Jeremy McNamara wrote:
Lubomir Christov wrote:
BUT I have the say that I have the same opinion as martin
([EMAIL PROTECTED]): Although personally I would prefer oh323 for
its very well described config file for now winner is chan_h323
Again, what is not clear about h323.conf? It follows
Hi
Is there any work being done on implementing IM/SIMPLE support
for SIP on Asterisk? Like a presence server?
rdgs,
/Staffan Kerker
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all.
In order to dial an IAXTEL number, even you have registered using
user/pass/iaxtel.com, at the end of the dial string a @iaxtel context must
be appended.
I will do this automatically in version 0.9.5
In the mean time, you can only dial IAXTEL numbers from the phonebook.
Sorry for the
can u guys see what is the problem here. i am a
newbee:D and i want help.
my conf files for reference:
...iax.conf
[general]
register = hsbohra:[EMAIL PROTECTED]
[NuFone]
type=user
secret=mysecret
context=nufone1
[NuFonePeer]
type=peer
secret=mysecret
context=usacall
24 2003 21:27 Jeremy McNamara :
I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
I have asterisk-oh323-0.5.7.tar.gz and * from CVS @ 20 Nov 2003.
PWLib 1.5.2, OpenH323 1.12.2
ATA-186(h.323)-gnugk-*-7940(SIP)
I see segmentation
asterisk*CLI load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so = (unixODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
-- cdr_unixodbc: username is root
Adam Hart wrote:
From: Jeremy McNamara [EMAIL PROTECTED]
I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
Be brutal. I want to know the gory details, so we can stop any future
pissing matches from even starting by having everything
Max Tulyev wrote:
24 2003 21:27 Jeremy McNamara :
I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
I have asterisk-oh323-0.5.7.tar.gz and * from CVS @ 20 Nov 2003.
PWLib 1.5.2, OpenH323 1.12.2
ATA-186(h.323)-gnugk-*-7940(SIP)
I
Daniel Chabrol wrote:
Hi List!
I get WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
524300 is not codec1 = 524300, can't do reinvite at my asterisk console.
The code there looks realy strange:
codec0 = pr0-get_codec(c0);
codec1 = pr1-get_codec(c1);
I don't know if they're finished yet, but Klaus-Peter Junghanns is
working on some quad BRI passive HFC-PCI cards that will probably
'killall -9 avm diva' both in functionality (native drivers for
asterisk) and in price.
roy
On Tue, 2003-11-18 at 17:01, Ray Burkholder wrote:
What types of ISDN
Hi,
-Original Message-
There is no sound on either side of the call, so I guess that
qualifies as
'it
doesnt work for me' :-P
Do you have:
Disallow=All
Allow=GSM
In the section of iax.conf for the user using DIAX ?
Hey, you're right, that helps! But I don't
If it was possible to get any support at all from Jeremy (or others),
I'd be glad to use it. I have sent numerous reports with where it failed
and what I did to remedy this without getting any response. With
developers this arrogant, chan_h323 should be removed from the asterisk
tree, unless the
Hi,
I am a beginner to Asterisk. Can anybody clear my following doubts regarding
the configuration needed?
1) What is the ideal system configuratin required?(like processer, RAM, h/d
space etc)
2) How many connections it can handle at a time?
3) How many Virtual PBXs it can handle?
4) Whether
Jeremy McNamara wrote:
As history shows I was totally blown off by Michael when I offered to
help better his driver. Then I was even told that I couldn't create
anything better...hence the birth of chan_h323 and this whole mess.
Yes, sure, whatever.
Jeremy McNamara
Michael.
check both directions
when you do a show channel ... does it show gsm in both ways ?
Florian Overkamp wrote:
Hi,
-Original Message-
There is no sound on either side of the call, so I guess that
qualifies as
'it
doesnt work for me' :-P
Do you have:
Disallow=All
On Tue, 25 Nov 2003, Girish Gopinath wrote:
1) What is the ideal system configuratin required?(like processer, RAM, h/d
space etc)
depends (on what codec, what users, whats the purpose)
2) How many connections it can handle at a time?
depends on 1 above
3) How many Virtual PBXs it can
Hi all,
I'm wondering how I could go about placing * into an existing office PBX
system so as to capture callerid for further processing via AGI into an
intranet app.
