I wish to set up two PSTN calls and then connect them similar to Jajah (is
this called 3pcc?). The PSTN interconnect is handled by a third party SIP
provider.
I can do this using the manager or call files. An example (using php) would
be:
fputs($oSocket, "Action: login\r\n");
fputs($oSocket,
Just in case anyone is looking at these for their call-centre or
similar, you *cannot* put an auto-answer call through to the headset -
it always is directed to the speakerphone.
This has been confirmed by AAstra support.
Obviously, this causes problems in a call centre !
Julian
Julian Lyndon
Hi,
I update from asterisk 1.2 to 1.4 and I have some problems.
In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a call
from an external providers
now in 1.4 I recieve only one ring
What can I do to solve this problem?
___
--Bandwidth an
On Fri, Oct 12, 2007 at 05:24:29PM -0500, Perssy Llamosas wrote:
> Hi,
>
> I am looking for a syntax highlighter for AEL2. Google is not helping,
> so I thought you guys could help me.
>
> I found this vim syntax highlighter for AEL but it doesn't help if you
> want to code in AEL2:
> http://vi
On Fri, Oct 12, 2007 at 05:29:24PM +0200, Philipp Kempgen wrote:
> Atis Lezdins wrote:
>
> > I have 8-core system that has web interface + sql + java + some other stuff
> > running, and at 30 simultenous calls i get loadavg maximum of 3.
>
> I wouldn't be too happy about a system with a
> loadav
Adrian Marsh wrote:
interested in how people are "clustering" Asterisk, if that's possible,
or how you might be achieving a redundant solution.
> I've a single Asterisk server driving the company. Its well backed-up, and
> I've a cloned machine that (in theory) with a DNS change could take over
On Fri, 12 Oct 2007 09:42:47 -0800, "Mojo with Horan & Company, LLC"
<[EMAIL PROTECTED]> wrote:
>Since you are using the OpenVOX FXO card, don't you need another
>module? I'm guessing you'd need wctdm INSTEAD of ztdummy.
Thanks. I've seen it mentionned in some articles, but I'm still in the
dar
On Fri, Oct 12, 2007 at 11:34:39AM -0400, C. Duncan Hudson wrote:
> I'm fairly new to Asterisk, so please bear with me if this is silly
> question. I'd like to run a script on my server that would take the
> "Call now to order" banner off my website automatically when I put my
> phone system on
Il Neofita wrote:
> In the extensions I used DIAL(SIP/100&SIP/101,30,tTr) if I receive a
> call from an external providers
Remove the r
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
__
I'm using Swedish on version 1.4.13. The full part of the
log is:
[Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any
format
[Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8
(alaw)): No such file or directory
The word 'ett' means 'one'. We
On Sat, Oct 13, 2007 at 04:41:11AM +0200, Philipp Kempgen wrote:
> > Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's
> > 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use
> > the jumper settings.
>
> ztcfg -vv
> will tell you the number of channel
On Fri, Oct 12, 2007 at 03:55:39PM +0200, Vincent wrote:
> Hello
>
> 1. I don't have deep knowledge of either Linux or Asterisk, but I seem
> to have successfully installed 1.4 with Zaptel (for support for an
> OpenVox PCI FXO card) on a stock Ubuntu 7.04 Server Edition:
I assume that this is a X
Am Samstag, den 13.10.2007, 15:01 +0200 schrieb Turbo Fredriksson:
> I'm using Swedish on version 1.4.13. The full part of the
> log is:
>
> [Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any
> format
> [Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (
Tzafrir Cohen wrote:
> The loadavg is the average number of threads[0] ready to run (or running).
To me it seems that there are important differences between
systems, especially Linux/Unix, as of which of the states in
following are counted in:
- running (i.e. using the CPU)
- runnable (i.e. wait
On Sat, Oct 13, 2007 at 03:31:09PM +0200, Philipp Kempgen wrote:
> Tzafrir Cohen wrote:
>
> > The loadavg is the average number of threads[0] ready to run (or running).
