Sorry for posting this OT:
If you are an asterisk integrator in the Dallas Area or are willing to
travel for a Presentation please mail me to [EMAIL PROTECTED]
Thank you,
Victor Perez
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for
altering "src.drv/makefile" toreplace the occurance of
"CARD_PATH".
I tried to install the driverusing the
default makefile, with the final result of "capi not installed - No such
device or address (6)"
Are there any updated document
installations would be very helpful.
Please email me to [EMAIL PROTECTED]
Thank you,
Victor Perez
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for at least local calls (DFW)
- National and international Toll Free DID numbers
Altought OT, I believe this information would be useful for all
asterisk integrators reading this forum so you can reply to this
message or mail me to [EMAIL PROTECTED]
Thank you,
Victor Perez
cmould wrote:
Hi:
Just saw your post while trying to solve a similar asterisk problem. Did
not see any responses. Was your problem solved and what was the solution?
Carey
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This has been an interesting discussion. I'll chime in with my
experience here.
I have two servers. One with the cheapest motherboard and athlon
processor I could find on Newegg.com. The other is a 1999 era
motherboard with a Via C3 processor, again a bargain basement special.
The Athlon
Jim Van Meggelen wrote:
If I may, I'd like to ask you some general questions about the
environment these systems are running in.
- How are these systems powered and grounded?
Not optimally by a longshot. On the Athlon machine, my main machine, all
the equipment is plugged into 2to3 prong
Jim Van Meggelen wrote:
Frankly, what is most interesting is the fact that your systems are
trouble-free. Certainly if you were to ask if such systems could be put
into production, you would probably be advised not to expect much.
One last thing. I have a somewhat special PSTN connection. I
with either asterisk or that
sipura firmware.
Victor Perez
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Hello,
I'm just wondering what is the situation
today, 24 Nov 2004, regarding asterisk timer for freebsd. I would like
to know ifthere isany way to run Meetme on Freebsdorif
there is anybody currently working on it.Cheers,
Victor.
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Hi all,
I think there is no asterisk-addons version
for freebsd. Am I right? I tried to compile the standard version but I couldn't
do iton FreeBSD, may be the idea is as crazy as try to install asterisk
for linux on freebsd!
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).
Could anybody help me?
Thanks,
Victor.
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Hello everybody
Do you remember I sent a case to the list about a digit 1 phantom I
received when I call the method get_data or stream_file? Fine. I realized
that It does not happend when I omit a subrutine I my code where I open a
TCP client socket by IO::Socket.
I think It is because
Does anybody know how to stop the AGI process after the user Hang-Up? 'Cause
it stills running if the transaction is ended by the user.
Thanks
Víctor
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[EMAIL PROTECTED]
Sent: Friday, October 29, 2004 10:01 PM
Subject: Re: [Asterisk-Users] $AGI-stream_file
On Fri, 29 Oct 2004 16:59:56 -0400, Victor Cartes
[EMAIL PROTECTED] wrote:
Hello everybody!
I've got a problem here. I writing an AGI in Perl and when I used
Can somebody tell me why this does not work in perl (AGI)???
#!/usr/bin/perl -w
use Asterisk::AGI;
$AGI =3D new Asterisk::AGI;
%input =3D $AGI-ReadParse();
$AGI-say_number(1984);
At the console I can see the program start and finish OK, but it doesn't
execute the
I realize I receive permanently the 1 digit. That's should be the reason
the say_digits seems not to work. Now I've got to change my question...
Why do I receive this digit if none send it to me?
- Original Message -
From: Victor Cartes [EMAIL PROTECTED]
To: Asterisk Users Mailing
Hello everybody!
I've got a problem here. I writing an AGI in Perl and when I used the
stream_file method It did not work. Then I realized that the next line has
no waited for the streamed file end, the program has just gone on.
What should I do to make the routine wait for the stramed file
I'm trying to configurate my first asterisk system.