I've already got the AGI scripts set up for what I want to do and have
tested using IAX2 peering, however I have no knowledge of
From: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: Girish Gopinath [EMAIL PROTECTED]
CC: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * Configuration
Date: Tue, 25 Nov 2003 15:24:11 +0500 (PKT)
On Tue, 25 Nov 2003, Girish Gopinath wrote:
1) What is the ideal system configuratin
I'm running an E400P. Every now and then Asterisk stops receiving
incoming calls.
This turns up in the messages log:
Nov 25 10:49:12 WARNING[65541]: File chan_zap.c, Line 5793
(pri_dchannel): Ring requested on channel 1 already in use on span 1.
Hanging up owner.
Nov 25 10:49:15
Hi
I have a problem with the fax detection.
I want to be able to turn that of on all zap channels.
the * is in between my E1 and my PBX and when I try to make a fax call out
on the E1 the * detects the fax tone and hangsup the outgoing zap channel.
How should I solve this ?
/Mike
Brian!
You've done something tricky again! I'm interested! *wink*
Where do I get it?
;)
Ben
__
Benjamin Wakefield
[EMAIL PROTECTED]
http://www.dcsi.net.au/
DCSI - We do Internet.
64 Queen Street
Warragul, VIC 3820 AU
Ph: (+61) 1300 665 575
Fx: (+61) 1300 556 595
Brian West wrote:
asterisk*CLI load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so = (unixODBC CDR Backend)
== Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
-- cdr_unixodbc:
Pavel Litvinenko wrote:
Brian West wrote:
asterisk*CLI load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so = (unixODBC CDR
Backend)
== Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
-- cdr_unixodbc: dsn is MySQL-asterisk
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
asterisk*CLI load cdr_unixodbc.so
Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so = (unixODBC CDR
Backend)
== Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing
'/etc/asterisk/cdr_unixodbc.conf': Found
--
Ok the basic requirement is unixODBC and the MyODBC driver(for MySQL) or
you can pick what ever you want(http://www.unixodbc.org/drivers.html).
The table structure is the same as pgsql or mysql ... just duplicate that.
I would like to verify that I have done this in such a way that the
database
Dan
I seem to be having the same problem as some of the other guys. With all
the previous versions I could make outgoing and receive incoming calls;
however with this latest version even if I have Diax open the call drops
through to the busy priority in my extensions.conf file. It's like
From a post to this very thread just a few hours ago:
exten = _9.,1,ChanIsAvail(Zap/1Zap/2)
exten = _9.,2,Dial(${AVAILCHAN})
exten = _9.,102,NoOp
exten = _9.,103,Congestion
Kind of a side question... what does ChanIsAvail actually check in the
above (loop attached, dialtone, or just
I have a number of Zap/ extensions defined in a queue with ringall
strategy. When this queue is called sometimes Asterisk seems to think
that one of these channels is busy, while it is NOT. The following is
shown on the console:
--Called 44
-- Called 36
-- Called 41
-- Called 35
Isamar Maia wrote:
Why don't you put it on the -dev list. Even if most of us might not be
able to help much, we could watch and learn.
The issue to create several lists was already decided and implemented?
If so, let me know since I didn't get this thread.
I implemented routines to
Hi Brian,
Excellent job, but how about calling the application 'cdr_odbc' instead of
'cdr_unixodbc', because up to now 'unix' is obvious/trivial when it comes to
* isn't it? Besides, I think 'cdr_odbc' is more in line with cdr_mysql and
cdr_csv and what have you ...
Keep it up !
Hans
On Tuesday 25 November 2003 03:19, Roy Sigurd Karlsbakk wrote:
If it was possible to get any support at all from Jeremy (or others),
I'd be glad to use it. I have sent numerous reports with where it
failed and what I did to remedy this without getting any response.
With developers this
http://www.bkw.org/~brian/cdr_unixodbc.tar.gz
asterisk root # cd /usr/src/
asterisk src # tar zxfv cdr_unixodbc.tar.gz
cdr_unixodbc/
cdr_unixodbc/cdr_unixodbc.c
cdr_unixodbc/Makefile
cdr_unixodbc/mkdep
cdr_unixodbc/cdr_unixodbc.conf.sample
asterisk src # cd cdr_unixodbc
asterisk cdr_unixodbc #
I called it that because i'm using the unixODBC libs. I guess I can change
that! :P I just posted the code and install instructions to the list.
