On Mon, 21 Feb 2011 22:12:47 -0600, Shaun Ruffell
sruff...@digium.com wrote:
I don't have much direct experience with the cards supported by the
wctdm driver, but based on what you show here, it appears more like
either a fundamental PCI bus incompatibility, the card isn't seated
properly in
Hi,
I have missed something so I wonder if someone could explain for me?
0424449647 desktop phone
0106024647 DECT phone
0424449630 Helsingborg queue
extensions.conf
---
[support]
exten = 0424449647,hint,SIP/0424449647SIP/0106024647
exten =
Try rrplacing MySQL(Query resultid ${conn_id} SELECT/ ramal/ FROM/
colaboradores/ WHERE/ ramal=${EXTEN});
With MySQL(Query resultid ${conn_id} SELECT `ramal` FROM
`colaboradores` WHERE `ramal`='${EXTEN}');
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi again
Could anybody pls share some thoughts about dialplan in lua? I mean some say
it works faster...I have tested my dialplan with pbx_config
(extensions.conf) , then with ael. Dialplan is not very complex (just some
selects in mysql, then based on select some if, then...etc) I think it is
Hello
Incoming calls from the FXO trigger an AGI script which simply NOOP
data sent by Asterisk through stdin.
The first two NOOP work fine, but after this, Asterisk isn't happy:
= extensions.conf
...
[from_fxo]
exten = s,1,Wait(2)
exten = s,n,Set(CID=${CALLERID(num)})
exten
This is very strange. Everything matches mine except Asterisk itself
(I'm using 1.6.2.16.1).
I did notice that you set the loadzone(s) for UK use - yet your e-mail
address in in Poland. Are you setting this up in the UK?
BTW - you have a typo in chan_dahdi.conf (busydetec=yes is missing the
exten = 3001,n,playback(vm-youhave)
I do have the file in /usr/share/asterisk/sounds:
-rw-r--r-- 1 root root 1452 2008-03-06 00:39 vm-youhave.gsm
but still it does not play it ?!
The goodbye at the end does play correctly.
** vm-goodbye is in /usr/share/asterisk/sounds?
Yes it is.
$ ls
[roaming-ext]
;Create a new roaming extension
exten = 3001,1(readop),Verbose(Create roaming extension)
exten = 3001,n,Read(digito,beep,3)
exten = 3001,n,Playback(you-entered)
exten = 3001,n,SayDigits(${digito})
exten = 3001,n,Verbose(Setting roaming extension 4${digito} to call
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Tuesday, February 22, 2011 5:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension to
H Danny,
Axelle, please post the CLI output from the 3001 call and I'll put up a
dialplan that should work for you.
So this is the output I get:
Connected to Asterisk 1.4.21.2~dfsg-3+lenny1 currently running on
openbts (pid = 20597)
[Feb 22 15:18:02] NOTICE[20626]: chan_sip.c:15642
Hi List,
Someone may have run into this problem. Very strange.
I have a customer running 1.422. They use a digium ISDN card connected to an
primary rate for their inbound currently.
We have tested inbound SIP from one of our trunks. We use these trunks with
all our asterisk customers without an
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Tuesday, February 22, 2011 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension to
Has this issue been fixed in this release of 1.8 (or even in the
previous 1.8.2.3)?
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
Thanks
Ish
On Tue, 2011-02-22 at 08:02 -0500, Asterisk Development Team wrote:
The Asterisk Development Team has announced security releases
On Tue, Feb 22, 2011 at 12:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Has this issue been fixed in this release of 1.8 (or even in the
previous 1.8.2.3)?
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
Thanks
Ish
snip
--
Ishfaq Malik
Software Developer
On 11-02-22 10:16 AM, Ishfaq Malik wrote:
Has this issue been fixed in this release of 1.8 (or even in the
previous 1.8.2.3)?
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
No. The ChangeLog would give you the information you're looking for.
Le 22/02/2011 12:32, Axelle a écrit :
Good idea the Verbose commands, at least I see a bit better what is
happening.
Maybe a core set verbose 3 too ?
I should have thought about that one. Thanks.
But I don't understand the CALLERID part: the roaming user is unknown
on my network, so how
Axelle, please post the CLI output from the 3001 call and I'll put up a
dialplan that should work for you.
snip
Not what I asked for, but here's what I can tell you.
Oh I'm sorry but then what are you asking for? I thought it was the
console messages on Asterisk.
From what you posted,
On Fri, 2011-02-18 at 12:36 +0100, Axelle wrote:
Hi,
I'm trying to automatically have the dialplan assign an extension to a
roaming phone on my network.
