>> On Thu, Jul 01, 2010 at 10:19:08PM -0500, Karl Fife wrote:
>>> Calls that come in on DAHDI FXO ports are routed to [context],
>>> extension 's'
>>>
>>> INSTEAD, I would like to route specific ports to specific extensions,
>>> For example:
>>>
>>> I want DAHDI/1-1 to go to 1234
>>> I want DAHD
Hi
I have created meetme with 3 user. When i going to mute user it gives
following error..
*Asterisk Version : 1.6.2.6*
-- Playing 'conf-muted.gsm' (language 'en')
[Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid
GSM data (1)
[Jul 2 22:46:51] WARNING[10823]: tran
Original Message -
From: "Barry Miller"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, July 01, 2010 11:53 PM
Subject: Re: [asterisk-users] DAHDI FXO calls and the 's' extension.
No,Jackie-O doesn't work here--it's just an example. Sheesh!
> On Thu, Ju
On Thu, Jul 1, 2010 at 3:26 PM, wrote:
>
>
>
>
> -Original Message-
> From: Ryan Wagoner
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Sent: Thu, Jul 1, 2010 6:19 pm
> Subject: Re: [asterisk-users] Update the LCD with the callee's name after
> dialing
>
> On Thu, Jul
Hi Giorgio,
Why don't you terminate calls on the cisco router via SIP?
--
Message: 11
Date: Fri, 02 Jul 2010 18:54:31 +0200
From: Giorgio Incantalupo
Subject: [asterisk-users] asterisk and cisco 2800
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 6/30/2010 3:56 PM, Alex Villacís Lasso wrote:
> whenever an ISDN port is in RED alarm (unsynchronized), we get a stream of
> warnings in /var/log/asterisk/full that look like this:
>
> [Jun 30 17:38:41] WARNING[9637] chan_dahdi.c: No D-channels available!
> Using Primary channel 78 as D-chan
I use a release of putty called putty tray available at
http://haanstra.eu/putty/ for its URL clickability
This is what ubuntu does, for some reason not for screen sessions
export PROMPT_COMMAND='echo -ne "\033]0;${us...@${hostname}:
${PWD/$HOME/~}\007"'
One of the distros sets it so that it says
On Sat, Jul 03, 2010 at 01:33:25AM +0300, eyal goltzman wrote:
> Hello,
>
> Can I use AMI Originate to call 2 outside numbers (SIP) and connect them?
> How?
Originate one channel to the application Dial to dial to the other
channel?
--
Tzafrir Cohen
icq#16849755 jabb
Hello,
Can I use AMI Originate to call 2 outside numbers (SIP) and connect them?
How?
Thanks
Eyal
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webina
On 07/02/2010 10:10 AM, Gordon Henderson wrote:
>
> I've just posted this to another list where we were talking about the same
> old issues we've been plagues with recently - I'd already posted some
> iptables rules, but added more to it for this...
>
> This script probably isn't compatable with an
On Fri, 2 Jul 2010, Gordon Henderson wrote:
> The file is at http://unicorn.drogon.net/firewall2
Lots of cool stuff in here. It's going to take a bit to understand it
all :)
--
Thanks in advance,
-
Steve Edwards sedwa
Kyle-
C5441 is a year 2000 DSP chip. If you're considering the hardware/TI DSP chip
path,
C64x or C64x+ series is higher performance (higher channel capacity) and more
relevant in terms of available suppliers, forum tech support, TI support, etc.
-Jeff
Kyle Kienapfel wrote:
>
> For those cod
> but i can't find header-files or dev-files there.
>
include folder
--
It's that easy? Ok, so stupid question from me.
But thanks for your help.
=
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.
On Fri, Jul 2, 2010 at 1:51 PM, wrote:
> but i can't find header-files or dev-files there.
>
include folder
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
___
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
Sent: Friday, July 02, 2010 12:52 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk header files for Debian Lenny
1.6.1.20
>
> This are the header files for 1.4, not for 1.6.
>
Then how did you install asterisk 1.6?
--
from here http://downloads.digium.com/pub/asterisk/asterisk-1.6.1-current.tar.gz
but i can't find header-files or dev-files there.
--
On Fri, 2 Jul 2010, Gordon Henderson wrote:
> I'll call this activity "slamming".
At least in the US, slamming already has a specific telephony meaning.
http://en.wikipedia.org/wiki/Telephone_slamming
--
Thanks in advance,
---
On Fri, Jul 2, 2010 at 1:38 PM, wrote:
> This are the header files for 1.4, not for 1.6.
>
Then how did you install asterisk 1.6?
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
__
-Original Message-
From: Paul Belanger
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Fri, Jul 2, 2010 7:35 pm
Subject: Re: [asterisk-users] Asterisk header files for Debian Lenny 1.6.1.20
On Fri, Jul 2, 2010 at 1:22 PM, wrote:
> Where can i get the
On Fri, Jul 2, 2010 at 1:22 PM, wrote:
> Where can i get the header files Asterisk 1.6.1.20 for Debian Lenny?
>
$ apt-get install asterisk-dev
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
On Friday 02 Jul 2010, Tim Nelson wrote:
> - "A J Stiles" wrote:
> > On Friday 02 Jul 2010, Ira wrote:
> > > At 11:14 PM 7/1/2010, you wrote:
> > > >Same activity from these IPs:
> > > >174.129.137.135
> > >
> > > Given that my Asterisk box is used for nothing but Asterisk and I
> > > know the
Hi,
one question again from an asterisk newbie.
Where can i get the header files Asterisk 1.6.1.20 for Debian Lenny?
I need them to install chan_capi for my Diva E1 Server Card.
Thanks in advance.
