We're installing from maintained packages (rpm) rather than compiling
code.
On Fri, 2011-02-11 at 17:47 +, satish patel wrote:
Here is the patch did you apply it ?
https://issues.asterisk.org/file_download.php?file_id=28206type=bug
Date: Fri, 11 Feb 2011 08:46:36 -0200
From:
On Mon, Feb 14, 2011 at 04:06:10AM +, Edwin Quijada wrote:
How would be the dialplan for this context from-lan ???
This list is for non-commercial support. If you want someone to do the
work for you, I suggest you go elsewhere and offer money.
--
On Sat, 05 Feb 2011 12:07:28 +0100, Gilles codecompl...@free.fr
wrote:
I've seen articles about Call files. Is this the easiest way to solve
this problem?
I'm reading the 3rd edition of the Asterisk: The Definitive Guide,
but since it's pretty big and there's no guarantee that the answer to
this
I have a problem using asterisk 1.6 with realtime sip.
When I add sip channel (my sip provider) to asterisk using realtime
sip (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip),
incoming calls don't work for me.
In asterisk CLI I get message:
NOTICE[19805]: chan_sip.c:21250
Hi guys,
I am getting YELLOW alarm on my span with E1 line. Anyone knows what
could be the problem ? Below is my config and status...
voice:/etc/dahdi# dahdi_hardware -v
pci::02:08.0 wct4xxp+ d161:1220 Wildcard TE220 (5th Gen)
voice:/etc/dahdi# dahdi_scan
[1]
active=yes
alarms=YEL
Hello,
Can you say us what is connected in which span ?
span 1 =
span 2 =
Yellow would say that the span did not receive external clock
Sebastien
AB2L
+33 (0)367100783
sebast...@ab2l.eu
Le 14/02/2011 11:34, Albert a écrit :
Hi guys,
I am getting YELLOW alarm on my span with E1 line.
Hi,
span 1 - e1 line (15 channels)
span 2 - nothing is connected to 2nd span
below is also status from proc fs
ps. i was also trying with timing provided by card (0)
span=1,1,0,ccs,hdb3
...
but result was the same
voice:/tmp# cat /proc/dahdi/1
Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1
As I know it's only telco who provide the clock. Then you must configure
your span with 1,1,0
1 is as TE (terminal equipment)
0 is as NT (network termination)
So, configure your span as TE, be sure telco not use crc4.
If already yellow, call your telco and ask him if he provide clock or not.
On 14.02.2011 12:17, Sébastien BERGER wrote:
As I know it's only telco who provide the clock. Then you must configure
your span with 1,1,0
1 is as TE (terminal equipment)
0 is as NT (network termination)
So, configure your span as TE, be sure telco not use crc4.
If already yellow, call
This might be a stupid question, but:
If I install a new Linux kernel on a machine running Asterisk, do I have to
recompile DAHDI?
If yes, what do I have to do to get it to build just the kernel modules?
(We use Debian here. Squeeze has just gone stable, and it requires a new
kernel.
Sebastian,
It seems it's working now!
Cable was fine, thought i have added crc4 to span config:
span=1,1,0,ccs,hdb3,crc4
Thank you very much for your help and hints! :)
oice:/etc/dahdi# dahdi_scan
[1]
active=yes
alarms=OK
description=T2XXP (PCI) Card 0 Span 1
name=TE2/0/1
On Mon, Feb 14, 2011 at 9:11 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
This might be a stupid question, but:
If I install a new Linux kernel on a machine running Asterisk, do I have to
recompile DAHDI?
If yes, what do I have to do to get it to build just the kernel modules?
(We
Maybe a DNS problem?
Try to ping from another machine your HOSTNAME.
Best regards,
Fellipe
Date: Mon, 14 Feb 2011 15:29:08 +0500
From: rush...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problems with realtime sip
I have a problem using asterisk 1.6 with
Your welcome !
AB2L
+33 (0)367100783
sebast...@ab2l.eu
Le 14/02/2011 13:08, Albert a écrit :
Sebastian,
It seems it's working now!
