I have following settings done on my Fedora8:
Downloaded
openh323-v1_19_0_1-src-tar.gz
pwlib-v1_11_1-src.tar.gz
Extracted them in /root/openh323 and /root/pwlib
Exported the following variables:
PWLIBDIR=/root/pwlib
export PWLIBDIR
OPENH323DIR=/root/openh323
export OPENH323DIR
Hello,
I asked myself the same questions as obviously, Asterisk 1.6 introduces new
T.38 functions and it's still a bit difficult to know what's really missing
to set a system up just as you suggested.
Is it possible to use Asterisk as a T.30/T.38 gateway ?
If positive, which software are needed,
Arturo Ochoa schrieb:
Dear list,
I got this scenario…
FAX Machine - FXS (tdm800) -Asterisk - SIP - OPENSER - SIP -
Asterisk - FXO(tdm400) - PSTN - FAX Machine
I’ been reading a lot of Faxes and t.38 protocol... and I found that
Asterisk 1.6 has the possibility to do FAX t.38 Gateway
Hi,
I have following settings done on my Fedora8:
Downloaded
openh323-v1_19_0_1-src-tar.gz
pwlib-v1_11_1-src.tar.gz
to my knowledfe chan_h323 should be compiled against
openh323-v1_18_0-src.tar.gz
and
pwlib-v1_10_3-src-tar.gz
cheers
--
2008/8/7 Mr Shunz [EMAIL PROTECTED]
have you tried to configure the ports in PTMP mode instead of PTP?
I found that for some PBX works better...
Yes, I tried but it didn't work. Anyway Philips pbx needs to work in PTP
mode.
can you post your misdn-init.conf and misdn.conf?
Of course :)
Hi,
addons 1.6 don't compile here. Any ideas?
Terve,
Stefan
[EMAIL PROTECTED]:/usr/src/asterisk-addons-1.6.0-beta4 make
CC=gcc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent makeopts
make[1]: Entering directory
`/usr/src/asterisk-addons-1.6.0-beta4/menuselect'
On Fri, Aug 8, 2008 at 1:43 AM, Arturo Ochoa
[EMAIL PROTECTED] wrote:
Thanks Memo,
I've already see that article before, the problem is that this solution is
useful when you want asterisk (via t38modem) to terminate the call...
Someone send you a Fax using t.28 and this software
P.s. I tried to use an Openvox B800P instead of the Digium B410P and it
worked immediately, without any problem or modification to the default
configuration files. Maybe Digium card is fisically damaged?
mmm... i have an installation with two OpenVOX B400P and misdn-1.1.7 connected
to a
On Thu, Aug 7, 2008 at 11:00 PM, Guillermo Salas M.
[EMAIL PROTECTED] wrote:
El jue, 07-08-2008 a las 13:31 -0600, Arturo Ochoa escribió:
Has anyone have experiencies on this kind of scenario... what
version?.. patches?... or any information regarding this goal will be
VERY helpful...
Hi
On Aug 7, 2008, at 2:31 PM, Arturo Ochoa wrote:
I got this scenario…
FAX Machine - FXS (tdm800) -Asterisk - SIP - OPENSER - SIP -
Asterisk - FXO(tdm400) - PSTN - FAX Machine
Asterisk 1.6 currently has T.38 origination and termination support.
It does not yet have fax gateway support.
On Aug 8, 2008, at 4:48 AM, Stefan Gofferje wrote:
Hi,
addons 1.6 don't compile here. Any ideas?
It looks like you're trying to compiled Asterisk-addons 1.6 against
Asterisk 1.4. You will need to install Asterisk 1.6 before you can
compile and install Asterisk-addons 1.6.
--
Russell
Tilghman Lesher wrote:
On Wednesday 06 August 2008 04:09:13 Pavel Jezek wrote:
A week ago, I tried give realtime priority to asterisk proces using -p
switch,
asterisk was running inside astcanary,
but yestarday asterisk probably starts eating all cpu and lock any
access to computer,
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Asterisk 1.6 currently has T.38 origination and termination support.
