show application monitor in the cli
Matteo.
Il mer, 2003-07-02 alle 09:22, Herv Thibaud ha scritto:
hi
is there a simple way to record a conversation with asterisk ?
thanks
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Benjamin Miller wrote:
Would it be more flexible to approach this differently, with a dtmf to
indicate that the agent is done with wrap up?
So they get off a call and can wrap up the call for as long as
necessary, and then hit * or something that marks them as available
again rather than working
and what the way to play records in the spool
Le mer 02/07/2003 à 09:28, Matteo Brancaleoni a écrit :
show application monitor in the cli
Matteo.
Il mer, 2003-07-02 alle 09:22, Hervé Thibaud ha scritto:
hi
is there a simple way to record a conversation with asterisk ?
thanks
I had a similar problem and solved it changing the params of input
gain on my pstn-gateway, change from a value of 10 to a value of 1 and
that eliminated the echo on the SIP Phones.
Dave Packham wrote:
Same prob here. 15 SIP phones only get eco when going to the PSTN...
if you find
That's a different part of the equation.
If Asterisk could interpret the Via: headers like the Cisco phones
do, that would solve the Asterisk-behind-a-NAT problem to a large
degree. Perhaps it already does; I've never tried putting Asterisk
behind a NAT, only SIP clients.
JT
Please don't
On Wed, 2003-07-02 at 04:26, John Todd wrote:
You may be correct about the Via: header, but you're incorrect in the
concept as to how it relates to Asterisk, notably in your reversal of
what side of the transaction is putting data in the Via: header to
make SIP work correctly.
This is
Date: Tue, 1 Jul 2003 14:37:20 +1000
From: Andrew Radke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] A solution for SIP and NAT
...
So I've started a really simple SIP and RTP proxy project, SaRP, on
sourceforge.net. Yesterday we uploaded 0.2 of the perl based release.
Hi,
What do you mean by pstn-gateway?
There is no input gain parameter in zapata.conf file?
It is about rxgain?
BR,
Dan
- Original Message -
From: Ing. Angel Gomez Garcia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003 11:48 AM
Subject: Re: [Asterisk-Users]
Patrick wrote:
[snip]
Hi John,
[snip]
How about Asterisk and NAT? Can you please comment if the examples below
also work.
1x SIP phone - NAT box - Internet - NAT box - Asterisk
10x SIP phone - NAT box - Internet - NAT box - Asterisk
This all depends on the NAT boxes that you use. The SIP phone.
Hi,
Has anyone worked out a way to use Asterisk in a Hot Desk environment??
I have not been able to think of a way for the user to have control over which IP
phone will ring when that users extension is dialed without the user needing to
reconfigure the phone..
Something like this would be
I have a SIP FXO 8 port VoIP gateway, and it has a parameter called
'input gain' wich is the one I modified, there might be a similar
parameter on the configuration for the hardware you are using.
Dan wrote:
Hi,
What do you mean by pstn-gateway?
There is no input gain parameter in
Klaus Darilion wrote:
[snip]
The project can be found at http://sarp.sourceforge.net/
There is also a similar project called siproxd:
http://sourceforge.net/projects/siproxd/
regards,
klaus
It has a broadly similar goal on the surface but a very very different
approach. siproxd relies on a
Hi evereybody,
I'm trying to use musiconhold during dial
tones.But I only can call earing dial tones instead of music.
Now will see my configuration files.
AGI File(using AGI script to EXEC
DIAL)
print "EXEC Dial Zap/g2/numberc||m\";
$res=checkresult();
Extension.conf
exten
Actually its easier than you think...
Allocate a control extension for each hot desk user.
implement call forward and cancel call forward...
I did this with a ifo storage in the astdb so it holds during a
restart...
your dial plan must be macro'd for this to work properly...
have a look at
How about the logon wizard of the snom 100? I think that does something
simlar to what you want. It's designed to allow different people to login to
a single phone.
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003 11:21 AM
The snom phones (and I assume others) allow you to have multiple SIP
accounts on a single phone. The user logs in to the phone which logs in *.
The downside is that you can only log in to accounts set up on the phone
rather than any account set up on * but is useful for shared desks etc..
