Do you have a Zaptel device in this machine?
Jeremy McNamara
Jordan Peterson wrote:
Is this me or what?
-- Playing 'demo-congrats'
-- Executing MeetMe("H323:996", "") in new stack
-- Playing 'conf-getconfno'
== Parsing '/etc/asterisk/meetme.conf': Found
WARNING[17425]: File app_me
Is this me or what?
-- Playing 'demo-congrats'
-- Executing MeetMe("H323:996", "") in new stack
-- Playing 'conf-getconfno'
== Parsing '/etc/asterisk/meetme.conf': Found
WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open
pseudo channel
-- Playing 'conf-inval
> > Bumping calls to clear a path for 911 is possible within Asterisk
already - see the "SoftHangup" application.
That sounds good, but what can trigger the SoftHangup app to drop other
calls automatically when 911 is dialed?
A short AGI script, perhaps?
It probably would not even require a sh
On Monday 23 June 2003 21:38, Jon Pounder wrote:
> >Also, it isn't very easy to 'test' either, as the staff at the 911
> > call centre won't appreciate your testing, and at least in
> > Australia, it is some sort of criminal/illegal offence to call
> > emergency for non-emergency situations.
>
> I
>
>>
>>
>>Also, it isn't very easy to 'test' either, as the staff at the 911 call
>>centre won't appreciate your testing, and at least in Australia, it is
>> some
>>sort of criminal/illegal offence to call emergency for non-emergency
>>situations.
>
> I had much the same thoughts. Currently my 911
Also, it isn't very easy to 'test' either, as the staff at the 911 call
centre won't appreciate your testing, and at least in Australia, it is some
sort of criminal/illegal offence to call emergency for non-emergency
situations.
Well, for testing purposes 911 could be replaced with any other number
Jon Pounder wrote:
I had much the same thoughts. Currently my 911 code is just commented
out for that very reason - I don't want to get in trouble for
accidentally making 911 calls to test it. Should I rely on that code
untested for when it is really needed most ? What are other people doing ?
C
Also, it isn't very easy to 'test' either, as the staff at the 911 call
centre won't appreciate your testing, and at least in Australia, it is some
sort of criminal/illegal offence to call emergency for non-emergency
situations.
I had much the same thoughts. Currently my 911 code is just comment
> Problem: 911 calls placed through Asterisk are associated with the
> physical location of where the CO trunks terminate. This is not really a
> problem when all extensions are located in the same building, but when
> Asterisk is used in a campus-like or otherwise networked environment, it
> can g
> > Bumping calls to clear a path for 911 is possible within Asterisk
> already - see the "SoftHangup" application.
> That sounds good, but what can trigger the SoftHangup app to drop other
> calls automatically when 911 is dialed?
A short AGI script, perhaps?
_
Great work!
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dylan
VanHerpen
Sent: Monday, June 23, 2003 7:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Module app_perl
Remove the space behind .com, like so http://asterisk.650dialup.com
And now that I *read* it back again, you can tell that English is not my
native language either
Dylan VanHerpen wrote:
Now that I reed it back, I can barely make sense of it myself!
Anyway, I was just thinking out loud, the example wasn't meant to be
parsed. Asterisk would need some lower le
Dylan VanHerpen wrote:
Now that I reed it back, I can barely make sense of it myself! Anyway,
I was just thinking out loud, the example wasn't meant to be parsed.
Asterisk would need some lower level changes to parse the extra field
holding the location information, and to apply the routing rul
Now that I reed it back, I can barely make sense of it myself! Anyway, I
was just thinking out loud, the example wasn't meant to be parsed.
Asterisk would need some lower level changes to parse the extra field
holding the location information, and to apply the routing rules to
substitute the Ca
I'm not sure I can parse your examples correctly. I'm not being
snide, but do you use Asterisk on a regular basis? Do you understand
how applications work, and how call handoff is done between Asterisk
servers? Your example doesn't seem to make sense, no matter how I
think about it.
Of cour
Hi, I’m trying to build a call center application that
allows attendants to come in the morning and dial a certain extension to make
their extension available.
I wouldn’t like to use the AgentLogin
app because their line would need to stay off-hook (is this correct?)
Is there any S
Remove the space behind .com, like so http://asterisk.650dialup.com/
Cheers, Dylan.
Uriel Carrasquilla wrote:
For some reason the page cannot be found.
http://asterisk.650dialup.com
what does it do?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of As
For some reason the page cannot be found.
http://asterisk.650dialup.com
what does it do?