My test plant has an E100P Card and a 4 FXS TDM 400P card.
I've an E1 configured for 15 channels.
Telco says that crc4 is configurated so my zaptel.conf is:
loadzone = it
defaultzone=it
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
Hi!
Does anybody know how to convert .gsm file format
to .g729 in order to use it for an IVR system?
Thanks in advance.
Vïctor
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Ryan Courtnage wrote:
Hi all,
We have an analog line from telco, on which 3-way calling is subscribed
to. This line is plugged into an FXO module on a tdm400p.
If an incoming call comes in on this line, can */zaptel send Flash to
telco via the FXO module? If it could, then an incoming call
Hi Rgis,
Were going to build an IVR system with a TE405P and 4 E1. Were sure
that the 120 channels will be filled by 120 simultaneous calls during
peak, so we want to have the good server to manage this.
We wonder a lot of things and maybe you could help us.
- Are you ever build a similar
Christian Victor schrieb:
I am trying to suppres the transmission of my CallerID when I place a
call using a .call file in /var/spool/asterisk/outgoing
Okay - now I have a little Progress. :-) Suppressing CallerID on a PRI
is done by setting the CallingPres parameter. But unfortunately
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Yes - that was what I thought too. But unfortunately leaving out the
parameter or setting it to '' will cause transmission of the default
number (usually subscriber number + 0)
On a PRI you
Hi!
What about the CallerID parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
I'm not the original poster, but I think this will not work. Just changing
the Calling Number (where the callerid field ends up in the isdn setup
message) to nothing will most of the
Henry Devito schrieb:
Most LEC's CLEC's, at least in our area, require sending a number (CSID)
before the call is completed. This is do to E911 features and ANI. If you
do not send a number the call will fail. If you truly want to block caller
ID I would contact your carrier and they should be
this but in this case I need to
call anIAX extension in the same machine. I have no problem to create one
IAX user and use one of the existing context of extensions.conf but I don't know
how to call the IAX user from the rest of thephones.
Any help, please.
Victor
Hi!
At the moment I am trying to use Asterisks /var/spool/asterisk/outgoing
folder to dial larger amounts of calls at specific times to realise a
wakeup call/reminder solution.
The problem is that when I dump more .call files in the spool directory
the free lines I get many Unable to request
Hi!
I frequently get errors like Call failed to go through, reason 0 in
/var/log/asterisk/messages
Are the reasons (0,3 and 5 in my case) explained anywhere? I did not
find any info in the wiki.
Thanks,
Christian
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Hi!
I am trying to suppres the transmission of my CallerID when I place a
call using a .call file in /var/spool/asterisk/outgoing
Callerid:
Callerid: and
Callerid: '' made the call transmit the default number (headnumber+0)
Callerid: 1234 made the call transmit 1234
Using *31* in front of the
Of Christian Victor
Sent: Monday, September 13, 2004 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Playback Fileformats
Brian West schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
,
the author of the current version according tochan_sip.c.
Cheers,
Victor Alvarez.
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Hi!
I wonder what sondfile formats Playback() could play. I know it plays
GSM but to save the CPU time I will avoid converting GSM alaw for my
E1 and user alaw compressed wavs.
Unfortunately the wiki does not list the supported filetypes but I know
that there are a few.
Thanks in davance
Brian West schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
but not the file formats (.wav etc) supported.
Christian
I wonder what sondfile formats Playback() could play. I know it plays
GSM but to save the CPU time I will avoid
Eric Wieling schrieb:
At the cli do
show file formats
Sorry - that does not work. And SHOW AUDIO CODECS shows me the codecs
but not the file formats (.wav etc) supported.
Upgrade your Asterisk. show file formats works for Asterisk
CVS-HEAD-07/18/04-11:25:14
I am running Asterisk
Hello!