>
> To me it seems that there are important differences between
> systems, especially Linux/Unix, as of which of the states in
>
Turbo Fredriksson wrote:
> I'm using Swedish on version 1.4.13. The full part of the
> log is:
>
> [Oct 13 12:51:16] WARNING[7810] file.c: File digits/ett does not exist in any
> format
> [Oct 13 12:51:16] WARNING[7810] file.c: Unable to open digits/ett (format 0x8
> (alaw)): No such file or di
D4rk F1ber wrote:
> So I have my asterisk box up and working internally at home and all is
> good so far. The next thing I wanted to do was make and recieve calls
> to regular land lines now.
>
> I don't have a POTS line and was looking for probably a SIP trunk.
>
> I have seen mentions of Skype
Jeng Yu wrote:
> Hi All,
>
> I have a quick one for you. Is there a way to mask
> (i.e. combine) the flags in the Dial() application? In
> other words, a way to do something like
>
> Dial(Zap/1,10,d|t|f)
>
> to get the effects of the three flags together in one
> shot? I have a need to combine t
> "Anselm" == Anselm Martin Hoffmeister <[EMAIL PROTECTED]> writes:
Anselm> You could also copy the file "en.gsm" which should exist
Anselm> there over to "ett.gsm" - wrong reading will result, but I
Anselm> guess people understand what is meant, like they would
Anselm> underst
On Sat, 13 Oct 2007, Lee Jenkins wrote:
> I have been using axVoice.com for some about 9 month to a year now and
> their service is pretty damn good. For home users they have unlimited
> plan for around 22.00-24.00 U.S. per month.
I think the "pay as you go" plans make more sense for most people
Turbo Fredriksson wrote:
> Any idea where the current
[swedish]
> files come from? I.e. who recorded
> them/whos voice it is?
Only you can tell where you got the sound files you use.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use I
Steve Edwards wrote:
> On Sat, 13 Oct 2007, Lee Jenkins wrote:
>
>> I have been using axVoice.com for some about 9 month to a year now and
>> their service is pretty damn good. For home users they have unlimited
>> plan for around 22.00-24.00 U.S. per month.
>
> I think the "pay as you go" plans
And my experience with the "unlimited" plans is after a certain point
-- which is sometimes quite obscure -- they start charging --
sometimes at a rather high rate, so be careful with those. Unlimited
means whatever I want it to mean!
on Saturday 10/13/2007 Lee Jenkins([EMAIL PROTECTED]) wrote
On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote:
> Steve Edwards wrote:
> > On Sat, 13 Oct 2007, Lee Jenkins wrote:
> >> I have been using axVoice.com for some about 9 month to a year now and
> >> their service is pretty damn good. For home users they have unlimited
> >> plan for around 2
I used packet 8 with ATA's and a 4 port card.
$19.99 a month and works great.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Be
I can't seem to get the [s]tart to work in my extensions...
- s n i p -
[default]
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-BUSY,1,Voicemail(${EXTEN}, b)
exten => 2403,1,Dial(sip/${EXTEN},20,t)
exten => _X.,2,Playback(pbx-invalid)
- s n i p -
If I dial '2403' with is off-
John Millican wrote:
> On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote:
>
> Be sure to read the fine print as most of the "unlimited" plans do actually
> have a limit on usage (even the ones I offer). Some are out in the open some
>
Then don't advertise it as *unlimited*
Seems si
That's kinda high then. I wouldn't be happy about that either. You
shouldn't be over 30% ever for anything real time. Instantaneous spikes can
really start to make your life miserable at that point.
-Original Message-
From: Erik Anderson [mailto:[EMAIL PROTECTED]
Sent: Friday, October
Turbo Fredriksson wrote:
> I can't seem to get the [s]tart to work in my extensions...
>
> - s n i p -
> [default]
> exten => s,n,Goto(s-${DIALSTATUS},1)
The first priority in an extension must be 1 not n.
> exten => s-BUSY,1,Voicemail(${EXTEN}, b)
Don't put any spaces in the app data.
On 10/13/07, Turbo Fredriksson <[EMAIL PROTECTED]> wrote:
>
> Setting debug shows that it skipps the 's' parts...
>
> Why?
because you don't seem to have
exten => s,1,something.
--
___
--Bandwidth and Colocation Provided by http://www.api-digital.
Turbo Fredriksson wrote:
> I can't seem to get the [s]tart to work in my extensions...
