HY all,
I am completely new to the asterisk so can any one help me with it
as I have some questions queries
1. first n for most what are the tools/equipment that I need
for eg a Computer, and a net connection is it all that I need for simple head
start to get hands on the asterisk
On Wed, Jun 2, 2010 at 7:57 AM, prashant shrestha
prashantshres...@live.com wrote:
I am completely new to the asterisk so can any one help me with it
as I have some questions queries
Welcome to the Asterisk community, Prashant
1. first n for most what are the tools/equipment that I need
for
Hello Russell!
That's a pitty.
But even so, this time the problem is a different one. Before we had the
problem, that I couldn't really make a call at all, or that the call crashed
quickly. This time, the call connects and seems to be stable enough, but I
have no sound from the other
On 02/06/10 06:57, prashant shrestha wrote:
HY all,
I am completely new to the asterisk so can any one help me with it
as I have some questions queries
1. first n for most what are the tools/equipment that I need
for eg a Computer, and a net connection is it all that I need for
simple head
Asterisk 1.6
CentOS 5.0
All -
I'd like to offer my users the ability to hangup a call by pressing **. I'm
using an attendant, so when ** is dialled I'd like processing to return to
the attendant so the user can place a subsequent call. I have setup
features.conf to include:
[featuremap]
- Original Message -
The Asterisk Development Team has announced the release of Asterisk
1.6.2.8. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Which release will http://issues.asterisk.org/view.php?id=17135 make it into;
was
Also check the codecs as if you are using g729 or g723, there is a chance that
they are not available in codecs directory ( /usr/lib/asterisk/modules).
-THQ- !!!ONE
Date: Tue, 1 Jun 2010 19:24:41 -0400
From: zisha...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re:
hello Ishfaq, Welcome to Asterisk community. If you are a beginner and want to
learn Asterisk download a book Future of Telephony it will provide you with
the basic knowledge.
Regards
Taimur Hasan
-THQ- !!!ONE
Date: Wed, 2 Jun 2010 10:17:10 +0100
From: i...@pack-net.co.uk
To:
Hi
Funnily enough that was my advice (with link) to Prashant who is the
topic starter
Ish
On 02/06/10 13:37, taimur hasan wrote:
hello Ishfaq, Welcome to Asterisk community. If you are a beginner
and want to learn Asterisk download a book Future of Telephony it
will provide you with
Continuing the top posting party...
And, funny enough, you were kind enough to do it without posting your signature
in huge HTML centered text as though you felt it necessary for everyone in the
world to know your initials and that you enjoy internet memes.
--Tim
- Ishfaq Malik
Hi Parashant,
As a new comer don't let it bother or discourage you. This happens here
sometimes on this mailing list, first somebody posts something, and the
replys later convert into an unnecessary and off the topic argument ground.
Soon you will figure out which posters are here to help, which
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},America/New_York,%F_%T)})
instead of this:
exten =
Hi Parashant,
As a new comer, don't let it bother or discourage you. This happens here
sometimes on this mailing list, first somebody posts something, and the replies
later get put at the top of the email and continue to do so. This makes it
incredibly hard to read what is going on. It has
Seems like Tim is the new inspector here. Nothing better for him to do.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-06-02 10:35 AM, Tim Nelson tnel...@rockbochs.com wrote:
Hi Parashant,
As a new comer, don't let it bother or discourage you. This happens here
Tim, may be you should receive your asterisk mailing lists mail on gmail, so
somthing similar, where you would not have these html, or clutter of emails
issue. Obviously you are upset on something which my email client didn't
show me, and probably others didn't see it either, but you did.
Zeeshan
Prashant wrote:
I am completely new to the asterisk so can any one help me with it
as I have some questions queries
1. first n for most what are the tools/equipment that I need
for eg a Computer, and a net connection is it all that I need for
simple head start to get hands on the
Hello Mike,
the semi-column did not really work :
Jonas Kellens jo...@mail1.be:i...@mail2.be: malformed address:
:i...@mail2.be may not follow Jonas Kellens jo...@mail1.be
and same result with a comma :
Jonas Kellens jo...@mail1.be,i...@mail2.be: malformed address:
,i...@mail2.be may
khalid touati wrote:
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},America/New_York,%F_%T)})
instead of this:
exten
Lol sorry Ishfaq i misread the mail.. i thought u are asking about the solution
lol but i think these type of mistakes are necessary in learning process.
-THQ- !!!ONE
Date: Wed, 2 Jun 2010 10:17:10 +0100
From: i...@pack-net.co.uk
To: asterisk-users@lists.digium.com
Subject: Re:
On Mon, 2010-05-24 at 14:41 +0100, Kingsley Tart wrote:
On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote:
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call
is always a ring group called '600', my problem is that after press 1 (but
this problem is present also with
I've been struggling with a Trixbox running Asterisk 1.6 for one of our
customers as of late.
The service provider in question is using BroadWorks and requires a single
trunk registration for the trunk group. We have 4 users(lines/numbers) in the
TG.
The sip trunk is setup as follows:
thanks i'll keep that in mind.
2010/6/2 Roderick A. Anderson raand...@cyber-office.net
khalid touati wrote:
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
--[ UxBoD ]-- wrote:
- Original Message -
The Asterisk Development Team has announced the release of Asterisk
1.6.2.8. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Which release will
I have remote access to the server so I checked the canreinvite .. they
are all set to no. I can't try the call from here, I will get back to you.