It does not yet have fax gateway support.
--
Russell Bryant
Russell, Can you please clarify what you mean. I think there is still a bit
of confusion as to what termination and gateway and Asterisk 1.6 is all
about,
Hello,
Please let someone throw more light on this command and it usage.. i
tried a search but can't to get anything useful.
Thanks,
Robor
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22
On Aug 8, 2008, at 7:39 AM, JR Richardson wrote:
Asterisk 1.6 currently has T.38 origination and termination support.
It does not yet have fax gateway support.
--
Russell Bryant
Russell, Can you please clarify what you mean. I think there is
still a bit
of confusion as to what
Hi,
Russell Bryant schrieb:
It looks like you're trying to compiled Asterisk-addons 1.6 against
Asterisk 1.4. You will need to install Asterisk 1.6 before you can
compile and install Asterisk-addons 1.6.
So, 1.6 must be _installed_ before compiling addons? It's not enough to
have it
Stefan Gofferje schrieb:
So, 1.6 must be _installed_ before compiling addons? It's not enough to
have it readily compiled in the neighbour dir?
Confirmed - works. Thank you!
Terve,
Stefan
--
Last words of a stormchaser:
Where is that rotation on the radar?!
On Thu, Aug 07, 2008 at 04:12:26PM -0700, bilal ghayyad wrote:
CRM: Customer Record Module which is any kind of application.
Well, no; CRM means Customer Relationship Management...
Cheers,
-- jra
--
Jay R. Ashworth Baylink [EMAIL PROTECTED]
Designer
On Fri, Aug 08, 2008 at 04:28:03PM +0300, Stefan Gofferje wrote:
Hi,
Russell Bryant schrieb:
It looks like you're trying to compiled Asterisk-addons 1.6 against
Asterisk 1.4. You will need to install Asterisk 1.6 before you can
compile and install Asterisk-addons 1.6.
So, 1.6
Anton schrieb:
Does anyone tried BRI with asterisk for DATA transfer? My
customer
wants BRI connection, but he wants it for the data, and I
have to
bring connection to his office, so I see the connection as
follows:
E1-(CORE_ASTERISK)-(IAX2)-(EDGE_ASTERISK)-BRI - so
Why would
JR Richardson wrote:
Asterisk 1.6 currently has T.38 origination and termination support.
It does not yet have fax gateway support.
--
Russell Bryant
Russell, Can you please clarify what you mean. I think there is still a bit
of confusion as to what termination and gateway and Asterisk
bkruse wrote:
I would checkout Switchvox :)
http://www.digium.com/en/products/switchvox/
-Brandon
Ken D'Ambrosio wrote:
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly,
Hello,
I have installed iaxmodem(1.1.1) and hylafax(4.4.4) but my Asterisk(1.2)
configuration seems to be wrong (calls are destroyed by asterisk).
Does someone know how to configure Asterisk (iax.conf, sip.conf and
extension.conf)???
This is my configuration of iaxmodem:
device
Stefan Gofferje wrote:
So, 1.6 must be _installed_ before compiling addons? It's not enough to
have it readily compiled in the neighbour dir?
That is correct, at least for the easy case.
Alternatively, you can specify the Asterisk location as an argument to
the configure script.
My office Asterisk box has a TDM04B card for three land lines and a GSM
gateway. I have noticed that the Zap channels get stuck a couple times
a week and I have to restart Asterisk to clear them. Here is what I see
in the console:
Connected to Asterisk 1.4.21.2 currently running on
Check out www.thirdlane.com they have a excellent end user portal.
Ken D'Ambrosio wrote:
I badly want to roll out Asterisk at my job. Unfortunately, my boss is
dazzled by shiny objects. We had a vendor in today who showed us their
system which, honestly, didn't suck -- but boy, is it going
HOW ?
Rizwan Hisham wrote:
have done it, and its working fine. but still expecting to receive
some new ideas.