W
rxgain and txgain are used, for example with the X100P. As I understand
it, the echo problem with a SIP to PSTN implementation in * has two
components:
- echo resulting from the digital to analogue conversion at the X100P
- acoustic feedback within the handset used
The former is reduced by
Hi
I would like to use the Asterisk PBX as part of a phone shop system instead
of the usual PBX plus PC. How can I do the the billing in a way that is
convinient to the phone shop attendant?
Regards
Shepherd
_
Add photos to your
Hi,
In zapata.conf I have tried to change the rxgain and txgain parameters, but
without any success.
I think it is X100P card driver related issue.
BR,
Dan
- Original Message -
From: Ing. Angel Gomez Garcia [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003 1:40 PM
ok, I'll bite :-)
What the heck is a phone shop system ??
On Wed, 02 Jul 2003 09:48:44 +, shepherd fungayi wrote:
Hi
I would like to use the Asterisk PBX as part of a phone shop system instead
of the usual PBX plus PC. How can I do the the billing in a way that is
convinient to the
You may want to check, but I think rtc is a x86ism and may not be
available to you on a mac.
On Wed, 2003-07-02 at 00:54, Serge Mankovski wrote:
I am trying to compile it under Yellow Dog 3.0 on iMac
I get this error
zaprtc.c:1077: warning: implicit declaration of function `barrier'
The problem with this solution is that you are moving the logic to the phone which can
very quickly become an admin headache.. the more the config and admin can be central
and server based the better..
How about the logon wizard of the snom 100? I think that does something
simlar to what you
Hi all,
I have the M1500 Plantronics now and have done some tests with the Mitsumi
BT USB adapter.
The latest drivers from the Mitsumi site supports Headset Profile too, but I
still cannot use it with my dongle (cannot be activated).
Anyone else succeeded in using a BT dongle with Headset Profile
Hi,
I have just updated to the latest CVS version ( Approx 13:05 GMT Today ).. I am
running chan_capi 0.2.2..
When a call is received Asterisk seg faults.. Not sure what information would be
usefull to post so let me know what info will help to debug the problem..
Later..
--
I'm very interested in the same thing for a hotel system I would like to
implement. Anyone know if the country codes be tied to a pricing lookup
table?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi
Sent: Wednesday, July 02, 2003 5:49 AM
Nevermind... After a reboot it appears to be happy again.. must just be a gremlin that
crept in somewhere..
Later..
Hi,
I have just updated to the latest CVS version ( Approx 13:05 GMT Today ).. I am
running chan_capi 0.2.2..
When a call is received Asterisk seg faults.. Not sure what
Never mind my previous post, A quick gogle search shows there is rtc on
ppc arch.
On Wed, 2003-07-02 at 07:08, Ing. Angel Gomez Garcia wrote:
Hi.
How can I know if rtc support is built into the kernel ?
Steven Critchfield wrote:
You may want to check, but I think rtc is a x86ism
Instead of this make notes of some of the faults in SIP that cause you
problems and start working towards SIP/2.1 or SIP/3.0. Just because you
weren't one of the people involved in designing the existing protocol
doesn't mean you can't work to change it.
SIP 2.0 has some unbeleivably braindead
Andrew Radke wrote:
Ok I guess it's time for me to weigh in on this since I started the
whole thing and am the main developer of SaRP.
NAT and SIP _can_ work okay under very very restricted circumstance.
Multiple SIP UAs behind one NATed IP _can_ work okay with a very
intelligent
Shepherd-
Having designed one of these in the past (in a higher level voice
environment), I can tell you that this is not a small undertaking. It's at
least as much an SQL job as a voice task.
Usually the way to accomplish this is to establish more-or-less a pre-paid
phone card system, where
That would be excellent. Thanks.
Jim Friedeck
Mark Spencer wrote:
Could probably make '#' terminate wrapup time immediately or something.
Mark
On Tue, 1 Jul 2003, Jim Friedeck wrote:
Mark,
How hard would it be to write a simple app to cancel wrap-up via an
Hi Justin,
Try:
exten=242,1,Dial(h323/[EMAIL PROTECTED])
Regards,
Szymon Czyz
Justin Eckhouse [EMAIL PROTECTED] wrote:
Hi,
I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting h.323. I'm using chan_h323.so, and I'm able to make
outbound
Wrap-up, as our existing phone system calls it, is a period of time that
an agent will not get an incoming call after hanging up the previous
call. This allows time for the agent to 'wrap-up' the preceeding
conversation by filling out forms, typing on the computer, or taking a
sip of coffee.
i think that the problem could be something more easy:
it is possible inside asterisk to log all che calls of all the users
and know the timing and the number called for each call?
if it is possible to do that, could be possible to make a program
that takes this files and generate the costs
We had the same problem, we fixed it downgrading the Capi version to
0.2.1b
Salut,
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de WipeOut .