Cable was fine, thought i have added crc4 to span config:
span=1,1,0,ccs,hdb3,crc4
Thank you very much for your help and hints! :)
oice:/etc/dahdi#
lolz, I agree. Better to spend more and use it for some time. It is not a
big installation about 4-5 extension so can spare the budget for it easily.
/Khurram
On Sun, Feb 13, 2011 at 5:45 PM, Michael Graves mgra...@mstvp.com wrote:
On Sun, 13 Feb 2011 09:28:19 -0700, Joel Maslak wrote:
My
Thanks Gordon, Grandstream is in my purchase list : )
/ag
On Sun, Feb 13, 2011 at 10:58 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Sat, 12 Feb 2011, ast guy wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking
On Mon, 14 Feb 2011 10:35:52 +0100, Gilles codecompl...@free.fr
wrote:
If someone's already built a callback like the above using an FXO
module, I would appreciate any feedback to try and solve this issue.
I learned more about Local channels, but this doesn't work either:
===
Good Day everyone,
Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by
Cisco, however now the phone does not and will not read the RINGLIST.dat
file. I've tried rebooting the phone, tried resetting the phone back to
factory, have deleted the RINGLIST.dat file and
waiting for replys..
On 02/11/2011 02:20 PM, Nikhil wrote:
Thanks for reply. Any other suggestions .
On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote:
i believe there is a way to do it using asterisk and flashphoner
++
2010/12/20 Gilles codecompl...@free.fr
Well I think you need major changes as application in android run in sandbox
instead of direct Linux APIs. Till now no news on it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Monday, February 14, 2011 6:46 PM
To:
Better to report a BUG to cisco.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller
Sent: Monday, February 14, 2011 6:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cisco 7960 asterisk 1.8.22
That's the problem, I am not sure if the problem lies with Cisco, or if it
lies with Asterisk. I figured I'd try here first before running in circles
with a TAC Case.
Regards.
I see blindness, not as a disability, but more of an ability. And Sight
actually, more of a disability because
On Mon, 14 Feb 2011 14:21:50 +0100, Gilles codecompl...@free.fr
wrote:
Could it be that while we're in the dialplan after getting a call from
the FXO, the FXO is just not available until after we exit the
dialplan?
Made some progress: Asterisk can dial my cellphone if the callback
goes through an
Hello all
I have a small issue with some mobiles numbers when I call these numbers
using asterisk I have all the time answer machine. But when I call these
numbers using my mobile or another phone there is no problem.
Any help will be appreciated
--
Tilghman,
Might not be your question to answer, but if I did get a BMI
license, this would allow me to use virtually any music I wanted for MOH?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, February 14, 2011 8:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] issue with some numbers
Hello all
I have a small
Thanks for the insight.
Although your logic works absolutely fine, I couldn't get it to work the way
I wanted so I simplified it.
I could post the script if anyone is interested.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
thanks for your response
i have tested with a regular phone and i get the same result
my question if there is any action to do in dial plan or extenssion.conf in
order to call this number becouse in dial plan i can
bloc a number to be call
exten = _OUT.,n,Set(match=${REGEX(^06XXX
DNS work fine on this machine:
; DiG 9.7.2-P3 -t SRV _sip._udp.sipnet.ru
;; global options: +cmd
;; Got answer:
;; -HEADER- opcode: QUERY, status: NOERROR, id: 53284
;; flags: qr rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 0, ADDITIONAL: 0
;; QUESTION SECTION:
;_sip._udp.sipnet.ru. IN
The simplest (IMO) option for this would be ex-girlfriend blocking. The
CON to this is that it would require an entry for each number to block.
Say all of your calls to 555-1234 go to the answering machine. In the EGB
scenario, you would put this line in your dialplan
Exten = _5551234,1,hangup
Le 14/02/2011 15:44, salaheddine elharit a écrit :
thanks for your response
i have tested with a regular phone and i get the same result
my question if there is any action to do in dial plan or
extenssion.conf in order to call this number becouse in dial plan i can
bloc a number to be call
You may need to provide some more scenario detail
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, February 14, 2011 7:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] issue with
OK thanks for your help,
i want to know if there is anythings to put in dial plan in order to disable
the answer maching for these numbers
thanks and Regards
2011/2/14 Danny Nicholas da...@debsinc.com
The simplest (IMO) option for this would be “ex-girlfriend” blocking.
