I have * 1.6.0.9 with dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.2, libpri
1.4.10 and wanpipe-3.4.1 running on CentOS 5.3 64bit.
I have 2 ports of the a104d configured for use with PRI lines and 2 ports
configured for use with Adtran Total Access 850 channel banks. The channel
banks have 6 four port
On Thu, May 07, 2009 at 07:24:21AM +0200, Massimo Nuvoli wrote:
John Novack ha scritto:
Not sure how you would do that, as the X100 card is an FXO card,
won't provide either battery or dial tone to the cordless. What you
will want for that is an FXS card or ATA. The X100 card will
Hi All,
My theory on the codec translation deepens:
Doing a core show translation on the A1 server (working) I get:
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc
g726 g722 amr
g723- ---- -- -- --
-- -
Starting from today i am receiving the following errors on asterisk..
What can be the reason for it?
[May 7 11:45:16] ERROR[14885]: chan_dahdi.c:10515 dahdi_pri_error: ACK
received for '0' outside of window of '3' to '4', restarting
[May 7 11:45:16] WARNING[14885]: chan_dahdi.c:3347
On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Some X100P cards (e.g.: those that are based on SI3034, but not those
basedon SI3035) support programmable impedance settings. Sadly the
wcfxo driver does not support it.
Fixing it should mostly be a matter of
I am sending SIP or H323 calls to a carrier, and I need to store in the CDR
why the calls are rejected or why they hang up. In SIP, it can be code 503,
500, 488, etc. How do I get the information in my dialplan? I don't mean
$(DIALSTATUS}, but the real numeric code
F.Alves
2009/5/7 Jim Dickenson dicken...@cfmc.com
I have * 1.6.0.9 with dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.2, libpri
1.4.10 and wanpipe-3.4.1 running on CentOS 5.3 64bit.
I have 2 ports of the a104d configured for use with PRI lines and 2 ports
configured for use with Adtran Total Access 850
Hi
OK I have upgraded to 1.6.0.9 and it looks like the issue disappeared,
but I am facing another problem yesterday my asterisk gave me this
into the logs:
[May 6 17:25:53] ERROR[20625] channel.c: ast_read() called with no
recorded file descriptor.
Message was repeated x times and cause my
On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote:
On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Some X100P cards (e.g.: those that are based on SI3034, but not those
basedon SI3035) support programmable impedance settings. Sadly the
wcfxo driver does
On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote:
X100P SE Setup Guide - Global Line Standards
http://novavox.co.uk/support/x100p.html
Richard Spencer supp...@novavox.co.uk
Another thing: their global-line-standard should basically (if
properly written) resolve
Hmm would this somehow get to the the D channel and stop incoming
calls from coming in the PRI? My box does not show the D channel in the
zap channel list so I suspect I can't destroy that.
In our situation if the PRI's D is down the calls would go to the next
PRI.
-James
On Wed, May
Justin Phelps wrote:
I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650.
I attempted to simply reuse the existing config files for the old phone
on the new phone, but the new phone would lock up on the 4th digit when
attempted to dial out certain numbers. So, I downloaded the
From: Grygoriy Dobrovolskyy megaho...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thu, 7 May 2009 12:20:07 +0200
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re:
Ricardo Melendez wrote:
Hi to All, I need to implement an automatic telephone messaging system
that works like this:
-the system generates the call based on mysql records or any database
-when the client answer the phone, the Asterisk PBX playback a
recorded message
-when finish,
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of James Van Vleet
Hmm would this somehow get to the the D channel and stop incoming
calls from coming in the PRI? My box does not show the D channel in
2009/5/7 Jim Dickenson dicken...@cfmc.com
*From: *Grygoriy Dobrovolskyy megaho...@gmail.com
*Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
*Date: *Thu, 7 May 2009 12:20:07 +0200
*To: *Asterisk Users Mailing List - Non-Commercial
Hello,
I am getting the following error on my CLI
[May 6 15:59:20] ERROR[25789]: cdr_csv.c:314 csv_log: Unable to re-open master
file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
I am a bit of a Linux newb so please be gentle. I assume this has something to
do with the
Just write an AGI and run it from the h context of your voicemail
Exten = 7000,1,Voicemailmain(1...@default)
Exten = 7000,h,AGI(cleanmail.agi)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian
Alexander
Sent:
On 7 May 2009, at 14:29, Brent Vrieze wrote:
Hello,
I am getting the following error on my CLI
[May 6 15:59:20] ERROR[25789]: cdr_csv.c:314 csv_log: Unable to re-
open master file /var/log/asterisk//cdr-csv//Master.csv : No such
file or directory
I am a bit of a Linux newb so please
From: Grygoriy Dobrovolskyy megaho...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Thu, 7 May 2009 15:21:03 +0200
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re:
Here is your problem. The directory /var/log/asterisk/cdr-csv must exist
for asterisk to write it's plain-jane (their term) text CDR file. This is
defined in cdr.conf (it's the last working section of mine). You can create
the directory or comment out that section of cdr.conf. Your choice.