I am trying to install a TN405P on a P4-3GHz-HT machine running Debian
Sarge with kernel 2.4.27. When I start Asterisk in -c mode it always
shows
== D-Channel on span 1 up
== Restart on requested on entire span 1
== D-Channel on span 3 up
== D-Channel on span 2 up
== Restart on
Scott Stingel schrieb:
It's normal, in fact I use it to be sure that everything's ok, since I think
it will not occur unless we have no alarms on the spans!
Hehe - it sounded too good to be true that everything worked well from
the beginning. ;-)
Thanks for your info!
Christian
John Stegenga wrote:
[sarcasm on]
Thank you ALL for your warm welcome to this list. I posted this message
yesterday, and since I'm only getting Digest I figured I'd see a response in
a day...
[sarcasm off]
C'mon. This is the Asterisk Users mail list, isn't it? This is where the
Voip WIKI tells
Greetings All,
I have a new post on the blog. It goes a little bit more in depth on
wcfxo.c and touches on zaptel.c. Two more screen shots. Loads of fun.
Take a look: http://zapteldoc.blogspot.com
Regards,
Victor
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it.
Couldanybody help me? Any idea about how to do it?
Regards,
Victor.
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.
retrieve_sip_conf_from_mysql.pl seems to be a
good B plan. I will have to recharge sip.conf manually but.. If this is the way,
I willfollow it. Anyway if plan A works with user and password, Why it
can't work with the rest of parameters??
I'll continue my work on Monday.
Have a nice weekend!
Victor
Holger Schurig wrote:
I'd thought I'd been through the whole Zapata Telephony Site. Could you
e-mail back and point to the specific links you had in mind?
Start with
http://www.zapatatelephony.org/philos.html
and dive into
http://www.zapatatelephony.org/project.html
and then into
. I'm not a web designer so I
was inclined to go with one of blogspot's limited theme choices. If
anyone has a preference of blogspot theme or can give a succint set of
instructions on manipulating the current theme - please let me know.
I'll defer to your judgement.
Regards,
Victor
be the
best place for community involvement with the document.
Everyone keep chiming in.
Thanks,
Victor
Jon Bebeau wrote:
Victor... You Go Boy!!! I think many of us, me at least, would welcome some
doc on the underpinnings of Zap and friends.
I'll be happy to be a second set of eyes to help edit
Tony,
I'd thought I'd been through the whole Zapata Telephony Site. Could you
e-mail back and point to the specific links you had in mind?
As I recall, the tormenta driver source and a brief discussion on the
linux port had the most releveant information.
Thanks,
Victor
Hello Everyone,
The blog for the project is up and has a couple of posts. Haloscan
commenting is enabled. I've included a site feed but I'm a little unsure
about it.
See http://zapteldoc.blogspot.com.
Regards,
Victor
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?
Regards,
Victor
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[EMAIL PROTECTED] schrieb:
So basically if there is a problem with line 1 out of 12, no calls
are going to get through...surely this isn't expected behaviour.
What about putting all the channels in a group (1 for example) and use
Zap/r1/number to dial. This wil use round robin to pick a
Hi Kurt!
maybe I oversee somth. very obvious, but I'm a little puzzled about the
following 'error':
When I make a call from the PBX to * I get number not available, but on
debug I see, that asterisk is searching just for the first digit in the
extension, which of course doesn't exist, eg:
I
You cannot compile zaphfc with latest CVS head. You have to donwload
specific date version using the download.sh included script.
BTW I have some problems with RC4. It works fine with my 2 isdn pci boards,
but it seems to be unable to drive my TDM400 ...
Try RC3, at the moment seems to be more
or something. But I
think I installed everything that is known to be required under Debian
Sarge.
Maybe someone of you has a clue.
Chris
Christian Victor schrieb:
You cannot compile zaphfc with latest CVS head. You have to donwload
specific date version using the download.sh included script.
BTW I have
. But I
think I installed everything that is known to be required under Debian
Sarge.
