
> >> there is voltage on the phone port? Would you plug in a phone that
> >> doesn't require a AC power and runs off the voltage from the ph
I used sftp to move the RC2 files to my * server. During the whole time
of the transfer, asterisk continuously reported:
Nov 15 20:09:15 NOTICE[27290]: chan_zap.c:7395 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
I ran top during that time, and there was no sign
What kind of switches? I would suggest go gigabit between your servers,
their switch, and the call center floor. And, use managed switches if
you are over 20 total stations. Could be someone's got a p2p and it is
killing you. Maybe unlikely, but quite possible. Use some good 3com
switches too,
> If using CCM >= 4.0, using SIP trunks will alleviate a lot of
headaches.
True, if you don't need any codec other that G711. According to
Cisco's docs (which could be wrong), SIP trunking in 4.X only
supports G711.
In any case I wanted the archives to have correct information
about CCM/H323/MTP
Why not have set lines 3-6 as separate sip registrations, and have
asterisk ring multiple phones?
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sergio
Chersovani
Sent: Tuesday, November 15, 2005 9:25 AM
To: Asterisk Users Mailing List - Non-Commerc
Millions of dollars in hardware sales? I admit I don't know what there
sales level is, but I doubt that it's 1 mil at this stage in the game.
And, if it were, that has nothing to do with profit. You can make 1 mil
in sales in a year but still walk away with a net loss.
Without the community wher
Andrew Kohlsmith wrote:
On Tuesday 15 November 2005 08:34, don vanfossen wrote:
Anyoen else getting multiple copies of each email now?
What? :-)
(yes there must be something funny in the coffee this morning)
-A.
You're not suppose to put that in the coffee... ;)
_
On Nov 15, 2005, at 9:29 PM, Logan wrote:
Philip Edelbrock wrote:
On Nov 15, 2005, at 5:40 PM, Logan wrote:
As stupid as this may seem ::cough::, how do you test to see if
there is voltage on the phone port? Would you plug in a phone
that doesn't require a AC power and runs off the volt
Unless, if this is what you mean, can you use one cat5 wire to run two
home runs? There are these adaptors that allow two signals by
merging/splitting 1&2,3&6 as the first feed, and 3&4,7&8 as the second.
This would work, but don't plan on going gigabit. Personally I prefer a
hub or switch, but
Philip Edelbrock wrote:
On Nov 15, 2005, at 5:40 PM, Logan wrote:
As stupid as this may seem ::cough::, how do you test to see if
there is voltage on the phone port? Would you plug in a phone that
doesn't require a AC power and runs off the voltage from the phone
line?
Thanks though.
Anton Krall wrote:
Guys. Has anybody been able to compile spandsp-0.0.2pre21c against 1.2rc2?
Seems spandsp-0.0.2pre21c is broken. :(
Compiles great against 1.2rc1 but no luck so far with rc2.
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Hans Witvliet wrote:
Hi all,
any experience (pos or neg) with either
a) Dell 1425
b) Acer Altos R510
Contemplating on buying a bladeserver for Firewall & Asterisk...
HtH, Hans
___
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Asterisk-Use
trixter aka Bret McDanel wrote:
> On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote:
>
>>As far as I was aware a license was only required in contries that had
>>software
>>patents, I know that there arnt here so I am just seeking clarification if
>>thats
>>all there is to it.
>
Juan Jose Comellas wrote:
Has anybody ever used the TxFAX application to send a fax to RxFAX on another
Asterisk installation. I'm trying to do just that and both apps remain
blocked in the ast_waitfor_nandfds() function without transmitting anything.