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Ask Bjørn
Hansen
Sent: Monday, June 23, 2003 5:12 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Modul
Problem: 911 calls placed through Asterisk are associated with the
physical location of where the CO trunks terminate. This is not really a
problem when all extensions are located in the same building, but when
Asterisk is used in a campus-like or otherwise networked environment, it
can get mes
On Monday 23 June 2003 03:24 pm, Jordan Peterson wrote:
> Jerk
And one who is contributing to the development of Asterisk.
If you aren't the patient type and would like immediate answers
to your questions, I strongly advise calling Digium and buying a
support contract. The support techs are very
Hi all,
I have problems with native bridging with this configuration;
CPE(Mediatrix SIP-G.729)->Asterisk->Cisco AS5300 (SIP-G.729)
Problem is, remote side get very bad sound while local end
is getting very clear quality. If I set below configuration and make asterisk
to encode
Jerk
On Mon, 2003-06-23 at 13:02, Steven Critchfield wrote:
> replying to 2 other threads with your problem is not the way to get
> people to answer your question. If you search the archive you will see
> that voice modems are not really supported. This is why you don't hear
> audio. Now quit bei
- Original Message -
From:
Jorge
Cisneros
To: [EMAIL PROTECTED]
Sent: Monday, June 23, 2003 3:57 PM
Subject: [Asterisk-Users] Ringing tones
oh323
When i make a call using oh323 channels, how i
can send a ringing sounds to indicate to the users
When i make a call using oh323 channels, how i can
send a ringing sounds to indicate to the users that the call is in
progress
thanks
replying to 2 other threads with your problem is not the way to get
people to answer your question. If you search the archive you will see
that voice modems are not really supported. This is why you don't hear
audio. Now quit being impatient and _DEMANDING_ support.
On Mon, 2003-06-23 at 13:07, Jo
On Sun, 22 Jun 2003 12:59:09 +0300, destan <[EMAIL PROTECTED]> wrote:
I want to read to debug messages and try to interpret them but they
happen too fast, how can I log these guys to a file, or is there a file
like this already?
Greetings...
There's a program called "script" which will spawn a n
> If in your voicemail.conf you have * configured to the send message in
> an email you will NOT get a stutter dialtone or any MWI light you may
> have. I've just removed my email address from voicemail.conf.. much
> better like that...
I can't see how that would make any difference. Can you find
What appears to be hogging CPU? What interfaces are you running?
Mark
On Fri, 20 Jun 2003, Derek Beaumont wrote:
> Here's the problem:
> I start asterisk, and it takes up around 3-4% of my CPU
> resources.
> However, this number continues to climb over the hours until it
> is close
I finally got Gastman to compile but I get a bunch of "failed
assertions" when I run it and attempt to make a new extension. I have
latest CVS on Mandrake 9.1. Last error is:
(gastman:22534): Gdk-CRITICAL **: file ../../gdk/gdkdraw.c: line 311
(gdk_drawable_unref): assertion `GDK_IS_DRAWABLE (d
Would anyone be so kind as to explain why no voice is heard through the
phone when calling?
Thanks.
On Mon, 2003-06-23 at 10:34, James Golovich wrote:
> No point in reinventing the wheel here. PersistentPerl (aka SpeedyCGI)
> can eliminate the startup cost for using perl with AGIs.
>
> It work
No point in reinventing the wheel here. PersistentPerl (aka SpeedyCGI)
can eliminate the startup cost for using perl with AGIs.
It works great, and even allows the processes to reuse database
connections
James
On Mon, 23 Jun 2003, Anthony Minessale wrote:
> That is probably possible and not t
Hi TC,
Yes, there is a strong possibility. I will know more in a few weeks, just
trying to link up with Mark to discuss it.
Erik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of TC
Sent: Wednesday, June 18, 2003 6:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [As
That is probably possible and not too difficult.
I learned what AGI was about 30 minutes after I was finished
with the last revision of app_perl where I added support to
launch a perl function in a thread
(BTW I am suspicious that you
may ironically need perl with no threads compiled for i
On Monday 23 June 2003 09:32 am, Brancaleoni Matteo wrote:
> Today my pbx stopped responding to my sip phones..
> looking into the log, here what I got:
>
> Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new):
> Unable to allocate socket: Too many open files
The open file limit is pe
Thanks for the replies.
It seems that AVM B1 is the only active PCMCIA card that can be used
with Asterisk. The kernel supports this card, so I guess that the
driver can be built on non-x86 systems.
Regards,
Michael.