Also if i'm thinking correct this will sidestep the issue with mysql and
gpl since unixODBC is lgpl?
bkw
On Tue, 25 Nov 2003, Vledder, Hans wrote:
Joseph Finley wrote:
I'm not sure if I am wording this correctly, but I'll try.
I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap
analog phones plugged into the FXS ports. I am able to get * to ring those
phones when a call comes in, but I cannot get the phones to dial
On Tuesday 25 November 2003 07:50, Vledder, Hans wrote:
Excellent job, but how about calling the application 'cdr_odbc'
instead of 'cdr_unixodbc', because up to now 'unix' is
obvious/trivial when it comes to * isn't it? Besides, I think
'cdr_odbc' is more in line with cdr_mysql and cdr_csv and
- Original Message -
From: Girish Gopinath [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 10:22
Subject: [Asterisk-Users] * Configuration
Hi,
I am a beginner to Asterisk. Can anybody clear my following doubts
regarding
the configuration needed?
1) What is
Hi,
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 3:37 PM
Subject: Re: [Asterisk-Users] New DIAX - version 0.9.4 - a big step
forward - available for download
Dan
I seem to be having the same problem as some of the other guys.
Brian
Good job!!! Is there any perfomance hit by using unixodbc as oppossed to for
example using cdr_mysql for mysql?
- Original Message -
From: Brian West [EMAIL PROTECTED]
Date: Tue, 25 Nov 2003 07:19:27 -0600 (CST)
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] cdr_unixodbc
Dave,
I'm wondering how I could go about placing * into an existing office PBX
system so as to capture callerid for further processing via AGI into an
intranet app.
If I understand you, all you want to do is use an incoming callerid to trigger
an event in your system. If I got that right, it
Pat,
What i'm trying to figure out is how to keep the sql statement
globally prepaired then just call SQLExecute but the docs for all this are
hard to come by. I really can't tell much diffrence in odbc over mysql in
speed but I don't have a bazillion calls going thru at once. It does
Then i started asterisk, it opens the D-Channel and
everything is still ok. I left the system in this state
and it survived one night without problems. But immediately
after the first call (B-Channel) the system memory is overwritten
and bad things happen.
I looked through the wct1xxp.c and
You're forgetting to answer the line first.
Mark
On Fri, 21 Nov 2003, WipeOut wrote:
I am trying to use the SAY NUMBER command from an AGI script but it does
not seem to be working..
If I use EXEC SayNumber 2 and execute the asterisk command from the
AGI it works and I hear the 2 said on
DTMF is used in some places. Japan uses FSK, but a rather different
message format. There isn't a whole lot of global standardisation in CLI!
Not only that but I believe they use different frequencies, and utilize a
parity bit as well.
Mark
___
Yep, we use it for international calling. Works great:
exten = _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)
How are you achieving that? If I am on a regular FXS connected phone that
line would match 90115, thus preventing me from getting the rest of the
phone number (620508132).
Now if dialing from
I'm wondering how I could go about placing * into an existing office PBX
system so as to capture callerid for further processing via AGI into an
intranet app.
If I understand you, all you want to do is use an incoming callerid to
trigger
an event in your system. If I got that right, it sounds
befor answering these questions, ask yourself the
following questions:
what is it that you want to achieve with your
deployment?
How many users are you planning to service with
Asterisk?
Are you trying to integrate with an existing PBX?
If this installation is a stand-alone, how do you wish
Does anybody have useful tips on creating good quality recordings for
use with prompts in asterisk? I'm interested in hearing input on
hardware (mics, dats, sound cards, etc) and software (recording
software, dsp) as well as recording techniques.
Jerimiah
Tularosa Communications
Hi,
I'm interested. I'm running chan_capi 0.3.0 with Fritz PCI ISDN card. Using
DIAX as softphone and dialing out to PSTN generally results in good sound
quality at softphone end (no echo), but PSTN end experiences quite a bit of
echo. I have enabled echosquelch in capi.conf, but it does not seem
Hi,
-Original Message-
check both directions
when you do a show channel ... does it show gsm in both ways ?
