2009/1/8 Godson Gera godso...@gmail.com
On Wed, Jan 7, 2009 at 10:49 PM, Olivier oza-4...@myamail.com wrote:
Hi,
I'm bumping on this :
cd /usr/src
wget http://www.misdn.org/downloads/mISDN.tar.gz
tar xvf mISDN.tar.gz
cd mISDN-1_1_18
make
snip
In file included from
On 12:55, Thu 08 Jan 09, Mikel Lindsaar wrote:
Thanks Mark,
The phone starts, I can get to settings and such... just it keeps
looking for this file.
I just opened a TAC ticket and getting it handled.
As far as I know it only looks for this file when the phone boots and
does a couple of
Hi!
I am in the process of upgrading our 1.2 servers to 1.6.
We have a lot of realtime extensions with app=Macro and
appdata=stdexten|080512|SIP/080512
But this does not work in 1.6. It is expecting , and not | as the
argument seperator. If I change the | to , then it does not work in
1.2.
Is
Hi.
When I call my Digium B410p, connected to France Telecom Dual T0 RNIS ,
I get :
Jan 8 10:53:56 obelix kernel: [ 369.769601] Setting B-channel 4 to
echo cancelable state on PCM slot 9
Jan 8 10:53:56 obelix kernel: [ 369.769657] Enabling pass through for
channel
Jan 8 10:53:56 obelix
Am Mittwoch, den 07.01.2009, 16:59 -0500 schrieb TianLun Song:
From the product description, i dont think Gizmo5 allows me to
register the client with my asterisk. If i am wrong, please let me
know
AFAIK, it does, I tried it once and I kinda remember it has an
additional SIP account config
Hi!
Currently I provision user account in users.conf. But I do not like that
VoiceMail writes to users.conf when the voicemail password is changed.
Is there a possibility to store the vmsecret in another place? (another
file or DB)?
thanks
klaus
If I understand this, you can do it with the pbx_realtime=1.4 section in
the [compat] header of asterisk.conf.
I found this information in the UPGRADE-1.6.txt file... funny enough.
Fifth paragraph:
* The delimiter passed to applications has been changed to the comma
(','), as that is what
Hello,
Thanks for your reply!
I want to confirm that any other things can cause this freeze issue or not,
and how can we prevent this such case.
If asterisk got freeze regarding the down connection time with dns server,
but when it is able to access then asterisk will resolve this freeze issue
Hi!
All the AEL examples have a semicolon after the closing curly bracket, e.g:
context test {
1 = Hangup();
};
but without ; it works fine too, e.g:
context test {
1 = Hangup();
}
So - what is the reason for the ; after the closing curly bracket?
thanks
klaus
Noah Miller wrote:
I don't believe that Polycom's version of SLA does anything with
Asterisk. You have to use asterisk's SLA implementation
(http://www.asterisk.org/node/48342).
- Noah
So asterisk can't do SLA with Polycom phones?
mark
___
2009/1/8 Klaus Darilion klaus.mailingli...@pernau.at
Hi!
All the AEL examples have a semicolon after the closing curly bracket, e.g:
context test {
1 = Hangup();
};
but without ; it works fine too, e.g:
context test {
1 = Hangup();
}
So - what is the reason for the ; after
Thanks.
Note to self. Read UPGRADE document better next time. :-)
/Morten
On Thu, Jan 8, 2009 at 1:55 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
If I understand this, you can do it with the pbx_realtime=1.4 section in
the [compat] header of asterisk.conf.
I found this information
2009/1/7 Julien Claassen jul...@c-lab.de
Hi!
I'm part of two mailinglists. I haven't heard much from one, but in the
other there are some mails now and then. isdn4linux and then there's a
specific misdn mailinglist (I think in connection with asterisk). I'll
check
my address book if
Hi
I have an asterisk 1.6 running, and our provider have an openser on their end.
When I get an incoming call I get this on my end
[Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call
from '' to extension '0840303395' rejected because extension not found.
If I wait
Hi!
I wonder how to configure a SIP peer which
- requires authentication for calls to sent to the peer
- needs to authenticate for incoming calls
I want to have different username/password for incoming and outgoing
direction.
Thanks
Klaus
___
--
hi
can you post your extension.conf?
thanks
David
2009/1/8 Ralf Träskman r...@adlibris.com
Hi
I have an asterisk 1.6 running, and our provider have an openser on their
end.
When I get an incoming call I get this on my end
[Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869
I don't believe that Polycom's version of SLA does anything with
Asterisk. You have to use asterisk's SLA implementation
(http://www.asterisk.org/node/48342).
So asterisk can't do SLA with Polycom phones?
