Hi,
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n)
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
On Tue, 2008-04-01 at 08:23 -0700, Rizwan Hisham wrote:
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n)
The 'n' flag tells chan_local not to optimize itself out of the call
path. Without the 'n' flag, chan_local will
Rizwan Hisham wrote:
Hi,
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n)
Yes, and you will too when you read localchannel.txt in your Asterisk
source code docs directory.
--
Consulting for
Rule of thumb: you first try without the /n; if the new behaviour is
different from expected, add the /n
:)
Just my $0.02
l.
On Tue, 01 Apr 2008 17:33:05 +0200, Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2008-04-01 at 08:23 -0700, Rizwan Hisham wrote:
Does anyone know the purpose of /n
Does AMD (answering machine detect) need ztdummy or some other timer to
function properly?
--
Drew Miller
Iowa Democratic Party
Information Technology Director
Office: (515) 974-1682
Cell: (515) 451-4509
AIM: ItsDrewMiller
MSN: [EMAIL PROTECTED]
No It does not require.
Regards,
Sanjay.
- Original Message -
From: Drew Miller [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, March 31, 2008 9:17:19 PM (GMT+0530) Asia/Calcutta
Subject: [asterisk-users] Simple Question
Does AMD (answering machine detect) need
The first include references another context within extensions.conf. Contexts
are defined by words in brackets. In your example, there would be a context in
extensions.conf that would look like:
[inbound]
Contexts allow for setting up difference services and difference user
capabilities all
Whats the difference between the following statements in extensions.conf
include=inbound
AND
#include inbound/*.conf
--
Regards
Rizwan Hisham
Software Engineer
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
Rizwan Hisham wrote:
Whats the difference between the following statements in extensions.conf
include=inbound
AND
#include inbound/*.conf
Hi, checkout this page:
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf
With the #include filename statement in extensions.conf,
Whats the difference between the following statements in extensions.conf
include=inbound
AND
#include inbound/*.conf
The first one includes a context the second one includes a file(s).
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.432 / Virus
Hi,
I am planning to deploy an asterisk installation but I need to convince
a few managers that its a good idea.
Theres something I don't quite understand though,
I plan deploy a box on the end of 4 channel BRI ISDN and provide it an
ADSL internet connection.
Should a phone behind the asterisk
PROTECTED] On Behalf Of
Mark Hayward
Sent: Monday, March 20, 2006 8:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] simple question on asterisk
Hi,
I am planning to deploy an asterisk installation but I need
to convince a few managers that its a good
Hey Gurus,
I have a very simple asterisk setup that basically lets me share a PSTN
line from one location to another. I would like to have the phones at
both locations ring when the PSTN # is dialed(inbound calls from PSTN
to asterisk).
I tried something like:
exten =
Martin Joseph wrote:
Hey Gurus,
I have a very simple asterisk setup that basically lets me share a
PSTN line from one location to another. I would like to have the
phones at both locations ring when the PSTN # is dialed(inbound calls
from PSTN to asterisk).
I tried something like:
exten
-Commercial Discussion
Subject: [Asterisk-Users] Simple question about ringing multiple
phones(extensions)?
Hey Gurus,
I have a very simple asterisk setup that basically lets me share a PSTN
line from one location to another. I would like to have the phones at
both locations ring when the PSTN
On Jan 28, 2006, at 12:54 AM, Ronald Wiplinger wrote:
Martin Joseph wrote:
snipI tried something like:
exten = 2020,2,Dial(SIP/2005,25,trIAX/2010,25,tr)
I thought this might cause both 2005 and 2010 to ring when 2020 was
dialed, but only 2005 rings?
Below works for me:
Title: simple question
Hi, list
where i can check the version of asterisk ?
after i do make update on \usr\src\asterisk , what i can do to check and make sure the update is totally successful ?
3ks a lot !
___
Asterisk-Users mailing
On Sun, 14 Nov 2004 16:44:12 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Is this quite simple to set up and can I attach asterix to my landline via a
standard modem?
Yes no go to http://www.voip-info.org/wiki-Asterisk
and read learn try and read try agin
Jason
Hi there, I was wondering if youd be able to answer a
question for me. I want to run an asterix system in my house. My main goal is
for it to pick up my landline (via a modem) and then have a push button system
i.e. push one for luke push two for johnetc and then divert it to the
desired
Something the user list in
Microsoft Messenger. I was thinking on some sort of web page that can
check the registration of the sip clients on the asterisk but want to
know if already exist to avoid to reinvent the wheel.
That is actually quite easy and there are some projects that achive this
: September 09, 2004 8:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Simple question about SIP community
On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote:
we have a community of people on an * box that use SIP softphones to
talk each
other. Can you
Marcello Lupo wrote:
Hi to all,
we have a community of people on an * box that use SIP softphones to talk each
other. Can you suggest me the quickest and simple way to let someone know who
is online without have to call one by one the persons to look if they are
present or not?? Something the
Hi to all,
we have a community of people on an * box that use SIP softphones to talk each
other. Can you suggest me the quickest and simple way to let someone know who
is online without have to call one by one the persons to look if they are
present or not?? Something the user list in Microsoft
On Sep 9, 2004, at 8:53 AM, Marcello Lupo wrote:
we have a community of people on an * box that use SIP softphones to
talk each
other. Can you suggest me the quickest and simple way to let someone
know who
is online without have to call one by one the persons to look if they
are
present or
it just came to my mind, and i haven't done any
researches yet if somebody tried this one with asterisk.. :) well just in case
somebody or someone on the list aware, i appreciate any advise.
in telco world, there's like 64kbps per channel and
voice can be carried on a 16kbps channel. is it
On Thu, 2004-01-22 at 21:55, Jess Magnaye wrote:
it just came to my mind, and i haven't done any researches yet if
somebody tried this one with asterisk.. :) well just in case somebody
or someone on the list aware, i appreciate any advise.
in telco world, there's like 64kbps per channel and
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal or another
SW wrote:
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal
: Re: [Asterisk-Users] simple question on sip.conf
SW wrote:
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] simple question on sip.conf
SW wrote:
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other
30 matches
Mail list logo