Olaf Menzel wrote:
On Friday 20 June 2003 13:28, Michael Manousos wrote:
Are th
On Sun, 22 Jun 2003, Steve wrote:
> Make sure you have the following installed:
> bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel, newt-devel,
> openssl096b, openssl-devel, readline and readline-devel.
readline and readline-devel have not been needed since November of last
yea
Many thanks, Martin ..
worked fine with dtmfmode=info
Dave
- Original Message -
From: "Martin Pycko" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 23, 2003 4:32 PM
Subject: Re: [Asterisk-Users] Asterisk CPU power requirements
> You need to find out which way your SIP g
On Mon, 2003-06-23 at 03:03, carlos del mayor wrote:
> TWO THINGS,GARY!
> 1-Sorry for the html, now it's off
> 2-The mail you're talking about has arrived two
> minutes ago.I KNOW read, thank you.
>
> I only wanted to know if somebody was working with
> this, in order to simplify a litle my work
THat's not it.
in zapata.conf you *also* need to have
signalling=pri_cpe or pri_net
Martin
On Mon, 23 Jun 2003, Michael Bielicki wrote:
> On Monday 23 June 2003 2:58 pm, Anton Yurchenko wrote:
> > Hello,
> >
> > I have an E100P, and in the zaptel.conf I have:
> >
> > span=1,1,0,ccs,hdb4,crc4,yel
Today my pbx stopped responding to my sip phones..
looking into the log, here what I got:
Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new):
Unable to allocate socket: Too many open files
Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc):
Unable to create RTP se
You need to find out which way your SIP gateway wants to receive the
DTMFs. There are three ways to do that. Read sip.conf.sample.
Martin
On Mon, 23 Jun 2003, Dave Alan Caruana wrote:
> hi there,
> I have an installed & working Asterisk server,
> which I am using to connect to a SIP service
> ab
Well how did you solve your previous problem then ?
Martin
On Mon, 23 Jun 2003, Thomas Haeger wrote:
> The problem before is solved. But now gives another problem ...
>
>
>
> == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
> == Registered channel type 'Tor' (Zapata Telephony
On Monday 23 June 2003 2:58 pm, Anton Yurchenko wrote:
> Hello,
>
> I have an E100P, and in the zaptel.conf I have:
>
> span=1,1,0,ccs,hdb4,crc4,yellow
> fxsks=1-10
delete the fxsks line and put:
bchan=1-15,17-31
dchan=16
>
> the light on the card is green( BTW what do all those states of the card
Hello,
I have an E100P, and in the zaptel.conf I have:
span=1,1,0,ccs,hdb4,crc4,yellow
fxsks=1-10
the light on the card is green( BTW what do all those states of the card
that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or
for the card?)
in the asterisks` zapata.conf I hav
You need G723 CODEC to be supportted on your asterisk server.
Best regards
Lubo
Dave Alan Caruana wrote:
My system is an asterisk machine,
with an E1 card (functioning) and
forwarding calls to a remote SIP
address ..
when a call connects I am getting the
following error :
NOTICE[1240577216]: File
My system is an asterisk machine,
with an E1 card (functioning) and
forwarding calls to a remote SIP
address ..
when a call connects I am getting the
following error :
NOTICE[1240577216]: File rtp.c, Line 330 (ast_rtp_read): Unknown RTP codec
19 received
can anybody tell me what this means
(& h
hi there,
I have an installed & working Asterisk server,
which I am using to connect to a SIP service
abroad. Although I can hear the IVR from the
ITSP, I cannot seem to send them digits from
my phone.
I have also noticed that the CPU usage on my
machine is up to 100% constantly and 99.9%
of that
Hi there!
I'm having a problem with TDM400P and Caller ID on Call Waiting.
Normal Caller ID works quite well, but I can't get CIDCW to work (tested
against Siemens phones).
I hear the tone, but a message on the console appears telling that the
phone doesn't support CIDCW; when it does (it's use
Anyone know if the Dialogic/Intel Proline 2v supported by Asterisk?
_
Help STOP SPAM with the new MSN 8 and get 2 months FREE*
http://join.msn.com/?page=features/junkmail
___
Asterisk-Use
I am getting the calls through but there is no voice , help guys !!!
- Original Message -
From: "Jordan Peterson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, June 23, 2003 11:14 AM
Subject: Re: [Asterisk-Users] no voice on dialogic d300
> This could be related to my not he
malaysia and sri lanka toowould be greately appreciated :-)- Original Message -From: "Michael Labuschke" <[EMAIL PROTECTED]>To: <[EMAIL PROTECTED]>Sent: Saturday, June 21, 2003 11:48 PMSubject: Re: [Asterisk-Users] Billsec on CDR> germany here :)))>>> *** REPLY SEPARATOR **
Hi.