--
_
-- Bandwidth and Colocati
I've just posted this to another list where we were talking about the same
old issues we've been plagues with recently - I'd already posted some
iptables rules, but added more to it for this...
This script probably isn't compatable with anything else, but I don't run
anything else. It's also d
- "A J Stiles" wrote:
> On Friday 02 Jul 2010, Ira wrote:
> > At 11:14 PM 7/1/2010, you wrote:
> > >Same activity from these IPs:
> > >174.129.137.135
> >
> > Given that my Asterisk box is used for nothing but Asterisk and I
> > know the small number of IPs that need to have access is there an
Hi all,
I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures
with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the
cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives
no errros, the span is up and active, green light on the card) but when
I
On Friday 02 Jul 2010, Ira wrote:
> At 11:14 PM 7/1/2010, you wrote:
> >Same activity from these IPs:
> >174.129.137.135
>
> Given that my Asterisk box is used for nothing but Asterisk and I
> know the small number of IPs that need to have access is there an
> easy way to use iptables to block ever
Hi Matt,
What eaxtly you mean by Fail2ban crapping out? I never had any problem with
it, and for me it is not only protecting asterisk, but also multiple
websites for wrong logging attempts, spams and SQL injections. Based on your
experience I would like to see if I need to be careful with its set
I've noticed from time to time, that fail2ban just craps out, so, this might
be of interest to the community assuming you use 192.168.100.0/24 on your
network
iptables -A INPUT -s 192.168.100.0/24 -j ACCEPT
iptables -A INPUT -s carrierip.x.x.x -j ACCEPT
iptables -A INPUT -s 127.0.0.1 -j ACCEPT
At 11:14 PM 7/1/2010, you wrote:
>Same activity from these IPs:
>174.129.137.135
Given that my Asterisk box is used for nothing but Asterisk and I
know the small number of IPs that need to have access is there an
easy way to use iptables to block everything but those 6 IPs and
provider addresse
On Fri, 2 Jul 2010, Kenny Watson wrote:
> for i in `ls -R /var/lib/asterisk/sounds/uk/*wav`; # do recursive ls and
> only list wav files and loop through each one
> do # start do loop
> CONV=`echo $i|sed 's/.wav/.g729/g'` # set CONV variable as filename with
> wav swapped for G729
>
Danny,
thank you for you feedback.
I have the following setting in sip.conf :
limitonpeer = yes
and for every sip peer definition I have :
asterisk*CLI> sip show peer test1
* Name : test1
Realtime peer: Yes, cached
Secret :
MD5Secret:
Context : from-TEST
Su
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Friday, July 02, 2010 3:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote Party ID issu
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, July 02, 2010 4:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer fails
Hello list,
this i
Hello all!
After a recent upgrade of Asterisk 1.4.18 to 1.4.33.1, everything
looks ok, except when fax T.38 calls are being passed through the
server, the fax transmission fails, giving a "train failure" message
when invoking "fax show stats" in the fax server (Asterisk 1.6.0.26).
Rxfax and Txfax
On 2 Jul 2010, at 13:31, unsero...@aol.com wrote:
> I just did not want to spam the list with useless content but just reply to
> you as you attacked me.
> This was the reason i only replied to you and not to the whole list.
> But as i realised now it seems to be usual to spam the whole mailing
I just did not want to spam the list with useless content but just reply to you
as you attacked me.
This was the reason i only replied to you and not to the whole list.
But as i realised now it seems to be usual to spam the whole mailing list with
useless information like your mail (or this mail
On 2 Jul 2010, at 12:29, John Novack wrote:
> regardless, people will post either way, and wasting archive space
> complaining about either one is pointless.
I was mainly pissed off about him directly replying to people (i.e. me) rather
than the list. It was you lot that started the religious
Tilghman Lesher wrote:
> On Thursday 01 July 2010 20:45:14 John Novack wrote:
>
>> John Ervin wrote:
>>
>>> Where are the rules for posting in this discussion group? Just curious.
>>>
>>>
It's a rule on this list, although it's frequently ignored.
>> Further s
Hi Guy's,
i'm having a wheird problem with an asterisk-trunk installation, dahdi via
hfc-s sounds extremely distorted, but the same installation (same libpri,
same dahdi etc) is ok with asterisk 1.4. I dont see any errors, neighter on
the console nor the error logs. Any ideas what was changed? The
- Original Message -
From: "Paul Belanger"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, 29 June, 2010 10:22:18 PM
Subject: Re: [asterisk-users] Voiceprompts i.e. voicemail and conferencing
in multiple codecs
On Tue, Jun 29, 2010 at 12:51 PM, Kenny W
Hello list,
this is the dialplan :
exten => s,n,Dial(SIP/test1&SIP/test2,,t)
exten => 10,1,Dial(SIP/test1)
exten => 20,1,Dial(SIP/test2)
So there is an incoming call that rings SIPaccounts test1 and test2.
Account test1 answers and wants to transfer the call to test2.
Transfer is : #20
Thi
On Thursday 01 July 2010 21:59:21 Zhang Shukun wrote:
> hi, all
>
> recently, i face a GotoIfTime problem
>
> GotoIfTime("08:00:00-07:00:00,mon-sun,*,*?95040263008,start")
>
> as you can see the section is 08:00:00-07:00:00 , which is the begin
> time is later than the end time
>
> what's this
On Thursday 01 July 2010 20:45:14 John Novack wrote:
> John Ervin wrote:
> > Where are the rules for posting in this discussion group? Just curious.
> >
> >> It's a rule on this list, although it's frequently ignored.
>
> Further searching shows there is NO written rule regarding top bottom or
> e
David
I am using FREEPBX/TRIXBOX
Sam
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Wednesday, 30 June 2010 8:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
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