Yes:
vectra*CLI show channel IAX2[florian]/14
-- General --
Name: IAX2[florian]/14
Type: IAX2
UniqueID: 1069774221.220
Caller ID: 651154495
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 25 November, 2003 08:56
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for
outbound calls
Yep, we use it for international calling. Works great:
exten
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 25 November, 2003 08:56
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for
outbound calls
Yep, we use it for international calling. Works great: exten =
On Tuesday 25 November 2003 09:56 am, Andrew Kohlsmith wrote:
Yep, we use it for international calling. Works great:
exten = _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)
How are you achieving that? If I am on a regular FXS connected phone that
line would match 90115, thus preventing me from
On Friday 21 November 2003 12:36 pm, marrandy wrote:
Reason.
I have a fax/ans phone with handset, that lets you monitor the caller, so if
you wish, you can pickup the call.
The asterisk is undergoing testing, it will then be online tested at the
house
so I can get more familiar in
Hi Michiel-
This may be related to a PRI frame buffer overflow problem that I get in
high-volume IVR applications. I get a lot of these errors mixed in with
frame errors. In my case its load related. Mark and Martin at Digium have
said they'll be looking into improving the buffering
Just an FYI I have cdr_unixodbc doing inserts using Text file driver
now
bkw
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
First at alll, I beg your pardon because maybe I explained bad my questions
(because my low level english)
I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2
compiled and installed.
I have modules alsa 0.9.8 compiled and installed
My PC has an audio card ac97
Do you have up to date libpri and asterisk ?
Also it'd be good if you could send pri debug span 1 (or 2) trace.
regards
Martin
On Tue, 25 Nov 2003, Alastair Maw wrote:
I'm running an E400P. Every now and then Asterisk stops receiving
incoming calls.
This turns up in the messages log:
OK, that was obviously a 'typo' ... It's fixed.
Martin
On Tue, 25 Nov 2003, Detlef Wengorz wrote:
Daniel Chabrol wrote:
Hi List!
I get WARNING[14351]: File rtp.c, Line 1202 (ast_rtp_bridge): codec0 =
524300 is not codec1 = 524300, can't do reinvite at my asterisk console.
The
My Zap channels having the problems are on a T1 connected to a CAC
channelbank, But it looks like the zt_rec in chan_zap error uses the
lowlevel zaptel ioctl's which are the same for T1 PRI...
Scott Stingel wrote:
Hi Michiel-
This may be related to a PRI frame buffer overflow problem that I
Hi!
I've contacted Grandstream directly via email and received a reply in
one day with prices and an order form to fill... Nice customer service!
--
Sérgio
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ariel Batista
Sent: segunda-feira, 24 de
On Tuesday 25 November 2003 08:56, Andrew Kohlsmith wrote:
Yep, we use it for international calling. Works great:
exten = _9011.,1,Dial(Zap/g0/${EXTEN:1},,t)
How are you achieving that? If I am on a regular FXS connected
phone that line would match 90115, thus preventing me from getting
Here are some links.
http://www.supermicro.com/PRODUCT/MotherBoards/GC_SL/X5SSE-G.htm
http://www.supermicro.com/PRODUCT/MotherBoards/GC_SL/X5SS8.htm
http://usa.asus.com/products/server/srv-mb/nrl-ls533/overview.htm
http://www.tyan.com/products/html/trinitygcsl.html
-John
On Tue, Nov 25, 2003 at
On 25/11/03 16:58, Cristian Vasiliu wrote:
Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
any motherboard with PCI 3.3 . Any sugestions!?
Wait for the TE405P to appear, which is a 5V version of the TE410P. It
should be shipping in the next week or two.
Alastair
On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
Does anybody have useful tips on creating good quality recordings for
use with prompts in asterisk? I'm interested in hearing input on
hardware (mics, dats, sound cards, etc) and software (recording
software, dsp) as well as recording
::Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
::any motherboard with PCI 3.3 . Any sugestions!?
::
This really is a problem with the state of flux the PCI bus is currently in
and the comprimises a vendor must make in order to best meet what is
available in the market
FYI: According to Digium, we should have the new 5v Quad T1/E1/PRI (Part#
TE405P) in stock sometime next week.
Sean
- Original Message -
From: Cristian Vasiliu [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 11:58 AM
Subject: [Asterisk-Users] PCI 3.3 V
Why
I think we have to figure out what the difference is. It'll take going
through the mgcp debug output to see what is going on.
Mark
On Mon, 24 Nov 2003, ProvoCityPower wrote:
We are working on a new implementation of asterisk. We are using a fiber-served
WorldWide Packet switch at the home
Compatible motherboards supporting 3.3v PCI are a bit hard to find outside
the US - Tyan makes one (model S2723), and one or two others (Intel?)