Enviado el: miércoles, 02 de julio de 2003 14:22
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Seg
That's all I would need, it would be easy enough to work out the cost after
that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angelo Sampietro
Sent: Wednesday, July 02, 2003 10:06 AM
To: Scott Stingel
Cc: [EMAIL PROTECTED]
Subject: Re[2]:
There is a CDR (Call Detail Record) which is accessible in two different
ways. The first is via a simple comma delimited file which can be parsed
and fed into whatever database that you want. The second way is to dump
the CDR directly into MySQL, and extract accordingly. So the only trick
there is
We had the same problem, we fixed it downgrading the Capi version to
0.2.1b
what's the diff between 0.2.1b and 0.2.2 ?
regards
Marian
Salut,
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de WipeOut .
Enviado el: mircoles, 02 de julio de
Hi All,
Does anyone know if there has been any developments in asterisk to use LineJacks to
dial out (connect to the PSTN)?
The card works perfectly with virtually anything else but asterisk.
Maybe the CVS versions have some work on it?
Cheers,
-Z
--
thanks a lot!
can you tell me where can i find more info about the CDR?
probably this will be the better way to give to the company a summary
with all the phone traffic :)
Angelo
Thursday, July 3, 2003, 4:37:32 PM, you wrote:
KCC There is a CDR (Call Detail Record) which is accessible in two
Which driver are you using?
Zara Trousk wrote:
Hi All,
Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)?
The card works perfectly with virtually anything else but asterisk.
Maybe the CVS versions have some work on it?
Cheers,
Is it just me or do others have a problem with the ATA-186
de-registering? Every couple of hours, if I dont make use of the ATA
connected line, I find that I have to unplug and let the ATA reboot. After that
it is good to go for awhile, but eventually I have to repeat the process. My
ATA
I have a two remote PBXs. I use the switch = statement on each PBX to
point to the other PBX.
Now I want extensions on PBX-1 to dial extensions and PSTN numbers that
are local to PBX-2.
However, I ALSO want people to be able to dial into a Zap channel on
PBX-1 and be able to dial extensions on
On Wed, 2003-07-02 at 10:36, Eric Wieling wrote:
I have a two remote PBXs. I use the switch = statement on each PBX to
point to the other PBX.
Now I want extensions on PBX-1 to dial extensions and PSTN numbers that
are local to PBX-2.
However, I ALSO want people to be able to dial into a
I've tried all versions (stable CVS), even the latest new generation one (NIXJ) but
no luck dialing out. All I can do is receive calls from the PSTN with it, but not
making calls.
Can you dial out with a linejack? Can you tell me how?
Cheers,
-Z
- Original Message -
From: Bruce
If you have to set up different users for the different contexts what'
the usefulness of having a /context on the switch = statement?
On Wed, 2003-07-02 at 10:47, Steven Critchfield wrote:
On Wed, 2003-07-02 at 10:36, Eric Wieling wrote:
I have a two remote PBXs. I use the switch = statement
The mysql schema is available in the doc/cdr_mysql.txt file (from the
asterisk source dir)
James
On Thu, 3 Jul 2003, Kim C. Callis wrote:
You can find the comma delimited file at /var/log/asterisk/cdr-csv or if
you are looking to do some easy querying on a database, you need to
create a
No, u CANT dial out with a linejack...
I have 2 of them, I use them as incoming only lines, and x100p for
incoming, outgoing lines...
There is no bug about this, is a feature, that isn't present at the
linejack.
On Wed, 2 Jul 2003, Zara Trousk wrote:
I've tried all versions (stable
Sorry, that last sentence I had put in there was a thought interupted by
my normal job. I meant for you to do the little research into adding the
context to the end of your switch statement, and if I was wrong then go
about the adding of users.