Asterisk can do SLA with Polycom, just not using Polycom's SLA
implementation (in
Here it is, the part I want to use is the things under [ip-only]
/ralf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire
Sent: den 8 januari 2009 15:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Try using auth=
From sip.conf.sample:
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can
I also need SLA for hundreds of phones. Looking at Asterisk's way of doing
it I got cold legs... I am now playing with OpenSIPS; didn't manage to make
it working there, but it should be much simpler.
And no, I am not telling you to drop Asterisk - it is good for other things,
and even some things
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Klaus Darilion
Sent: Thursday, January 08, 2009 8:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AEL and
Any context you have specified in sip.conf? There the extension is searched
for. And if that's not default, it might not find it.
br
Walter
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ralf
Hello!
* Version: 1.6.0.3-rc1
Scenario: * - Proxy - routed back to myself (The only thing changing
is the Request URI)
(And the Record-Route, Via that are added, of course).
Outgoing Context is faxserver-out, incoming context is faxserver (at
least should be).
Outgoing context is straight
Hi
This is my sip.conf
Ralf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mailinglists
Sent: den 8 januari 2009 16:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem incomming from
hi
try this... add the s extencion to ip-only whit just one line verbose (S ext
called)
the s extencion is like default extencion
just to see what happen...
maybe the operator isnt sending all the info to you.
David
2009/1/8 Ralf Träskman r...@adlibris.com
Hi
This is my sip.conf
Hello,
I'm running three asterisk boxes, spread across three different countries. One
of the offices is running Asterisk 1.2.18 on the Druid Telephony Platform(not
my choice, has been in before I started and haven't had the time to remove it).
My situation I have is based on the contexts
Hi
I dont understand how to do that, I put in this line to ip-only in
extention.conf
exten=s,1,Dial
No differens
Thanks for all help
/r
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire
Sent: den 8 januari 2009 16:46
To:
I am using the console/dsp and ALSA on a motherboard with HDMI.
If I am using analog sound on the motherboard it works.
If I switch to the digital sound everything else works but asterisk...
No sound I mean.
/etc/asound.conf for analog is:
pcm.snd_card {
type hw
card 0
device 0
}
On Wed, Jan 7, 2009 at 6:01 PM, Steve Murphy m...@digium.com wrote:
On Wed, 2009-01-07 at 02:56 +, Grey Man wrote:
On Tue, Jan 6, 2009 at 3:53 PM, Steve Murphy m...@digium.com wrote:
That sounds a bit dangerous to me. If you go down the path of setting
the answer time based on dial plan
On Thursday 08 January 2009 09:56:04 Darrin Henshaw wrote:
My situation I have is based on the contexts already in place, particularly
for outbound calls, I need to do a Goto sending the call back into the
extension it currently is. The reason for this is they want to implement
call recording
Jeez, I feel like a tool right now. I totally missed the fact that I can send
it to a priority by itself. Thanks Tilghman.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Thursday,
ahhh.
Thus I guess the outgoing credentials will be supplied in the
[authentication] section and in incoming credentials will be defined in
the peer section.
thanks
klaus
Leif Madsen schrieb:
Try using auth=
From sip.conf.sample:
[authentication]
; Global credentials for outbound
Watkins, Bradley schrieb:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Klaus Darilion
Sent: Thursday, January 08, 2009 8:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
A couple of our customers are having issues with doing attended transfers.
What happens is Caller A receives a call, they transfer to Caller B,
tell Caller B who is calling, etc.. and then
hit the Transfer key again to transfer the call.
Caller A's side hangs up as expected, but the call is
Looking for two things:
1. Anyone that has dialplan logic for an executive assistant. My owners
want their extensions to ring on her phone, and be very obvious to her which
extension is ringing. They also want her to have presense. She's got Polycom
IP 650 with sidecar, they have IP
Hi,
Before diving into this, I would very pleased to know if someone could yes
or no, successfully compile mISDN 1.1.8 on Lenny (latest RC1 or beta2
version) ?
Regards
After a fresh install on Lenny, I can reproduce at will :
apt-get install build-essential linux-headers-2.6.26-1-686
cd
exten= s,1,Dial
you miss the and the dial params
try whit verbose (some string) and read the terminal...
David
2009/1/8 Ralf Träskman r...@adlibris.com
Hi
I dont understand how to do that, I put in this line to ip-only in
extention.conf
exten=s,1,Dial
No differens
Thanks for all
On January 2, 2009 01:44:14 pm David wrote:
2007
2006
Andrew Kohlsmith 290
2005
Andrew Kohlsmith 731
Damn... I'm slipping! 2nd place in 2005.