Maybe someone of you has a clue.
Chris
Christian Victor schrieb:
You cannot compile zaphfc with latest CVS head. You have to donwload
specific date version using the download.sh included script.
BTW I have some
Hi!
Hi,How to accepts the call and plays a voice message on the line without
billing the caller ? This may
be necessary for IVR applications that want to explain features of the
service offered .
Usually this is done on the network side of the setup. You should ask
your telco for this.
Chris
Hi!
I have a problem compiling the zaphfc driver for my HFC-PCI cards. I use
Asterisks latest CVS and bri-stuff.0.1.0-RC4.
The install.sh compiles zaptel and libpri without problems. But when it
tries to compile qozap and zaphfc it show the following errors:
qozap.c:206: error: structure has
Greg Hill schrieb:
I can makes calls with my asterisk using X-lite softphone, I can even
call the 877 number and it works perfectly! But when I call the 877
number over the PSTN, it does nothing. A busy signal.
Here in Germany when you order a line with 3 digit extensions you often
only get
Hi!
Does anyone of you have an idea how to detect an answering machine on a
dialout call?
I am working an a voicemail system wich calls the subscriber but I don't
want to fill their answering machine.
Maybe I could detect somehow if there is incoming voice when playing the
message. usually
is there a way to write uniqueid from call to a varable?
Victor Medrano
[EMAIL PROTECTED]
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hI i need some help with asterisk and h323, when a call asterisk
i have one way audio . the version i;m using Asterisk
CVS-04/22/04-23:15:24 and chan_h323 , but it works with another phone
like arrayvox.
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I just tried to upgrade my sipura to firmware 1.0.35a and now I can't connect to it.
It still works but any connection to ports 23 and 80 makes it reboot. Even the flash
tool makes it to crash when trying to connect. Anybody else experiencing this problem?
Regards,
Victor Perez
?
Regards, Victor Perez
I don't know if this helps, but I started having this problem after I sent out a fax.
My fax machine was connected to line 1 at that time. I tried changing the FAX
detection settings but no luck.
Regards,
Victor Perez
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I can transfer a call from my sipura using flash, *98 and number, the problem is If
I hangup before the destination extension picks-up, the transfer is lost.
Is there a way to transfer and hangup without having to wait for the destination
extension to pickup?
Regards,
Victor Perez
I checked chan_oh323.c and indeed it only takes one parameter now so I am wondering
what was that old parameter for and when did they take it off so I may try pulling
that version of asterisk to try with.
Regards,
Victor Perez
post a working example or point me to documentation in
english (found stuff in russian and korean so far)
Regards,
Victor Perez
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Is there a way to set default caller id info to pass to * when the telco does not
provide it?
Regards,
Victor Perez
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Has anybody tried to install * in any of these minimalist linux distros like tinylinux?
Which linux distro would you use to run * in old P2, P3 boxes?
Regards,
Victor Perez
[EMAIL PROTECTED]
(469) 221-4189
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Apr 9 11:35:15 localhost kernel: wcfxo: Unable to request IRQ 0
I have this same setup (asterisk on mandrake 9.2) already working in other pc... this
is an old AT pc... any ideas?
Regards,
Victor Perez
a workaround in our Merlin we are thinking on
using FXS cards for incoming, so the merlin gets a dialtone from * and dials the VoIP
extension. FXO trunks would be used for outgoing.
Has anybody tried such a crazy setup?
Regards,
Victor Perez
of hardware would you recommend to setup some analog extensions as
DID trunks between a PBX and *?
Thanks in advance,
Victor Perez
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Hello all,
I've posted on this problem before. Well here goes
again.
I have an intel mobo with a p4 2.4ghz proc, 1GB Ram. It has
built-in ethernet and vga and 6 pci slots.
I dreamed of making this my household communications server:
internet router, firewall, vpn and asterisk.
Hello again,
Thanks for the timely responses.