On Wed, 2003-07-02 at 11:02, Eric Wieling wrote:
or the agent sets a 'disposition' for that call, before it will
exit wrapup and place back in queue.
Jim Friedeck wrote:
Wrap-up, as our existing phone system calls it, is a period of time that
an agent will not get an incoming call after hanging up the previous
call. This allows time for
I've been reading the Linejack strikes again messages, and have another Newbie question
is it possible to use a Voip Product as a Dialout line for * ?
I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin
box?
The Vega100 does either sip or h.323.
Thanks.
Bradley
Are the 2 SIP UA's configured for the same codec?
Michael Kane
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
www.to-talk.com
508-295-2826
- Original Message -
From: Kevin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003 11:05 AM
Subject:
Yes, but I have been able to mitigate it by setting the following
parameters. I have the problem with ATA's that are behind firewalls
and not, but mostly with the ones that are behind firewalls.
CfgInterval:1800
SIPRegInterval:100
On Thu, 3 Jul 2003, Kim C. Callis wrote:
Is it just me or do
Hi All...
I have a maddening problem...
I have Asterisk configured to pick up a line after 4 rings. I do this to
allow my fax machine to pick up a particular distinctive ring pattern, so I
don't have to pay for a dedicated fax line.
If someone calls the line, lets it ring 3 times and then
On Wed, 2003-07-02 at 13:34, Jim Archer wrote:
Hi All...
I have a maddening problem...
I have Asterisk configured to pick up a line after 4 rings. I do this to
allow my fax machine to pick up a particular distinctive ring pattern, so I
don't have to pay for a dedicated fax line.
If
Sjphone is set for Remote preferences for Codec Preference Selection
Do you I want it Local preferences?
Kevin,
-Original Message-
From: Michael Kane [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003 12:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Sip call dropping
On Wednesday 02 July 2003 01:34 pm, Jim Archer wrote:
Hi All...
I have a maddening problem...
I have Asterisk configured to pick up a line after 4 rings. I do
this to allow my fax machine to pick up a particular distinctive
ring pattern, so I don't have to pay for a dedicated fax line.
Same trouble J
Regards
Humberto
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kim C. Callis
Sent: Jueves, 03 de Julio de 2003 10:34 a.m.
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] ATA-186
de-register
Is
it just me or do others
How do you tell asterisk to detect for fax tones?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, July 02, 2003 2:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On 2003-07-02 at 13:54, Steven Critchfield ([EMAIL PROTECTED]) wrote:
Get a TDM10B, cancel your distinctive ring, and let asterisk answer
immediately and detect fax tones
Unfortunately, not all fax machines send the CNG tone, so using a
separate fax number with distinctive ring is far more
When the line is picked up, and a prompt is being played for the caller,
asterisk listens for the beep, pause, beep, noise that an originating
fax makes. When asterisk detects this it jumps to a fax extension in the
current context to complete the call. This fax extension can be a dial
string to
You must put and fax exten in your context:
For example:
[default]
...
exten = fax,1,Dial(Zap/2|30|d)
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joe
Antkowiak
Enviado el: miércoles, 02 de julio de 2003 21:13
Para: [EMAIL PROTECTED]
Thanks, but this is not a great solution. It still leaves the line off
hook for the length of the timeout and limits real calls to the timeout as
well.
--On Wednesday, July 02, 2003 2:11 PM -0500 Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 02 July 2003 01:34 pm, Jim Archer wrote:
You Answer on analog channels and then you need to have
a fax extension in the current context.
regards
Martin
On Wed, 2 Jul 2003, Joe Antkowiak wrote:
How do you tell asterisk to detect for fax tones?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi Jim,
You're probably not receiving disconnect-supervision on your analog
lines, or have Zaptel configured incorrectly to recognize it. Check
the list-archives (available from www.asterisk.org). You could try the
busydetect-statement in zapata.conf. Also check Asterisk's main Makefile
for
On Wed, 2003-07-02 at 13:12, Joe Antkowiak wrote:
How do you tell asterisk to detect for fax tones?
create and exten for fax
exten = fax,1,Dial(${MYFAXDEVICE})
--Karl
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent:
On Wednesday 02 July 2003 02:42 pm, Jim Archer wrote:
Thanks, but this is not a great solution. It still leaves the line
off hook for the length of the timeout and limits real calls to the
timeout as well.