-A.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Thu, 2009-01-08 at 16:20 +, Grey Man wrote:
On Wed, Jan 7, 2009 at 6:01 PM, Steve Murphy m...@digium.com wrote:
On Wed, 2009-01-07 at 02:56 +, Grey Man wrote:
On Tue, Jan 6, 2009 at 3:53 PM, Steve Murphy m...@digium.com wrote:
That sounds a bit dangerous to me. If you go down
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
=YES
Of
Jeremy Mann wrote:
Looking for two things:
1. Anyone that has dialplan logic for an executive assistant. My owners
want their extensions to ring on her phone, and be very obvious to her which
extension is ringing. They also want her to have presense. She's got
Polycom IP 650
Klaus Darilion wrote:
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=',
On Thu, 2009-01-08 at 19:24 +0100, Klaus Darilion wrote:
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
WARNING[11982]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax
The is not necessary. Asterisk treats '=' and '=' the same. But you
will need some parameters for Dial to actually do anything.
On Thu, 2009-01-08 at 16:07 -0200, David fire wrote:
exten= s,1,Dial
you miss the and the dial params
try whit verbose (some string) and read the terminal...
Initialize FOOBAR to some know value (ie NO) and change it when you need
to. Then it will always be defined.
Klaus Darilion wrote:
Hi!
I use the following condition:
if (${FOOBAR}=YES) {
...
}
The problem is, that if FOOBAR is not defined at all Asterisk generates
a warning:
I don't think your problem is somehow related to the debian release ...
However since mISDN 1.1.8 was released to support kernel 2.6.24 25 the
18/06/2008
and the kernel 2.6.26 was released the 13/07/2008 an incompatibility
between both
could very well be possible.
Well and it's quite simple,
The Asterisk.org development team has announced the release of Asterisk
1.2.31, 1.4.22.1, and 1.6.0.3. These releases are available for
immediate download from http://downloads.digium.com/.
This update for Asterisk includes a security fix for chan_iax2. Please
see the associated security
2009/1/8 Benoit maver...@maverick.eu.org
I don't think your problem is somehow related to the debian release ...
However since mISDN 1.1.8 was released to support kernel 2.6.24 25 the
18/06/2008
and the kernel 2.6.26 was released the 13/07/2008 an incompatibility
between both
could very
Asterisk Project Security Advisory - AST-2009-001
++
| Product| Asterisk|
My apologies. I had installed a tool for posting Asterisk releases, and
forgot to turn off all the automatic stuff. This was a reply that
reflects my personal opinions on the matter, not all of the Asterisk
Development Team.
My bad :(
Leif Madsen.
Asterisk Development Team wrote:
snip
Grey Man wrote:
A single CDR should not be able to cover several Dial attempts or even
multiple destinations within the same Dial attempts. If you think of
Dial as just another application and of CDRs being designed to reflect
the lifetime of a channel then things are simplified. If a Dial or
Many thanks to all, the Queue(name,r) works like a charm.
On Tue, 2009-01-06 at 19:59 +, Mateusz Pawlowski wrote:
Hi,
I was asked to create a Queue which instead of playing MoH it generates
the ringing tone. I had a look around but could find anything, I would
welcome and help.
For no reason other than it would be cool, I'd like to be able to dial an
extension and have it play a random MP3. Since, however, MP3s are
kinda-sorta weird due to patents, I'm not sure what the right approach for
this is. Any pointers on how to go about this?
Thanks!
-Ken
You can install the custom mp3 player or just convert your mp3's to gsm with
lame. I did this with a free Beethoven overture download and it sounds
pretty good over the phone line.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Ken,
An empty conference call or a parking lot with MOHMP3 both come to mind.
--Dave
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Thursday, January 08, 2009 4:15 PM
To:
Noah Miller wrote:
I don't believe that Polycom's version of SLA does anything with
Asterisk. You have to use asterisk's SLA implementation
(http://www.asterisk.org/node/48342).
So asterisk can't do SLA with Polycom phones?
Asterisk can do SLA with Polycom, just not using
On Thu, 2009-01-08 at 16:15 -0500, Ken D'Ambrosio wrote:
For no reason other than it would be cool, I'd like to be able to dial an
extension and have it play a random MP3. Since, however, MP3s are
kinda-sorta weird due to patents, I'm not sure what the right approach for
this is. Any
You have to enable presence on the polycom phones, then they will read hints
from the default context of your dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Willis
Sent: Thursday, January 08,
Jerry, back in August you were thinking about putting 4 T1 cards in a
single box--did you end up doing that and how did it work out? We're
looking at 700-800 lines for an app and are trying to figure out how
many machines we'll need.
Has anyone else done more than 2 quad T1 cards?
--
Scott
Hi!
It looks like the correct adress. I'm not sure, why this is. I'm sorry, I'm
out of it.