Gary Baribault
On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote:
Do you agree something is blocking the audio in one direction? Can you
do a 'rtp
I have checked, the users have ulaw, then alaw, the phones are set to
711u then 711a which is the same thing (I think).
Gary Baribault
On 06/02/2010 08:32 AM, taimur hasan wrote:
Also check the codecs as if you are using g729 or g723, there is a
chance that they are not available in codecs
On Wed, Jun 2, 2010 at 12:11 AM, velusamy Krishnan velu.techni...@gmail.com
wrote:
Dear Tilghman,
In some cases while transfer the call to an agent, I need to set the
some value(ex. customer ID) in UUI. So that agent would know who is calling.
Kindly provide me some solution.
Hi,
I set up an asterisk server that i use with iax accounts. Everything is working
fine, but, for personnal reason i need to insert a home made proxy on my server
which role is to encrypt data.
Here i encountered what i suppose to be a timeout prb : i can register with my
account but i can not
I don't know if this makes any difference, I created a lot of this
configuration with the Asterisk-GUI (SVN-branch-2.0-r4980) and when I
edit the users.conf file, there are two entries 'type = peer' for each
extension and they are highlighted in red!
Gary Baribault
On 06/02/2010 08:32 AM, taimur
On Wed, Jun 02, 2010 at 10:26:12AM -0400, khalid touati wrote:
Hi Guys,
for people who may have the same issue:
i was just not using STRFTIME the right way, after consulting docs, i'm
using it like this:
exten =
thank you Barry, you're right, it is also working.
well, happy that i have a bunch of choices that work (after wrong output).
thanks for all!
2010/6/2 Barry Miller asterisk-us...@notanet.net
On Wed, Jun 02, 2010 at 10:26:12AM -0400, khalid touati wrote:
Hi Guys,
for people who may have the
On Mon, May 24, 2010 at 9:41 AM, Kingsley Tart kings...@skymarket.co.uk wrote:
I know nothing of Trixbox but I had a problem with my own dialplan where
there was a delay with the user selecting 0 from my IVR menu. It turned
out that because my extensions all started with 0 (they were real phone
It could be, not sure, that proxy is spoofing ip adress.
http://en.wikipedia.org/wiki/IP_address_spoofing
Jose Flores Galicia
floj...@gmail.com
BriefCode Code Based Training
2010/6/2 isca...@free.fr
Hi,
I set up an asterisk server that i use with iax accounts. Everything is
working
Hello everyone!
So I hacked app_jack.c today, as best I could. Whic came mostly down to
inserting ast_log() messages.
I discovered the following with JACK:
When it starts, it tries to read 512 bytes and only gets 0. That clears up
after a while.
Sometimes a good time later than the
Oh P.S.:
I changed my jackd startup options as well from:
jackd --tmeout 4500 -R -d alsa -d hw:1 -r 48000 -z shaped
and then in case a:
-p 64 -n 2
Case b:
-p 1024 -n 3
case c;
-p 128 -n 2
Case c is my default setting.
The rountrip time to talk.google.com is 60.440ms average. My upstream
Honestly I would try and get your JACK configuration working with a more
stable/supported channel driver such as chan_sip. The Google Talk integration
may or may not work, depending on the scenario, but because you're using two
technologies that are not heavily developed in Asterisk, it might
The Asterisk Development Team has announced the release of version 1.4.11.1 of
libpri. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/
This release fixes a regression in multi component FACILITY messages and
includes a minor bug fix for BRI
Is there a reasonably easy way to increase the volume on a DAHDI
channel? The VOIP phones in the house work OK, but for the phones
connected to DAHDI channels on a Digium TDM400P card, the volume is very
low and it's hard to hear if there is any background noise at all. If
this is documented,
Hello Leif!
The issue about gtalk and jack was originated by me. Yet in those days, the
problem was of a different nature.
Still, why do I use these two and no other channel or software? Well I've
mentioned, that Asterisk is the only commandline phone on Linux I know, that
supports JACK.
On Wed, 2010-06-02 at 15:35 -0600, Greg Woods wrote:
Is there a reasonably easy way to increase the volume on a DAHDI
channel? The VOIP phones in the house work OK, but for the phones
connected to DAHDI channels on a Digium TDM400P card, the volume is very
low and it's hard to hear if there is
El 02/06/10 16:35, Greg Woods escribió:
Is there a reasonably easy way to increase the volume on a DAHDI
channel? The VOIP phones in the house work OK, but for the phones
connected to DAHDI channels on a Digium TDM400P card, the volume is very
low and it's hard to hear if there is any
I have an asterisk system in Costa Rica that connects to a SIP provider in
Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickly.
In trying to debug this I turned on SIP debug in Asterisk and the SIP provider
enabled packet capture on his end.
What I saw was me sending
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, John.
On Fri, May 21, 2010 at 23:35:41 -0300, John Novack wrote:
Another thing I want to try is to connect Asterisk with Siemens PBX
so that the extensions on Asterisk can communicate with the
extensions on the Siemens PBX and vice versa. For
Daniel Bareiro wrote:
SNIP SNIP
Thanks for the explanation and clarification of nomenclature. And in what
cases it would be correct to use the RJ designation?
Thanks for your reply.
Regards,
Daniel
RJ is short for Registered Jack. Strictly speaking it is only correct
when it is an
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