But I understand there is a new 5 volt version of the T1/E1 card soon to be
released as well.
-Scott
Scott M. Stingel
Emerging Voice Technology Inc.
- Original Message -
From: marrandy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 10:47 AM
Subject: Re: [Asterisk-Users] Can you monitor a call via the asterisk speaker system
and do a call pickup if you wish
On Friday 21 November 2003 12:36 pm, marrandy
Steven Critchfield wrote:
On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
Does anybody have useful tips on creating good quality recordings for
use with prompts in asterisk? I'm interested in hearing input on
hardware (mics, dats, sound cards, etc) and software (recording
software, dsp)
On Tue, Nov 25, 2003 at 08:51:03AM -0600, Mark Spencer wrote:
Then i started asterisk, it opens the D-Channel and
everything is still ok. I left the system in this state
and it survived one night without problems. But immediately
after the first call (B-Channel) the system memory is
Hi all,
Just wondering if someone have already done something like that :
SIP Client_A --- 1)call --- ASTERISK --- 2)outgoingcall-PSTN--Client_B
|
|
I used windows sound recorder to record it (with noise less background)
used sox to convert to gsm. It turned out pretty good.
Jerimiah Cole wrote:
Does anybody have useful tips on creating good quality recordings for
use with prompts in asterisk? I'm interested in hearing input on
hardware
Cristian Vasiliu wrote:
Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
any motherboard with PCI 3.3 . Any sugestions!?
You have four options:
A 32 bit slot in a Dell 600SC
Almost any 64bit PCI slot (except for a small number of 33MHz only 64
bit slots)
Wait for the soon
Pavel Litvinenko wrote:
Joseph Finley wrote:
I'm not sure if I am wording this correctly, but I'll try.
I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple
cheap
analog phones plugged into the FXS ports. I am able to get * to ring
those
phones when a call comes in, but I
Tom Walsh wrote:
[SNIP]
(hope this doesn't wrap)
http://www1.us.dell.com/content/products/productdetails.aspx/pedge_600sc?c=u
scs=555l=ens=biz
it did ! (lol)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Hi,
I would like to get info about integrating either Teamspeak or Ventrilo with
Asterisk.
Ventrilo and Teamspeak is free voice conferense servers/clients that are
commonly used for online voice conference over internet (IP).
Both has their own clients!
I have had some request about accessing
On Tue, 25 Nov 2003, Tom Walsh wrote:
::Why PCI 3.3V for E1/T1 card!? I can not use it because I can not find
::any motherboard with PCI 3.3 . Any sugestions!?
Dell offers a tower that I know for certain has 3.3V PCI bus. PowerEdge
600SC.
Dell PowerEdge 1750 (1U rackmount with up to 2* Xeon
unsubscribe
Robert G. Werner wrote:
The problem with -newbies (or even some PC name for it) is that people
won't use it.
Rarely do people self select themselves as more ignorant than they
really are. I'm afraid the noob problem just can't be resolved with
any structural changes.
- Original Message -
From: Areski [EMAIL PROTECTED]
To: Asterisk-Users Mailing-list [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 12:13 PM
Subject: [Asterisk-Users] Outgoing-call and enter user in Conference - repost
Hi all,
Just wondering if someone have already done
--- Steve Underwood [EMAIL PROTECTED] wrote:
Steven Critchfield wrote:
On Tue, 2003-11-25 at 09:24, Jerimiah Cole wrote:
Does anybody have useful tips on creating good quality recordings
for
use with prompts in asterisk? I'm interested in hearing input on
hardware (mics, dats,
Please read past rants about the action you took to create this message.
Hint: You broke the thread by replying to an unrelated thread.
On Tue, 2003-11-25 at 11:24, e-smith wrote:
Hi,
I would like to get info about integrating either Teamspeak or Ventrilo with
Asterisk.
Ventrilo and
On Tue, 2003-11-25 at 11:57, David Luyens wrote:
Does anyone know of some people having developped a prepaid application
on asterisk?
Please use the Truly lazy way to start a new thread and click on the
mailing list address in a message so that the rest of us who have a clue
don't have to get
Steven Critchfield wrote:
Please read past rants about the action you took to create this message.
Hint: You broke the thread by replying to an unrelated thread.
Could all of the thread police please just reply personally to the
offending party?