I am calling TxFAX with the 'caller' parame
On Tue, 2005-11-15 at 20:00 -0600, Brent August Torrenga wrote:
> When dialing in after hours callers get to use the directory. I know
> that if you put "h" or "H" with a Dial() command you get the behavior of
> being able to terminate a call by pressing *. However, nowhere in the
> entire extensio
When dialing in after hours callers get to use the directory. I know
that if you put "h" or "H" with a Dial() command you get the behavior of
being able to terminate a call by pressing *. However, nowhere in the
entire extensions.conf does there appear the "h" or "H" option, so I
know it is not tha
Esto lo hemos estado detectando en la ciudad de Guadalajara
On 11/15/05, Servers-R-Us <[EMAIL PROTECTED]> wrote:
Hola Alvaro,En qué parte de México tienes * y desde qué partes de México te conectas?Nosotros tenemos clientes en BCS, Oaxaca, Puebla, Morelos y el DF que se
conecan con Infinitum y no
I have an ongoing problem and do not know where to begin
troubleshooting it. We run a helpdesk, and call recording is
extremely important. But we have found that calls are recorded at
random. We receive the call via our toll-free number over an IAX
connection. The call is then either handled by
Amir,
> 1. What hardware do I need for the server to accept incoming and
> outgoing analog calls.
http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P
With at least one FXO module
> 2. What books, guides or companies or individuals can help me >setup.
a- http://s
Hello,
My phone's VMWI (Visual Message Waiting Indicator) is able to detect SDT
message signal.
But how I would configure asterisk to send SDT message signal to a certain
extension?
Thanks,
___
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I'm testing Asterisk 1.2. I read the UPGRADE.txt and followed the
instructions. I had to change a couple caller-id instructions, but that
was about it.
Then I noticed that voice mail wasn't working. It turns out my previous
config was calling VoiceMail2 instead of VoiceMail. That was easy to
fix,
Kenneth Shaw wrote:
Currently I have automon enabled, but I have absolutely no idea how to
get it to work. I have the latest CVS HEAD release.
This was just talked about in Mantis... set DYNAMIC_FEATURES in the
dialplan to include the features that you want enabled.
___
In AAH create a DID using the number 719705 and direct it to
ring wherever you wish it to (extension.. extension group etc.)
- Original Message -
From:
Cristian Paun
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 15, 2005
21:08
Subject: [Asterisk
If using CCM >= 4.0, using SIP trunks will alleviate a lot of headaches.
On Tue, 2005-11-15 at 16:33 -0800, Dan Austin wrote:
> I posted a couple weeks back about our experiences with H323 trunks on
> CCM.
> As of version 4.0, the Cisco documents state that a 3rd party H323
> gateway
> requires
Actually, that's not specifically true - a hub of theoretically any size
could be made just by carefully and accurately connecting together the
tx and rx pairs of a bunch of rj45 cables together. I don't think the
wire gauge would allow for more than a couple nodes though. I've got a
bud who
On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote:
> As far as I was aware a license was only required in contries that had
> software
> patents, I know that there arnt here so I am just seeking clarification if
> thats
> all there is to it.
This issue was beaten to death before,
Title: Cisco Call Manager and H323 trunk correction (MTP)
I posted a couple weeks back about our experiences with H323 trunks on CCM.
As of version 4.0, the Cisco documents state that a 3rd party H323 gateway
requires a Media Termination Point..
At the time I said that I have Asterisk wor
I just had that problem... I had to enter an extension with my fwd # as
the extension under the context after @ and tell it what local
extension to ring in extensions.conf
;free world dialup incomming call
exten => 720727,1,Dial(IAX2/shawn, 30)
Cristian Paun wrote:
I have an
As far as I was aware a license was only required in contries that had software
patents, I know that there arnt here so I am just seeking clarification if thats
all there is to it.
[EMAIL PROTECTED] wrote:
Yes. The G.729.x source code (in C) can be downloaded for free from
ITU-T as the spec ac
You do not wish to help me, I have " Internet
Telephony Gateway " from it I can to call on
asterisk and to type on local numbers from ports
FXS and FXO (an example: voicemail
the problem works perfectly on all inquiries), but
that asterisk cannot transfer dtmf in
ports FXS and FXO when I w
On Tue, 2005-11-15 at 19:01 -0500, Cory Andrews wrote:
> Rafael - To generate PDF documents, you may need to install Adobe PDF
> Writer, or the full version of Acrobat that includes the reader and
> writer. Just a guess.
Wouldnt an open source solution that is basically just libraries work
bett
Rafael - To generate PDF documents, you may need to install Adobe PDF
Writer, or the full version of Acrobat that includes the reader and
writer. Just a guess.