>
> - s n i p -
> [default]
> exten => s,n,Goto(s-${DIALSTATUS},1)
> exten => s-BUSY,1,Voicemail(${EXTEN}, b)
>
>
The extensions 's' needs to start with a priority of 1. For example:
exten => s,1,A
How do I know when bridging is effective, Francis?
A.
Anthony Francis <[EMAIL PROTECTED]> wrote: That is brought to you by the sip
reinvite, in short yes, unless you set
canreinvite = no to either side of that.
Apa Minerala wrote:
> Am I correct in understanding that if the call comes in g729
> "Philipp" == Philipp Kempgen <[EMAIL PROTECTED]> writes:
>> files come from? I.e. who recorded them/whos voice it is?
Philipp> Only you can tell where you got the sound files you use.
I thought they came with Asterisk (v1.4.13).. Sorry, that was a separate
package Got the suppl
On Fri, Oct 12, 2007 at 11:09:59PM +0200, F6HQZ wrote:
> Check if you have a ground loop.
> If yes, this is probably the cause of this hum.
> Open the loop.
Actually, hum involving analog POTS lines is usually the result of the
line becoming unbalanced to ground.
Cheers,
-- jra
--
Jay R. Ashwort
Alan Lord wrote:
> [CC] chan_zap.c -> chan_zap.o
> chan_zap.c: In function ‘process_zap’:
> chan_zap.c:11149: internal compiler error: Segmentation fault
> Please submit a full bug report,
> with preprocessed source if appropriate.
> See http://gcc.gnu.org/bugs.html> for instructions.
> make[1
Jay R. Ashworth wrote:
> On Fri, Oct 12, 2007 at 11:09:59PM +0200, F6HQZ wrote:
>
>> Check if you have a ground loop.
>> If yes, this is probably the cause of this hum.
>> Open the loop.
>>
>
> Actually, hum involving analog POTS lines is usually the result of the
> line becoming unbalanced
Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
> Turbo Fredriksson wrote:
>
>> I can't seem to get the [s]tart to work in my extensions...
>>
>> - s n i p -
>> [default]
>> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> The first priority in an extension must be 1 not n.
Actually, I did. I just
Turbo Fredriksson wrote:
> Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
>
>> Turbo Fredriksson wrote:
>>
>>> I can't seem to get the [s]tart to work in my extensions...
>>>
>>> - s n i p -
>>> [default]
>>> exten => s,n,Goto(s-${DIALSTATUS},1)
>> The first priority in an extension must be
> "PvK" == Philipp von Klitzing <[EMAIL PROTECTED]> writes:
PvK> Some of the bigger MFC printer/copy/fax combo devices by Brother
PvK> (and maybe also other vendors?) provide a fax-via-smtp feature
PvK> and can built fax networks that way.
As far as I can tell, the Brother boxes require the u
Yes, Jay and Philip, you are right, but you can also have hums if ground
cables for chassis protection against electrical hazards are making loops,
if certain of them are in parallel and if they have different length between
the equipments to protect. Many audio stages (unbalanced side, not balance
Uuugh..for the life of me, i cannot delete sound files using
"EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)"
through AGI
the AGI debug log indicates the command executes successful ( equals 0)
but my files are clearly still there.
If i try System(rm ...) in my extensions.conf diaplan it'll w
Thx a lot for response. pbx_exec is very useful.
Pirlouwi.
2007/10/12, Tilghman Lesher <[EMAIL PROTECTED]>:
>
> On Friday 12 October 2007 04:28:42 Pirlouwi wrote:
> > I wonder if there is a way to build my own asterisk application (let us
> say
> > apps/app_myappl.c),
> > and to launch other exist
Dominic Son wrote:
> Uuugh..for the life of me, i cannot delete sound files using
> "EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)"
>
> through AGI
agi show exec
Usage: EXEC
Executes with given .
So I'd try
EXEC System rm foo
or
EXEC System rm\ foo
But I don't understand why yo
On Sat, 13 Oct 2007, Turbo Fredriksson wrote:
>> Turbo Fredriksson wrote:
>>
>>> I can't seem to get the [s]tart to work in my extensions...
When a dialplan doesn't work as you expect, "show dialplan [context]" is
your friend.