The “CON” to this is
On Sun, Feb 13, 2011 at 11:07:53PM -0600, Tilghman Lesher wrote:
On Friday 11 February 2011 16:37:49 Danny Nicholas wrote:
Hi gang,
In 500 words or less (if possible), please explain what is a
legal music-on-hold file? My boss hates the stuff provided with the
distribution
Hi,
I was trying to use SIP session timers with Asterisk 1.8 and two Snom
phones: is it possible to force SIP timers usage only on some specific
SIP channels or is it somehow a global setting?
Thank you very much,
Guido.
--
On Mon, Feb 14, 2011 at 5:40 AM, James Miller paramedi...@gmail.com wrote:
Good Day everyone,
Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by
Cisco, however now the phone does not and will not read the RINGLIST.dat
file. I’ve tried rebooting the phone, tried
I lost you - you say that you get the same result dialing from a land-line
as from Asterisk. If the call from LL is also going to answering machine,
you are done.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
On Sun, Feb 13, 2011 at 9:25 PM, Roi Stork roi.st...@gmail.com wrote:
Here's the messages log. There's a line that says ERROR: Unsupported DS E1
CHIP (00:00)
That's pretty bad. Could you post the output of wanrouter hwprobe verbose
?
Moises Silva
Senior Software Engineer
Sangoma Technologies
I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the ring
type setting:
Chirp 1
Chirp 2
24 24-ring-tone-1.raw
Att1 ring_att1.pcm
.
.
.
However, when you attempt to play one it says Loading Ringer
The Queue() application can automatically pause members who fail to
answer; this would be the solution to your problem. With that solution
in place, though, the agent will still need to be able to un-pause
when they return to their desk, and since that is the case, they
really should be
On Mon, Feb 14, 2011 at 9:04 AM, James Miller paramedi...@gmail.com wrote:
I did the command listed, and its actually requesting RINGLIST.DAT, so I
changed the filename to match its request but now its showing in the ring
type setting:
Chirp 1
Chirp 2
24 24-ring-tone-1.raw
Att1
I did that and this is what I got when I tried to play the 24 ringtone:
13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency
ring_emergency.pcm octet
In the ringlist.dat file in the first column I typed the display name then
hit the tab key. Now on some it only moved a
Hi,
I manage an SBC which stands between my company server farm and some SIP
telco trunks. The system works fine, for inbound and outbound calls.
Now I've configured the SBC to also act as a registration proxy, forwarding
SIP registrations coming from the Internet to my asterisk servers.
It all
On Monday 14 February 2011 08:23:08 Danny Nicholas wrote:
Might not be your question to answer, but if I did get a BMI
license, this would allow me to use virtually any music I wanted for
MOH?
The answer is, as long as the music publisher for each piece of music has
an agreement with
And there is also ASCAP: American Society of Composers, Authors and
Publishers.
Also a smaller than either group: SESAC.
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday,
Probably I will go with cisco 7945g I hope its support is good with
asterisk. Have you used it ? is it simple in configuration?
/Khurram
On Sat, Feb 12, 2011 at 1:47 PM, Andrew Latham lath...@gmail.com wrote:
On Sat, Feb 12, 2011 at 9:31 AM, ast guy ast...@gmail.com wrote:
Hi,
I have
Thanks??? The easiest legal option seems to be to purchase individually
licensed on-hold tracks for $10-99 US per pop. Ouch! Or, you can spend
$750 or so US per year for the 3 musketeers license.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Currently most every phone works well, if the patch for Cisco
subscriptions gets tested then I would say any non-skype phone.
Andrew, do you have any details on this patch? Is it in the bug tracker? I'd
be happy to try it out.
I've taken a patch from
On Mon, Feb 14, 2011 at 6:20 PM, Kai-Uwe Jensen kujen...@gmail.com wrote:
Currently most every phone works well, if the patch for Cisco
subscriptions gets tested then I would say any non-skype phone.
Andrew, do you have any details on this patch? Is it in the bug tracker? I'd
be happy to try
Now in my asterisk config files, there are lines like:
secret=some_password_in_plain_text
Is it possible to hide these plain text password?
