I am trying to test my dialogic card with asterisk
, ihave an E1 card in asterisk and onedialogic(d300) in
anothermachine. Both are connected through a cross
cable.
Asterisk(digium E1)calls the
dialogic(d300) card to its answer demo. The answer demo shows that the call is
received but
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 the
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),
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
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/
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 =
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
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
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
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 *** On
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 hearing
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
___
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
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
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
that
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
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
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,yellow
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
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 gateway
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
year.
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
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 per
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 works
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
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 to
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 me
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,
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
- 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
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 being
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
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
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
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
Hi, Im 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 wouldnt like to use the AgentLogin
app because their line would need to stay off-hook (is this correct?)
Is there any SET
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
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
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
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
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
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
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
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 ?
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
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
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 short
47 matches
Mail list logo