Cory Andrews
Senior Partner
+++
VOIPSupply.com
454 Sonwil Drive
Buffalo, NY 14225
+++
voice - 716.630.1555 X22
email -
Rafael R. GV wrote:
Hello
1.- I am testing a2billing in a SER-Asterisk implementation but using
Mysql versions 4.1.15 and 5.0.15 because I want to integrate its cdr
with some tables from Ser database, the a2billing-mysql-schema does not
work properly in mysql-5 and in 4.1.15 works well but no
On Nov 15, 2005, at 6:16 PM, Logan wrote:
[snip]
I am trying to set up a small hobby box on Debian Linux to play
around with. This will in no way be in a production evironment or
even a semi-production environment. Asterisk will be installed on
my personal Linux box. I have a generic Soft5
Already supported (simple patch exists). http://bugs.digium.com/view.php?id=5374
On Wed, 2005-11-16 at 12:32 +1300, Matt Riddell wrote:
Asterisk guy wrote:
> dropping extra frame of G.729 since we already have a VAD frame at the end-
Turn off VAD, it is not supported by Asterisk.
_
Mark Quitoriano wrote:
> you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9?
Yes, you can read the upgrade.txt file inside the RC2 distribution for
information on the required changes.
--
Cheers,
Matt Riddell
___
http://www.sine
Logan wrote:
I was wondering if it was feasable to istall
Asterisk on this box and have that modem (or whatever modem) with a
regular telephone wired to the "Phone" port.
I'm a bit of a noob, also, but I don't think the Phone port on those
cards are real FXS ports. I.e., I think they just co
Hello
1.- I am testing a2billing in a SER-Asterisk implementation but using
Mysql versions 4.1.15 and 5.0.15 because I want to integrate its cdr
with some tables from Ser database, the a2billing-mysql-schema does not
work properly in mysql-5 and in 4.1.15 works well but now I found this
issue rela
Asterisk guy wrote:
> dropping extra frame of G.729 since we already have a VAD frame at the
> end-
Turn off VAD, it is not supported by Asterisk.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://f
Hi...
I was going to clear a couple of things up because it seems that my
message wasn't very clear--it teaches me not to post to a mailing list
sick and half-sleep. Here we go...
I am trying to set up a small hobby box on Debian Linux to play around
with. This will in no way be in a product
Yes. The G.729.x source code (in C) can be downloaded for free from
ITU-T as the spec actually are in C code. But, even if you use this raw,
non-tuned version you will need to pay a license fee to ITU-T per channel.
The same for G.723.x
The version that Digium offer is optimized, meaning that
Done. Done. Done :)
http://bugs.digium.com/view.php?id=5762
Julian.
BJ Weschke wrote:
On 11/15/05, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
Bad form replying to my own post, but I'm getting my ass chewed here
because we need to listen to a call I think has been trashed ;)
I've now trie
On Tue, 2005-11-15 at 16:00 -0500, Jerry Geis wrote:
> Anyone doing anything with bluetooth headsets?
>
> I'd like to use one with asterisk. If that means through a softphone OR
> directly
> connected to asterisk - either way is fine.
http://mundy.org/blog/index.php?p=78
on automatic proximity d
Andrew Kohlsmith wrote:
On Tuesday 15 November 2005 13:33, Kevin Hanson wrote:
Can you explain the earlier post in this thread that seems to imply that
Digium support thinks differently?
Yes. Digium support was misinformed. Kevin Fleming of Digium replied to you
about that in this
Richard Malcolm-Smith wrote:
> Do I need licenses to use the codec in New Zealand?