Not so. Once the user presses any DTMF, the first thing you can
do is to set
At 22:10 2003-07-01 -0400, you wrote:
To find out what version yuor using, dial *999 and a debug/trace window will
appear. In the SIP messages it will indicate the type of UA your using and
the version. example below: try another call attempt with this window
open and capture the call flow and
Ah! I see, thanks!
And thanks to everyone else. I have plenty to ways to proceed now. I'm
going to try the cheap solutions first.
Jim
--On Wednesday, July 02, 2003 3:34 PM -0500 Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 02 July 2003 02:42 pm, Jim Archer wrote:
Thanks, but this
Hey Jim , you are correct in respect to the Service provider must pay for
the bandwidth as I/we will be hair pinning calls back into the Internet. As
far as voice quality is concerned (which is my biggest concern) the
solution(box) FWD uses will not be the solution I will implement. I am
Erik over at XTEN is pretty good at getting the latest build out for both
lite and prolet me email my partner they(xten just emailed us the latest
and greatest for both. I also thought as of a week or so ago 1035 was the
last build they would publish until their new product offering. (which
your right.1016 is the latest and greatest...
Michael Kane (President/CEO)
To-Talk Communications LLC.
37 Sandusky Dr.
Wareham, Ma. 02571
www.to-talk.com
508-295-2826
- Original Message -
From: Moshe Yudkowsky [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003
Ok, how do I detect the pressing of any touch tone so I can set the timeout
back to 0?
--On Wednesday, July 02, 2003 2:51 PM -0500 Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 02 July 2003 02:12 pm, Joe Antkowiak wrote:
How do you tell asterisk to detect for fax tones?
Zaptel devices
Actually there is a WaitForRing app that would probably solve this more
easily.
Mark
On 2 Jul 2003, Steven Critchfield wrote:
On Wed, 2003-07-02 at 13:34, Jim Archer wrote:
Hi All...
I have a maddening problem...
I have Asterisk configured to pick up a line after 4 rings. I do this
basically make priority 1 of any extension other than timeout be a set
timeout 0.
On Wed, 2003-07-02 at 16:32, Jim Archer wrote:
Ok, how do I detect the pressing of any touch tone so I can set the timeout
back to 0?
--On Wednesday, July 02, 2003 2:51 PM -0500 Tilghman Lesher
[EMAIL
On Wednesday 02 July 2003 04:32 pm, Jim Archer wrote:
Ok, how do I detect the pressing of any touch tone so I can set the
timeout back to 0?
Pressing DTMF while waiting for a timeout (or while playing a file
with Background) will redirect you to a new extension.
For example, if you call us
Mark, I took a look at this app. How would I do this? Put wait for ring
at the top of the loop? How do I detect and act on the return value?
--On Wednesday, July 02, 2003 4:38 PM -0500 Mark Spencer
[EMAIL PROTECTED] wrote:
Actually there is a WaitForRing app that would probably solve this
Ok, thanks. I was hoping I would not have to set timeout back to 0 for
each extension...
--On Wednesday, July 02, 2003 4:54 PM -0500 Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Wednesday 02 July 2003 04:32 pm, Jim Archer wrote:
Ok, how do I detect the pressing of any touch tone so I can set
Hello All!
There is description of my problem with Asteriks below.
Asteriks CLI says:
File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded on call
Sip debug on the server gives the next:
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.26:5060
From: sip:[EMAIL
Yo all,
As there has been some intrest, here's my updated version:
I post it to -dev as well as -users, as it may be of intrest to
both.
Inspired by the example in the tips tricks-section of
http://www.junghanns.net/asterisk/;, I built a more elaborate
set of features. Currently, my
Sorry 'bout that vacation message. Procmail usually is smart enough to
avoid sending replies to mailing lists. I put in a rule to prevent this
from happening again.
--
Moshe Yudkowsky * http://www.Disaggregate.com
___
Asterisk-Users mailing list
Yes you can. Configure it either as a SIP gateway or an h.323
gatekeeper.
Bradley Greep wrote:
I've been reading the Linejack strikes again messages, and have another Newbie question
is it possible to use a Voip Product as a Dialout line for * ?
I have a Vegastream 100 Voip to PRI. box.
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