Andrew:
Asterisk doesn't dump any messages except when a call comes in and asterisk
tries to ring an extension - it leaves a device busy type of message.
I checked /proc/interrupts. The fxs card is still there after it dies, but
the interrupts
Steve,
I have the tdm card on it's own IRQ. That's one of the first things I tried.
Both of my fxo cards are on the same IRQ and they seem to hold together.
It's interesting that you bring up the timing issue. Why would the tdm card
be so sensitive? I can understand a drop in voice quality but
Tilghman,
I have a feeling we're getting somewhere.
I ordered three cards the very day they went on sale through the digium
website.
Yes, it's revision C. I guess I'll talk to digium about this.
Thanks,
Victor
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Andrew:
I tried the asterisk -vvvc suggestion and I didn't get any messages when the
card died.
Here's /proc/interrupts before I take out the sound card:
CPU0
0: 102777IO-APIC-edge timer
1:471IO-APIC-edge keyboard
2: 0 XT-PIC cascade
a lot.
Victor
CPU0
0: 102777IO-APIC-edge timer
1:471IO-APIC-edge keyboard
2: 0 XT-PIC cascade
8: 4IO-APIC-edge rtc
14: 9159IO-APIC-edge ide0
15: 6IO-APIC-edge ide1
17:1995769 IO-APIC-level
i did with cisco callmanager with smdi integration .
and h323 .
works very well .
Victor Medrano
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Samy Touati
Sent: Monday, December 15, 2003 11:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Using
did you configure dialpeer voip in the cisco .
pointing to * ip addr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Pavel
Litvinenko
Sent: Tuesday, November 25, 2003 10:12 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco to use * as a gateway?
time lost both dt and battery. A reboot brings it
back.
Suggestions?
TIA,
Victor
hello to all.
I have a PC with a E400P Card with 4 E1 with a RJ48 jack
is posible to convert any RJ48 jack to 30 phone line ?
is for example.
1 E1 come from the telco operator to PC
and with asterisk i a lot of things and before send it to a phone but, where i plug
this phone ?
Bye
Title: Message
I have
the same problem any one
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexandru
CoseruSent: Monday, October 27, 2003 1:10 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] SIP -
H323 Seg fault.
A very
Title: Message
is
true i have the same problem with my vg200 and mc3810 ( cisco
device)
regards ,
Victor
Medrano
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexandru
CoseruSent: Tuesday, October 28, 2003 1:01 PMTo:
[EMAIL
Title: Message
both
works fine , cisco expensive .. snom 200 GOOD
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
JozwiakSent: Tuesday, October 28, 2003 3:40 PMTo:
ASTERISK USERSSubject: [Asterisk-Users] Cisco or Snom
You can run any agi script including pascal
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Friday, October 24, 2003 7:55 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] AGI questions..
Hi,
First off, can AGI scripts be created using
You can make samples , and the current config get extension .OLD , they
did nor override
Just create a new one .
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of rnc Info
Lists
Sent: Friday, October 24, 2003 2:18 PM
To: [EMAIL PROTECTED]
Subject: Re:
Download binary with java , works fine with 2000 + Xp
regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven M.
Sokol
Sent: Friday, October 24, 2003 4:43 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Compiling gastman under Win32
Can
get to 35 g711 , the
asterisk hang.
some one ,
??
i'm using
asterisk-0.5.0 and oh323 5.5
regards
,
victor
medrano
Hi to all,
does Digium's X100P, TDM40B E100P works in
PCI-X slots? I want to install those cards with new Intel server
boards
Thanx in advance,
Victor...
Use modprobe instead of insmod. If you use insmod
then you have to first insmod zaptel.
regards
Martin
On Fri, 28 Feb 2003, Victor Sanchez wrote:
i have error in install module of tor2.
but it compile good.
what happen ?
ivr2:/usr/src/zaptel # make clean; make install
rm
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