Yes.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://ww
On 11/15/05, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
> Bad form replying to my own post, but I'm getting my ass chewed here
> because we need to listen to a call I think has been trashed ;)
>
> I've now tried using the Monitor command:
>
> 1) Incoming Call Answered By Extension A
> 2) Conver
Bad form replying to my own post, but I'm getting my ass chewed here
because we need to listen to a call I think has been trashed ;)
I've now tried using the Monitor command:
1) Incoming Call Answered By Extension A
2) Conversation between caller and A
3) A decides to transfer call to B
4) A ta
The zttest results should be 100% ideally, but everything should work ok
with >= 99.98%. here's a great discussion:
http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting
Maybe try what they recommend with hdparm, specifically unmasking
interrupts and setting udma2 mode. It woul
On Tue, 15 Nov 2005, Mailing List wrote:
- Original Message - From: "Jason Pyeron" <[EMAIL PROTECTED]>
To:
Sent: Tuesday, November 15, 2005 11:09 AM
Subject: [Asterisk-Users] unexpected debug output from console
Why am I getting debug from server<-->sip.broadvoice.com on the console
Hey juan,
Well I have 3 405p cards in one machine a p4 2.4 with a gig of ram. Running
good all 12x t1's are connected to channel banks.
Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: carlos at race.com
-Original Mess
I just installed asterisk 1.2 rc2 and ran a 'make samples' and asterisk
starts just fine with no errors in the logs. However, if I issue
a reload I get the following:
Nov 15 17:08:22 WARNING[27009] res_musiconhold.c: Music on Hold class 'default' already exists
It is almost like the previous mus
Do I need licenses to use the codec in New Zealand?
smime.p7s
Description: S/MIME Cryptographic Signature
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/m
I am trying to set up the following:
Asterisk Server (1.0.9-stable) with BriStuff (0.2.0.RC8j) and florz patch
2 HFC cards (cologne chip)
I want to connect 1 to the ISDN net (TE) and to the other connect the old
Siemens Gigaset 4135 (NT mode)
I have tried loading zaphfc.0 with modes=1, modes=2 a
I used to use ConText, but now I prefer Notepad++.
Both are free and for Windows.
They both let you easily edit Unix formatted text files.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - -
That was on the machine with the TE410P.
- Waldo
On Nov 15, 2005, at 4:00 PM, asterisk wrote:
Run zttest on the machine with the TE410P.
I'm attaching the output of zttest (zttest.txt) and concurrently
running vmstat 1 (vmstat.txt). Please see if you can make something
out of it because I d
[EMAIL PROTECTED] wrote on 11/15/2005
02:53:54 PM:
> Ending last year I used\sold several hundred of product#: FM-INL92SW.
>
> Google for it...you'll find some for cheap.
Along those lines: are there drivers for the
X100/X101 that allow it to act as a normal v.92 modem, even if it's just
under
I like ultraedit for an editor... has built in ftp and translates dos to
unix and unix to dos nice editing features (www.ultraedit.com) the ftp
features allow to edit and save from ftp site. and you can browse as well.
Will Glass-Husain wrote:
On Windows, I really like TextPad (shareware - www
trixter aka Bret McDanel wrote:
>On Tue, 2005-11-15 at 12:57 -0700, Colin Anderson wrote:
>
>
>>* I like edlin cause its old school :P
>>
>>copy con extensions.conf ^z
>>
>>Better not screw up!
>>
>>
>
>"command not found" :P
>
>
Get modern!
Just set up an email-based template system lik
The only time I've ever received that message was when I was not receiving
CallerID on the line (though this was with a Rhino channel bank).
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Tuesday, November 15, 20
Run zttest on the machine with the TE410P.
> I'm attaching the output of zttest (zttest.txt) and concurrently
> running vmstat 1 (vmstat.txt). Please see if you can make something
> out of it because I don't really know what the numbers mean.
>
> Thanks,
> Waldo
>
--
My understanding is that this should work as long as you load the
modules correctly. I will be interested in your results.
Jan
Juan Manuel Coronado Z. wrote:
Hi everyone,
I would like to know who of you out there have more than 3 Digium
wildcards working on the same server, and have no proble
Anyone doing anything with bluetooth headsets?
I'd like to use one with asterisk. If that means through a softphone OR
directly
connected to asterisk - either way is fine.
I found the chan_bluetooth app and downloaded it but it does not compile
and have not been successful in reaching the auth
On Tue, 2005-11-15 at 12:57 -0700, Colin Anderson wrote:
> * I like edlin cause its old school :P
>
> copy con extensions.conf ^z
>
> Better not screw up!
"command not found" :P
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360
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