--
*Jian *
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On 2/14/2011 4:36 PM, Jian Gao wrote:
Now in my asterisk config files, there are lines like:
secret=some_password_in_plain_text
Is it possible to hide these plain text password?
I think 'md5secret' is what you're looking for.
http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret
--
On 02/14/2011 03:36 PM, Jian Gao wrote:
Now in my asterisk config files, there are lines like:
secret=some_password_in_plain_text
Is it possible to hide these plain text password?
Who are you hiding them from? Anyone with access to the Asterisk server
can already do far more damage than
On Mon, Feb 14, 2011 at 6:46 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 02/14/2011 03:36 PM, Jian Gao wrote:
Now in my asterisk config files, there are lines like:
secret=some_password_in_plain_text
Is it possible to hide these plain text password?
Who are you hiding them from?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeremy Kister
Sent: Monday, February 14, 2011 3:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hide the plain text
I am building a server for a client. I want them to try out the new
Google Voice feature using my GV account. But I don't want expose my
GV's password.
*Jian *
On 11-02-14 01:46 PM, Kevin P. Fleming wrote:
On 02/14/2011 03:36 PM, Jian Gao wrote:
Now in my asterisk config files, there are
On 02/14/2011 04:08 PM, Jian Gao wrote:
I am building a server for a client. I want them to try out the new
Google Voice feature using my GV account. But I don't want expose my
GV's password.
There is no method to obscure a Google Voice password in the config
file. chan_sip supports obscured
Who are you hiding them from? Anyone with access to the Asterisk server
can already do far more damage than extracting these passwords.
You may (like we do) want to store config files in a version control system
in a common repository. People who have access to that repository don't
necessary
On 11-02-14 05:10 PM, Kevin P. Fleming wrote:
On 02/14/2011 04:08 PM, Jian Gao wrote:
I am building a server for a client. I want them to try out the new
Google Voice feature using my GV account. But I don't want expose my
GV's password.
There is no method to obscure a Google Voice password
On 11-02-14 05:08 PM, Jian Gao wrote:
I am building a server for a client. I want them to try out the new Google Voice
feature using my GV account. But I don't want expose my GV's password.
Actually in this case, your best bet is just going to be to create a separate
account where you don't
Hi,
How can I configure my asterisk server so that I can receive incomming calls
comming from the same IP from where my server also receives phone
registrations?
The problem is that since the moment any extension registers at that IP
(actually I have a registration proxy running at that IP),
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
Bizarre bug?
root@minipbx:~# asterisk -V
Asterisk 1.4.37
root@minipbx:~# uname -a
Linux minipbx 2.6.32-dockstar #2
Sounds like a clock slip/ntp issue
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Monday, February 14, 2011 10:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
On Mon, Feb 14, 2011 at 3:10 PM, Danny Nicholas da...@debsinc.com wrote:
Thanks??? The easiest legal option seems to be to purchase individually
licensed on-hold tracks for $10-99 US per pop. Ouch! Or, you can spend
$750 or so US per year for the 3 musketeers license.
Other options include
Hi all,
I'm trying to get outgoing fax working from an SPA2102 to a PSTN fax machine
via a T.38 enabled trunk. I've got
t38pt_udptl = yes
faxdetect=no
in my sip.conf file. The ATA has all of the T.38 options turned on, echo
cancellation is off, as well as silence suppression off. The only
The syntax that I use is:
SetVar: username=justincase
I'm using Asterisk 1.6.2. Hope that helps.
Mike.
Bruce B bruceb...@gmail.com wrote:
Hi Everyone,
I am trying to pass a variable using the .call files but it turns out
blank.
Can someone please point out what might be wrong here:
On Tuesday 15 Feb 2011, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
Bizarre bug?
I'm guessting, this is a brand new machine on its
Thanks for the input. Lack of proper documentation really causes issues with
things like this. I think it's noted somewhere on Voipinfo that SetVar was
to be used with version 1.0 and prior to 1.6. It turned out that the issue
was with s,1,Answer() for first leg of the call. Taking that out fixed
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