I use an application that fits exactly your needs but is not free.
You can ...
- load a call file containing the numbers to call.
- load the call file from a db.
- design you own context using a simple editor that generates an AEL
compatible dialplan and manages the existing contexts saved in
Can any one suggest a little code to either ring a cell phone when a new VM
message is recorded, or send a text message?
Basically outside sales people want to know they have a new message, but
don't want to be interrupted to take a forwarded call.
While a message by message notice would be
On Wed, May 6, 2009 at 10:53 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Not sure how you would do that, as the X100 card is an FXO card, won't
provide either battery or dial tone to the cordless.
What you will want for that is an FXS card or ATA.
The X100 card will connect to a
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Cary Fitch
Can any one suggest a little code to either ring a cell phone when a
new VM
message is recorded, or send a text message?
Basically outside
On 7 May 2009, at 15:04, Cary Fitch wrote:
Can any one suggest a little code to either ring a cell phone when a
new VM
message is recorded, or send a text message?
Basically outside sales people want to know they have a new message,
but
don't want to be interrupted to take a forwarded
Jonathan Moore wrote:
On Wed, May 6, 2009 at 10:53 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Not sure how you would do that, as the X100 card is an FXO card, won't
provide either battery or dial tone to the cordless.
What you will want for that is an FXS card or ATA.
The X100
On Thu, 7 May 2009, Brent Vrieze wrote:
I am getting the following error on my CLI
[May 6 15:59:20] ERROR[25789]: cdr_csv.c:314 csv_log: Unable to re-open
master file /var/log/asterisk//cdr-csv//Master.csv : No such file or
directory
I am a bit of a Linux newb so please be gentle. I
On Thursday 07 May 2009 03:33:14 Adrian Marsh wrote:
So where are the codec translations set?
I assume you're talking about the numbers within the table? They're
calculated at runtime, based upon shortest possible path (in terms of time)
from one codec to another. Most codecs translate only to
On Thu, 7 May 2009, Cary Fitch wrote:
Can any one suggest a little code to either ring a cell phone when a new
VM message is recorded, or send a text message?
Basically, a call from their own number would be the clue that there is
a voicemail waiting.
When I call my cell from my Asterisk
Sorry. I am not sure which patch I wanted you to try now. The issue I
posted may be related to your issue.
On Wed, 2009-05-06 at 14:53 -0400, Carlos Ruiz Diaz wrote:
I think I misunderstood your mail.
There is no patch available yet, right?
I went to the page you linked but I did not found
Thank you.
Please, any patch you think can have possibility to work, just let me know
and I will do the tests and post the results.
Regards
On Thu, May 7, 2009 at 10:33 AM, Matthew Nicholson mnichol...@digium.comwrote:
Sorry. I am not sure which patch I wanted you to try now. The issue I
digitmap
dialplan.digitmap=[2-9]11T|[*]xxT|891xxxT|[1-7]xxxT|8[0-46-8]xxT|8500T|851xxxT|9,1[2-9]xT|9,xxxT
dialplan.digitmap.timeOut=3|3|3|3|3|3|3|3|3/
Do the above changes look in line with common practice JohnM?
--
Justin Phelps
www.onitato.com
850.866.6864
Date: Thu, 07 May
Cary Fitch wrote:
Can any one suggest a little code to either ring a cell phone when a new VM
message is recorded, or send a text message?
Basically outside sales people want to know they have a new message, but
don't want to be interrupted to take a forwarded call.
While a message by
On Thu, 07 May 2009 10:16:55 -0400, Jon Pounder j...@inline.net
wrote:
yeah I agree with the above - I never really found echo to ever be a
problem, my only complaint was on some less than stellar cpu's I was
having dtmf recognition problems.
BTW, can someone explain to a libart major like me
Yes, similarly, if I forward my desk phone to my cell, and then call my desk
phone from the cell, the call goes straight to the cell voice mail retrieve
system, with out even needing a password. (I.E. Cell system thinks it is
the cell phone calling. BIG CELL PHONE SECURITY HOLE! Anyone who can
Thanks to everyone for the help. I suppose questions this easy to
answer can be a nice diversion, at least they are for me. I thought it
might be as easy as adding the directory but the double slashes // in
the CLI error message threw me off.