On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
hi all,
i just finished developing some incoming call features in a
I think the original request was for a USER portal, I'm not sure that
SwitchVox and CogoBue have a user portal.
Cheers,
Dean
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Signorello
Sent: Friday, 8 August 2008 10:20 AM
To: Asterisk
Russell Bryant schrieb:
Stefan Gofferje wrote:
So, 1.6 must be _installed_ before compiling addons? It's not enough to
have it readily compiled in the neighbour dir?
That is correct, at least for the easy case.
Alternatively, you can specify the Asterisk location as an argument to
the
Ok, so it's clear now that this feature is missing on Asterisk, but as
Russell states, it's on the roadmap.
So, Can you guys give an alternate idea on what to do on this scenario:
One customer has this situation:
The headquarters are located on MTY, Mexico. They have 2 landlines on
Edinburgn TX,
hi,
i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts.
the problem is that some variables (and maybe all, not sure) like
ANSWEREDTIME does not kept if the caller hangs up.
my agi script continues to run after caller/callee hangup but the
variables are not set properly if
Sarò in vacanza fino a martedì 19 agosto con scarsa possibilità di accedere a
e-mail e telefono. Per richieste urgenti, vi prego di contattare Wildix srl al
numero di telefono 0461 74 30 891 o all'indirizzo e-mail [EMAIL PROTECTED],
altrimenti vi risponderò al mio rientro.
Dimitri Osler
I
Hello,
I'm looking at getting a dedicated server from ServerBeach to host
some light Asterisk/VoIP/SIP stuff. Has anyone used them for this
before? I'm pretty sure I've heard good things (in general) about
them but VoIP is a very different animal than web hosting - especially
for the network
My organization may be able to help you out on this, I am forwarding
this email to my sales team.
Kristian Kielhofner wrote:
Hello,
I'm looking at getting a dedicated server from ServerBeach to host
some light Asterisk/VoIP/SIP stuff. Has anyone used them for this
before? I'm pretty
Les.net hosts a significant chunk of their services in a few of the
ServerBeach data centers. I've had great quality with Les.net. ServerBeach
picked Les as their Geek of the Week last year:
http://www.serverbeach.com/aboutus/geek_of_the_week.php?id=8year=2007 .
--
Alex Robar
[EMAIL PROTECTED]
You are using AGI or DeadAGI ?
Paradise Dove wrote:
hi,
i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts.
the problem is that some variables (and maybe all, not sure) like
ANSWEREDTIME does not kept if the caller hangs up.
my agi script continues to run after
Switchvox does have a user login where they can change their specific
settings, etc. You can even checkout the demo here:
http://www.switchvox.com/sv?cmd=demo_questions
-bk
Dean Collins wrote:
I think the original request was for a USER portal, I’m not sure that
SwitchVox and CogoBue have a
I'm not sure, but I don't think he's going to be back in the office until the
19th of August. :P
___
Chris Hoff
Telecommunications Administrator
SEI LLC
Voice +1 701 298 8865 Ext 2189
Mobile +1 701 361 5976
Fax +1 701 298 8860
Email [EMAIL PROTECTED]
Hi,
I started testing chan_mobile. Target is having some old phone with a
duosim (second card with same number) put to silent somewhere in the
rack with the *. That phone should mainly take incoming calls and after
45secs put them to the mailbox AND permit me to talk via my nice Cisco
desktop
I am using Asterisk 1.2.24. I know it should be upgraded, but that is
not an option at this point for this working system.
I am experimenting with using an external application to control whether
a call should be connected. Most of the time it works. Sometimes the
dial plan never comes back
I'm using AGI and set AGISIGHUP=no
to make it keep on running on channel hangup
On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
You are using AGI or DeadAGI ?
Paradise Dove wrote:
hi,
i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts.
the
I am a returning Asterisk user.
It has been a few years since I played with it and trying to get a server up
for proof of concept
What is the easiest method of having asterisk dial 5 numbers simultainiously
and deliver a pre recorded message?