Kindest regards
Julien
Music was my first love and it will be my last (John Miles)
FIND MY WEB-PROJECT AT:
http://ltsb.sourceforge.net
the Linux TextBased
When I set up my Asterisk box at home I didn't want to have to dial 9
to dial off premises, so I gave all my local phones three digit
extensions with this format: 1[1,0]*. My thought is that there are no
area codes that start with 0 or 1, so if I use those numbers, I can
create 20 local
Quoting Thczv F. Thczv thczv.th...@gmail.com:
When I set up my Asterisk box at home I didn't want to have to dial 9
to dial off premises, so I gave all my local phones three digit
extensions with this format: 1[1,0]*. My thought is that there are no
area codes that start with 0 or 1, so if I
Way make it complicated. Make shore Asterisk match internal numbers first.
Else external number.
Trixbox works like that by default.
No nead for 9 to call external numbers.
//Mattias
On Fri, Jan 9, 2009 at 12:07 AM, Thczv F. Thczv thczv.th...@gmail.comwrote:
When I set up my Asterisk box at
This has worked fine for me (as far as I know). Is there some flaw I
am not seeing? I see a lot of small businesses that require a 9 to
dial out, even though they don't have very many extensions. Couldn't
they do what I did and not have to dial 9?
Many older systems _cannot_ process
Thczv F. Thczv wrote:
When I set up my Asterisk box at home I didn't want to have to dial 9
to dial off premises, so I gave all my local phones three digit
extensions with this format: 1[1,0]*. My thought is that there are no
area codes that start with 0 or 1, so if I use those numbers, I
On Thu, Jan 8, 2009 at 7:34 PM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
My apologies. I had installed a tool for posting Asterisk releases, and
forgot to turn off all the automatic stuff. This was a reply that
reflects my personal opinions on the matter, not all of the Asterisk
On Thu, Jan 8, 2009 at 7:22 PM, Asterisk Development Team
asteriskt...@digium.com wrote:
Actually I could see appending a 'servername' to the UUID as useful in a
clustered environment. Every time I don't think I need to do that, I end
up having to do it. And since this would be a configurable
On Thu, Jan 8, 2009 at 6:22 PM, Steve Murphy m...@digium.com wrote:
In general, stuff like FXO's activating, would signal an incoming call,
but gee, I could wire up some odd circuit to make it a local
extension...
Some external rule to the pbx determines which devices and peers you'll
On Thu, 8 Jan 2009, Brent Vrieze wrote:
Thczv F. Thczv wrote:
[snip]
and not
have to use a timeout when dialing long distance.
[snip]
I think you are over thinking this. We set our Asterisk server up with
multiple outgoing dial rules to handle local and long distance. Keep in
mind
On Fri, 2009-01-09 at 01:40 +, Grey Man wrote:
On Thu, Jan 8, 2009 at 7:22 PM, Asterisk Development Team
asteriskt...@digium.com wrote:
Actually I could see appending a 'servername' to the UUID as useful in a
clustered environment. Every time I don't think I need to do that, I end
up
On Fri, Jan 9, 2009 at 3:48 AM, Steve Murphy m...@digium.com wrote:
But, since it is timestamp based, and unique in that the final part was
incremented per request in the same sec, it made a great item to sort
on, and allowed me to implement linkedID's.
Again that's mixing fields that
On Wed, Jan 7, 2009 at 23:39, Noah Miller noahisaacmil...@gmail.com wrote:
Hi Mark -
You really want to do SLA with all 23 lines of the PRI? That's a
lotta lines to be shared. You'd need two sidecars for each phone
(Cisco or Polycom).
Actually there will be multiple PRI's :)
This
On Fri, 2009-01-09 at 01:53 +, Grey Man wrote:
I would be interested in additional information in the CDRs as I'm
sure others would. My worry is it's not a critical peice of CDR
information and because it sounds like information being generated at
the dialplan level it could end up being
On Thursday 08 January 2009 22:24:45 Grey Man wrote:
On Fri, Jan 9, 2009 at 3:48 AM, Steve Murphy m...@digium.com wrote:
My advise is not to. I have no prob with uuids, except that they are
36 bytes, and overkill for uniqueness. linkedID + system name would be
totally sufficient; One glance
On Fri, Jan 9, 2009 at 6:37 AM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
I think Steve is as interested as anybody else in achieving a solution, but
you're hand-waving when it comes to the establishment of a UUID. There
is no such construct that we can use, but there are very
700-800 is the maximum limit without transcoding on very optimized setup. I
would call it suicide without a failover solution. Why dont you consider the
dns srv for load balancing among 2 servers ?
2009/1/8 Scott Plante spla...@insightsys.com
Jerry, back in August you were thinking about
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