Anyway adding the directory worked and I am now
The meetmeadmin() dialplan function lets you specify a user to mute,
un-mute or kick. But how do you get a list of users in your dialplan?
When a user joins a conference, the user number assigned is the last user
number +1. If you have a long running conference with callers joining and
leaving
On Thu, 7 May 2009 13:40:20 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Another thing: their global-line-standard should basically (if
properly written) resolve http://bugs.digium.com/view.php?id=11057 .
Though I guess the new code will actually be in DAHDI, as Zaptel is
frozen.
Ah yes,
Justin Phelps wrote:
digitmap
dialplan.digit
map=[2-9]11T|[*]xxT|891xxxT|[1-7]xxxT|8[0-46-8]xxT|8500T|851xxxT|9,1[2-9]xT|9,xxxT
dialplan.digitmap.timeOut=3|3|3|3|3|3|3|3|3/
Do the above changes look in line with common practice JohnM?
Short Answer:
They do.
Longer answer,
BTW, can someone explain to a libart major like me (;-)) where echo
comes on in a telephone conversation? I seem to recall it's due to the
length of the line between the CO and the local party, but I'm not
sure.
I'll try.
Echo occurs when part of the signal traveling in one direction
on the
I have more info now.
I have used AMI to watch events, including DTMF events. The phone comes off
hook and I see a line on the * CLI saying the line is off hook. I then touch
a key and I see the DTMF event in the AMI. At the same time I see a message
on CLI that says hangup the line.
--
Jim
hi list, i have a question about the args of queue:
when we use Queue() app, there are some arguments than can use. help from
CLI:
Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule)
well.. i'm trying to identify who has taken the call on a queue, and,
BTW, can someone explain to a libart major like me (;-)) where echo
comes on in a telephone conversation? I seem to recall it's due to the
length of the line between the CO and the local party, but I'm not
sure.
Yes, I'll tackle that. It takes a finite amount of time for the electrical
signal
Here is what I do:
[macro-cfmc_queue_private]
; Due to a change around version 1.6.0.8 MEMBERINTERFACE and MEMBERNAME are
not set but CHANNEL has the agent we are connected with.
; We need to look for Agent/number is either CHANNEL or MEMBERINTERFACE
and it will be in one or the other
exten =
Is there a way to not have a transcode happen when saving voicemail? ie.
the voicemail gets stored in the same format that the channel calling it
is using - g711, gsm, g729, etc. ?
Just playing with some really weedy processors and wanting to avoid
transcodes at all costs...
Gordon
I've got multiple satellite office all linked back to the main office
via VPN. Each office has their own asterisk server which registers back
to the main office's Asterisk server. Each office also has a 1Mb
downstream / 384k - 768k upstream connection. The branches are using
Speex for their
I do not have examples, but if you are using the 1700 series router in order
to originate the ipsec vpn, you may use command qos pre-classify (please
search for it on cco.cisco.com)
On Thu, May 7, 2009 at 9:54 PM, Brent Davidson
br...@texascountrytitle.comwrote:
I've got multiple satellite
Im getting these messages when making calls from a sip extension/other asterisk
peer out my pri.
[ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/1-1]
[ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/1-1]
It happens whenever i send dtmf. I have all of my devices set to inband, be it
thanks for answer, but is not working...
with your dialplan asterisk returns chan status UNKNOWN and nothing else..
before my mail, i try to use ${MEMBERINTERFACE} and ${MEMBERNAME} into
myMacro , but his values are ever of course, i put yes on
setinterfacevar=yes, on queues.conf file, you
In trying to figure out why I can not connect a phone set to an Adtran Total
Access 850 and get it working I have noticed the following.
On a fresh install and minimal changes to conf file I notice that when I
have a phone connected to my channel bank and I dial *78 asterisk intercepts
this, as
Tobias Wolf wrote:
This be true for AGI, but there is also FastAGI and with it the excellent
asterisk-java package:
http://asterisk-java.org/
It supports writing AGI Scripts in JAVA, which communicates over TCP with
Asterisk. AMI is supported too ...
Last but not least it has a nice
Dear List,
I have an asterisk 1.6.2 installation. I'm trying to configure func_odbc to
read some mysql tables... but every time I tried I got this message:
ERROR[24968] func_odbc.c: Unable to execute query [SELECT bloqueada FROM
funciones WHERE extension='750']
I'm sure the DSN is right because
On 8/05/2009 2:10 a.m., Steve Howes wrote:
On 7 May 2009, at 15:04, Cary Fitch wrote:
Can any one suggest a little code to either ring a cell phone when a
new VM
message is recorded, or send a text message?
Basically outside sales people want to know they have a new message,
but
don't want
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