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Bradley Sumrall wrote:
I am a returning Asterisk user.
It has been a few years since I played with it and trying to get a server up
for proof of concept
What is the easiest method of having asterisk dial 5 numbers
Try DeadAGI and it should work..
Paradise Dove wrote:
I'm using AGI and set AGISIGHUP=no
to make it keep on running on channel hangup
On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
You are using AGI or DeadAGI ?
Paradise Dove wrote:
hi,
i'm using
On Fri, Aug 08, 2008 at 10:16:31AM -0500, Carlos Chavez wrote:
My office Asterisk box has a TDM04B card for three land lines and a GSM
gateway. I have noticed that the Zap channels get stuck a couple times
a week and I have to restart Asterisk to clear them. Here is what I see
in the
Hi,
I have configured all IAX clients with encryption. I use Zoiper as a
softphone. When I make a call in the LAN from desktop-PC to *, the call
is - according to wireshark not encrypted. Wireshark identifies the
packets as normal G.711 mu-law packets. However, * reports the client as
encrypted:
Here is a simple Perl implementation to generate call files . .
You'll still need something for it to execute after the call files are
generated; either a simple AGI app that streams a file, a Macro, or a
nice dialplan layout.
In any case, you could call something like this very rapidly
Yes, everyone will have the same message.
You think building the call fill in the spooler is the most effectient?
Can you refer me to a page that will explain pulling the info from a sql db
into a call file?
Last thing, I dial out to an extension, not a registered sip provider, my
provider
I'll answer my earlier question regarding System commands and zombie
ringing channels that can't be killed.
If I start with Answer in the dial plan, I don't have the problem.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
That does not make too much sense to me... Configuration should be ok...
[Aug 8 23:30:13] SSL certificate ok
[Aug 8 23:30:13] == Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
[Aug 8 23:30:13] WARNING[23835]: tcptls.c:463 ast_make_file_from_fd:
FILE * open failed!
dear list
I'm very new in telephony and asterisk especial. so, it would be great
if somebody see at my dialplan. it works (except e1 which is untested at
this day), but may be any caveats here? thank you
alexander
p.s. in Russia national prefix is 8 and international 810
Hey Guys,
Asterisk Trunk had some changes made about 7-8 months ago, that is also
in 1.6.x that
hindered the GUI from working (because of more strict config file
writing, which
is exactly what we needed to not fall into situations we had before).
This work was done for one main reason:
When
Warning: Cannot modify header information - headers already sent by (output
started at /home/telecom/public_html/vicidial/admin.php:1175) in
/home/telecom/public_html/vicidial/admin.php on line 1187
Warning: Cannot modify header information - headers already sent by (output
started at
Thanks, It works now!
but i get this warning as well: Running DeadAGI on a live channel
will cause problems, please use AGI
is it serious? what problems will occur!??
On Fri, Aug 8, 2008 at 11:30 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote:
Try DeadAGI and it should work..
Paradise Dove
On Fri, 2008-08-08 at 23:00 +0300, Tzafrir Cohen wrote:
On Fri, Aug 08, 2008 at 10:16:31AM -0500, Carlos Chavez wrote:
My office Asterisk box has a TDM04B card for three land lines and a GSM
gateway. I have noticed that the Zap channels get stuck a couple times
a week and I have to
I don't really know :)
I run DeadAGI with that error for many months now and nothing ever happened
Paradise Dove wrote:
Thanks, It works now!
but i get this warning as well: Running DeadAGI on a live channel
will cause problems, please use AGI
is it serious? what problems will occur!??
On
When I try to place a call I'm getting an error:
WARNING[16224]: channel.c:780 channel_find_locked: Avoided initial deadlock for
'0x818a3b8', 9 retires!
I'm running asterisk-1.2.27 on Gentoo, VIA mini-itx board.
I'm running the same asterisk version on two other x86 computer and did not see
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