I've found a problem when I pickup a remote sip phone with *8.
There're both budgetones 102 and are both in the same group.
When one sip phone is ringing, I can pickup the call from
another sip phone, but the first one keeps playing a loud
busy signal... that don't go away until I receive anot
The problem before is solved. But now gives another problem ...
== Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
== Registered channel type 'Tor' (Zapata Telephony Driver w/PRI)
== Starting D-Channel on span 1
ERROR[1024]: File chan_zap.c, Line 5947 (start_pri): Unable to o
Hi,
How do I process multiple lines of the extention.conf on dial out before actually
connecting the call to the user??
Here is the problem..
I have an access number for cheap internationsl calls, This number has to be dialed
and then a DTMF string needs to be passd to the service for the numb
Hi all,
can somebody help me with pri configuration?
Here my zapata.conf:
; Zapata telephony interface
;
; Configuration file
[channels]
switchtype=euroisdn
signalling=pri_cpe
;group=1
channel => 1-15,17-31
;group=2
channel =>32-46,48-62
;group=3
channel => 63-77,79-93
;group=4
channel =
Muchas gracias, Brian, así lo haré.
Tiene usted un buen español, no lo pierda.
Saludos
Carlos
--- Brian Capouch <[EMAIL PROTECTED]> escribió: >
carlos del mayor wrote:
> > TWO THINGS,GARY!
> > 1-Sorry for the html, now it's off
> > 2-The mail you're talking about has arrived two
> > minutes ago.I
On Tuesday, Jun 17, 2003, at 20:43 America/Los_Angeles, Anthony
Minessale wrote:
Here is a copy of the first release (comments appreciated)
http://asterisk.650dialup.com
Although I haven't had time to play with it: very neat!
- ask
--
http://www.askbjoernhansen.com/
carlos del mayor wrote:
TWO THINGS,GARY!
1-Sorry for the html, now it's off
2-The mail you're talking about has arrived two
minutes ago.I KNOW read, thank you.
I only wanted to know if somebody was working with
this, in order to simplify a litle my work
(documentation and all that it's what i was
Le dim 22/06/2003 à 16:18, Hervé Thibaud a écrit :
...
> I try to connect directly the both to fwd.pulver.com and now i have a
> perfect sound but the question is perhaps links after opening session
> is only on the local networks with 10Mb/s.
> Once i can (when i'll have an external user to call)
TWO THINGS,GARY!
1-Sorry for the html, now it's off
2-The mail you're talking about has arrived two
minutes ago.I KNOW read, thank you.
I only wanted to know if somebody was working with
this, in order to simplify a litle my work
(documentation and all that it's what i was looking
for).
Gary, I d
Hi..
I want to try and link two * systems together using IAX..
I want the extensions on both servers to be seemlessly available to users of the other
system.. and dialing out will hapen on the second server.. structure below:
Extentions--(SIP)--ServerA--(IAX)--ServerB--(PSTN)--World
TWO THINGS CARLOS !
one, please turn off your html formatting.
second, it was answered (and even can be seen in your posting..)
cdr_mysql.conf which you will find in /etc/asterisk
actually the original is cdr_mysql.conf.sample
have read pls.
On Mon, 23 Jun 2003 09:02:22 +0200 (CEST), ca
I don't mind the slight diminution in my hearing faculties that accrues
each time I have a call come in while I'm talking from (I think I have
this straight) the ADSI tone, but yesterday one of my callers asked me,
"You OK?" after it sounded, so it must be at least minimally audible to
the othe
I'm only asking for some examples of cdr_mysql.conf, nobody has done anything with cdr and mysql? If you think is better another DB,,, tell me, please!
thanks in advance
carloscarlos del mayor <[EMAIL PROTECTED]> wrote:
can you be more explicit, please? or give me some examples? please, i'm little
Spain, would be great!
thanks,
carlosStephen Davies <[EMAIL PROTECTED]> wrote:
On Fri, 20 Jun 2003, Tan Aks wrote:> Isn't there any way to make callprogress work for people in Europe? What is> it that is needed to make it work?I've done call progress for the UK. Patch to the -dev list by the endof
> is there somebody who can help me with getting ADSI phones
> in Europe
>
> I' am a little bit desperated. I need such a phone to play with * and adsi
> features.
> But i don't find a vendor who produce or a distributor who distribute such
> phones in Europe.
> I have found this link in
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