I tried the following without success:
exten = 3001,1(readop),BackGround(beep)
exten = 3001,n,Read(digito,vm-youhave,3)
exten =
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Tuesday, February 22, 2011 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Assigning an extension
Hello,
i have asterisk installed and i have configured a client iax and sip without
any issue, when i call a internal extension sip from iax there is no problem
but when i call a iax extension from sip extension the result is KO(wrong
number)
any help please
thanks and Regards
--
On Tue, 22 Feb 2011, salaheddine elharit wrote:
i have asterisk installed and i have configured a client iax and sip
without any issue, when i call a internal extension sip from iax there
is no problem
but when i call a iax extension from sip extension the result is
KO(wrong number)
any
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, February 22, 2011 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] calls between iax
You're not properly reading in the response after each NoOp you send out. Each
time you send something to asterisk in AGI, you must read the response in your
script.
Thanks,
--Warren Selby, dCAP
On Feb 22, 2011, at 4:39 AM, Gilles codecompl...@free.fr wrote:
Hello
Incoming calls from
I have a situation where an Asterisk server is NATted, sitting behind a PIX.
One public IP is used for one purpose, now a second public IP is required for
another.
Is there a way to have Asterisk use more than one public IP when behind NAT?
(I already use the externalIP setting)...
If not,
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Tuesday, February 22, 2011 3:34 PM
To: Asterisk Users List
Subject: [asterisk-users] Multiple public address to one Asterisk
serverbehind NAT?
I have a
You could run two copies of asterisk on different private IP addresses.
Have your current install bound to the first private IP with the
externalIP set to the first public and the second install running on
the other IP with the other externalIP set.
On Tue, Feb 22, 2011 at 2:34 PM, Michelle
Finally I could get it to work by running a shell script which parsed
results from 'queue show' CLI command in dearch of 'Not in Use' members. It
was done with an AGI.
Regards,
Daniel
On Tue, Feb 8, 2011 at 11:52 AM, Carlos Chavez cur...@telecomabmex.comwrote:
On Mon, 2011-02-07 at 10:44
There is only one NIC internally (only 1 internal IP) so binding to 0.0.0.0
won't do anything. Asterisk uses the externIP setting to publish a different
address when behind NAT, that's what externIP does. But there is only one
externIP settings.
I'm thinking about openSER/proxy/etc type
Yeah, this is messages which i saw before. Weird is that its hidden
somewhere under registration form and there was no notification about
cancellation for registered users.
Anyway, its a pity that AstriEurope is cancelled.
Are there other similar conference in Europe in 2011 ?
Regards,
Albert
Hi Andrew,
thanks for your answer. I haven't notice this typo before, i was
replacing this config so many times ;-)
I did as you suggested, replaced with your config but result is still
the same.
Some technicians from telco came yesterday to investigate and confirmed
that something is wrong
On Tue, 22 Feb 2011 14:54:38 -0600, Warren Selby
wcse...@selbytech.com wrote:
You're not properly reading in the response after each NoOp you send out. Each
time you send something to asterisk in AGI, you must read the response in your
script.
Thanks for the tip. It's working now.
--
On Tue, 22 Feb 2011 14:54:38 -0600, Warren Selby
You're not properly reading in the response after each NoOp you send
out. Each time you send something to asterisk in AGI, you must read the
response in your script.
On Wed, 23 Feb 2011, Gilles wrote:
Thanks for the tip. It's working now.
How/Where would I do that?
TIA
CF
On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote:
On 2/21/11 4:46 PM, C F wrote:
I just installed an FXS module onto a 4 channel tdm thats about 5
years old and it wont work. Running dmesg I can see the following
error:
Zapata
On Tue, Feb 22, 2011 at 11:00:22PM -0500, C F wrote:
On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote:
On 2/21/11 4:46 PM, C F wrote:
I just installed an FXS module onto a 4 channel tdm thats about 5
years old and it wont work. Running dmesg I can see the following
Figured it out...
1) Incoming SIP immediately routed out a Dahdi PRI trunk is answered
just before it dials the trunk
2) CNG detected after call is bridged
3) Call redirected to fax extension AFTER the bridge is torn down and
the hangup extension is run in the original Dial context
4) Fax
On Tue, Feb 22, 2011 at 11:49 PM, Albert alber...@wp.pl wrote:
Yeah, this is messages which i saw before. Weird is that its hidden
somewhere under registration form and there was no notification about
cancellation for registered users.
Yes, it's in a popup when you try to register. I imagine
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