Reply with
show dialplan default
and you may get a more sp
Kevin P. Fleming wrote:
> As the message says, this is a bug in your compiler, and should be
> reported to the packager (or in this case, since you are using LFS,
> directly to the GCC maintainers). I will tell you that we have been
> building Asterisk 1.4 with GCC 4.2.1 for quite a while without
Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
>> exten => s,1,Answer()
>> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> This still doesn't make sense because you did not Dial()
> before jumping based on ${DIALSTATUS}.
Ok, make sense. But still no go:
- s n i p -
[default]
exten => s,1,Answer()
On Saturday 13 October 2007 18:08:49 Turbo Fredriksson wrote:
> Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
> >> exten => s,1,Answer()
> >> exten => s,n,Goto(s-${DIALSTATUS},1)
> >
> > This still doesn't make sense because you did not Dial()
> > before jumping based on ${DIALSTATUS}.
>
> Ok, make
tried both as suggested...though AGI says it's succesful:
AGI Rx << EXEC System rm /var/lib/asterisk/sounds/abandons.gsm
AGI Tx >> 200 result=0
the abandons.gsm file is still there...
i have to delete it through my agi because i'm recording sounds, and i
want users to hear their recording and re
Turbo Fredriksson wrote:
> Quoting Philipp Kempgen <[EMAIL PROTECTED]>:
>
>>> exten => s,1,Answer()
>>> exten => s,n,Goto(s-${DIALSTATUS},1)
>> This still doesn't make sense because you did not Dial()
>> before jumping based on ${DIALSTATUS}.
>
> Ok, make sense. But still no go:
>
> - s n i
Dominic Son wrote:
> tried both as suggested...though AGI says it's succesful:
>
> AGI Rx << EXEC System rm /var/lib/asterisk/sounds/abandons.gsm
> AGI Tx >> 200 result=0
>
> the abandons.gsm file is still there...
Umm, then I don't know what's going wrong.
> i have to delete it through my agi
On Saturday October 13 2007 12:47 pm, Doug Lytle wrote:
> John Millican wrote:
> > On Saturday October 13 2007 12:09 pm, Lee Jenkins wrote:
> >
> > Be sure to read the fine print as most of the "unlimited" plans do
> > actually have a limit on usage (even the ones I offer). Some are out in
> > the
Jeng Yu wrote:
> I would like to hear if anyone out there in Asteriskland has used the
> Dock-N-Talk (DNT) box to connect cell phones to Asterisk box.
The only problem I noticed is that after a random amount of time the box
will lost contact/synch with the cell phone. I'm using DockNTalk for
about
Hi.
You mean to use the AGI funtion in the particular programming
language? yeah. i tried, same results.. : T
i guess i'll have to put it in a database, and flag it to remove
manually for now...
On 10/13/07, Philipp Kempgen <[EMAIL PROTECTED]> wrote:
> Dominic Son wrote:
>
> > tried both as sug
On Saturday 13 October 2007 18:35:28 Dominic Son wrote:
> tried both as suggested...though AGI says it's succesful:
>
> AGI Rx << EXEC System rm /var/lib/asterisk/sounds/abandons.gsm
> AGI Tx >> 200 result=0
The EXEC command takes two arguments, an application name and an
argument. So, it, in fac
Try sip show channels from the CLI
- Original Message -
From: "Scott Moseman" <[EMAIL PROTECTED]>
To:
Sent: Friday, October 12, 2007 6:12 PM
Subject: [asterisk-users] Display channels and codecs
> Is there an easy way to show all active channels AND the codecs
> they're using? Other th
On Sat, Oct 13, 2007 at 05:45:26PM -0700, Dominic Son wrote:
> Hi.
>
> You mean to use the AGI funtion in the particular programming
> language? yeah. i tried, same results.. : T
I guess that this is a permissions issue. Check what you get in the
standard error.
--
Tzafrir Cohen
I don't seem to be able to find the necessary hardware
specs for an Asterisk server. What I have in mind is a
dedicated server to serve 50 or so people. All users
will use SIP phones and there will be an ISDN gateway
for outgoing/incoming calls. Do you have any
suggestions about the server specs (C
57 matches
Mail list logo