The amount of people interested in the rant is
A note to all those who are avoiding writing up an AGI becasue it looks
two complicated..
I have just written up my first and its awesome.. It makes Asterisk open
to all sorts of possibilities.. let your imagination run wild..
I put off writing an AGI script because a) I could not find any
Found some info on the Web that may help some
of the ADSI programmers out there.
The following guide is for a WebSphere implementation
but the average developer type should be able to pull
enough out of it to help writing ADSI scripts for
Asterisk. Seemed to have good overview of ADSI
see below
Michael Manousos wrote:
Max Tulyev wrote:
24 2003 21:27 Jeremy McNamara :
I would like to hear from anyone else that has real world experiences
with both chan_h323 and asterisk-oh323.
I have asterisk-oh323-0.5.7.tar.gz and * from CVS @ 20 Nov 2003.
PWLib 1.5.2, OpenH323 1.12.2
Book costs $49.50
PDF Download - free
--- Jonathan Biggs [EMAIL PROTECTED] wrote:
Found some info on the Web that may help some
of the ADSI programmers out there.
The following guide is for a WebSphere
implementation
but the average developer type should be able to
pull
enough out of
I am working on a project on 3rd party call control for a call center, for
which I think Asterisk may be useful. What I would like to do is:
- Have a call come in to Asterisk.
- Asterisk asks another machine, over a slow IP link, such as a modem, how it
should route the call. Asterisk passes
Hi!
Indeed great move, Brian!
What i'm trying to figure out is how to keep the sql statement
globally prepaired then just call SQLExecute but the docs for all this are
hard to come by. I really can't tell much diffrence in odbc over mysql in
speed but I don't have a bazillion calls
- Original Message -
From: WipeOut [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, November 25, 2003 2:14 PM
Subject: [Asterisk-Users] AGI Rocks!! (A happy camper)
A note to all those who are avoiding writing up an AGI becasue it looks
two complicated..
I have just written
I was just looking at AGI with PHP myself. I just have a real dumb question. How does
Linux know to send $stdout(or echo) to *? What if there are other apps open as well
waiting for input. WOn't they get the output?
Also, how does the AGI know to read from $stdin is * input?
Costas
--
Hi!
- Ventrilo www.ventrilo.com
- Teamspeak www.teamspeak.org
And these accomplish what that asterisk doesn't do already?
Concerning teamspeak:
- graphical client where a channel or sub-channel moderator can perform
all kinds of action like moving users to a different channel, mute
Hi!
I would like to get info about integrating either Teamspeak or Ventrilo with
Asterisk.
For conferencing Teamspeak is great, no doubt. Being able to join TSS
conferences from * would be a great thing indeed. Maybe this is a
suggestion worth to post on the TSS forum as well?
Cheers,
On Tue, 2003-11-25 at 13:51, Alistair Cunningham wrote:
I am working on a project on 3rd party call control for a call center, for
which I think Asterisk may be useful. What I would like to do is:
- Have a call come in to Asterisk.
- Asterisk asks another machine, over a slow IP link, such
I know... bad form to add to my own posting but:
I found out that the Unknown error only appears when ringing multiple
extensions at nearly the same time. When ringing two Zap channels (with
) it takes a little longer but eventually the error will crop up and
one of the ringing channels will
I have finally crashed Asterisk for the first time and I'm wondering if
anyone has seen this.
This is a configuration with SIP endpoints and an IAX2 channel to
another Asterisk PBX.
The main PBX dropped a core file after a SEGV (signal 11 ) with the
following trace:
#0 0x42079133 in strchr ()
Good idea. When do you want it? :P but that does give me an idea.
http://www.bkw.org/~brian/cdr_unixodbc.tar.gz
I have done some cleaning. I added the ability for the cdr driver to
retry the db connection. Like if your sql server went a way and it lost
the connection it will retry the
I want to be able to demo * on a notebook at a client's site. This means no FXO
gateways; just 2 sip phones (like SNOM) and maybe a softphone (GnoPhone?). I already
have RH9 running on my notebook.
I would like to have one SIP phone dial and go through IVR before making a choice and
ringing
Under your situation, Asterisk will play a role with IVR+ACD as well as transfer the calls to your destination you requested, it will not be very difficult to write if you know Asterisk well.
Howard SongAlistair Cunningham [EMAIL PROTECTED] wrote:
I am working on a project on 3rd party call
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