[EMAIL PROTECTED] could be a better start for beginners (but beware, the
installation CD will format your HD without asking).
http://asteriskathome.sourceforge.net/
On Tue, 2006-04-25 at 10:47 +0800, rommel malana wrote:
Goodday,
I'm an opensource fanatic and I have already installed
Hi Pasqualotto,
Actually, I've seen your post on Asterisk-Users list yesterday, but I could
not understand back then. Now, I've checked your sip configuration again, I
think you make a mistake in type of sip account. I use friend not
peer. I am not sure though.
Following is what I had in my
I vote for the raw file format, due to the reasons listed here:
http://www.orderlyq.com/asteriskqueues.html
Also there is a note on the same page as follows:
Note: As of Asterisk 1.2.0, Mpg123 is no longer used by Asterisk, which has
its own MP3 player. However, it's still a good idea to play
]
Soner Tari wrote:
I vote for the raw file format, due to the reasons listed here:
http://www.orderlyq.com/asteriskqueues.html
Of course you need to convert all mp3 moh files to raw format manually,
but it's easy as described there.
We were using the rawplayer method on our server
We switched to that after mpg123 wouldn't compile on our newer 64bit
machines
You might have successfully compiled with make linux-devel perhaps. I did.
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To
I have Data Sheet for 942 from Linksys web site. It says this on page 4
(close to bottom):
Physical Interfaces:
2 100baseT RJ-45 Ethernet Ports (IEEE 802.3)
And that was one of the reasons I was considering 942. Do you think the data
sheet may be wrong?
- Original Message -
From:
Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 20, 2006 1:01 PM
Subject: Re: [Asterisk-Users] Grandstream GXP-2000
On Mon, 20 Feb 2006, Soner Tari wrote:
I have Data Sheet for 942 from Linksys web site. It says this on page 4
(close
I'm interested in your first problem too. If you were able to add the same
CallerID to, say, 2 calling cards, which one is supposed to be selected
after CallerID is acquired? Since this is not defined, the easiest solution
is that it should not be allowed in db, hence A2Billing complains
Bill, check the following thread to see if you can find some answers:
http://lists.digium.com/pipermail/asterisk-users/2005-August/123548.html
- Original Message -
From: Bill Michaelson [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 08, 2005 8:39 PM
Assuming SIP users dial 9 to get FXO lines, and callerid's for them are set
as 11 and 12 in sip.conf, and Zap lines are also 1 and 2, you could do it
easily:
exten = 9,1,Dial(Zap/${CALLERIDNUM:1})
You can manipulate dial string further with arithmetic expressions too.
Hope this helps,
Soner
Alvaro, I don't have a solution but a question, if A is not behind NAT why
do you have nat=yes for it?
Soner
- Original Message -
From: Alvaro Parres [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, November
If you have 2.6 kernel and/or are using udev, you want to check
README.Linux26 and README.udev in zaptel, if you haven't already done so.
Hmmm we have this TDM400 with one FXO module on it, we're using it for
testing purposes.
I can:
modprobe zaptel
modprobe wctdm
and these get load
- - - - - - - - - - -
speex - - - - - - - - - - -
ilbc - 4 3 3 3 3 2 4 - - -
Cheers,
Soner
- Original Message -
From: Soner Tari [EMAIL PROTECTED
Hi All,
When I do 'show translation' on a Linux asterisk 2.6.9-11.EL #1 Wed Jun 8
16:40:06 CDT 2005 x86_64 x86_64 x86_64 GNU/Linux
and
Asterisk CVS HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on
2005-10-19 19:07:08 UTC
I have very strange lpc10 and ilbc rows (sorry the columns
Ozan,
Put the following to sip_custom.conf:
[OpenSER]
type=peer
username=8333688231
secret=test
host=212.154.104.198
fromuser=8333688231; some of the following may not be necessary
fromdomain=212.154.104.198
nat=yes
dtmfmode=rfc2833
disallow=all
allow=g729; whichever codec you want
Sorry, I meant:
exten = _0.,1,Dial(SIP/[EMAIL PROTECTED],60,T)
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, September 12, 2005 3:25 PM
Subject: Re: [Asterisk-Users
,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2005-09-10 19:22:35','\Soner Tari\ 21','21','19','ichat',
'SIP/21-efcb','Zap/5-2','Dial','Zap/5|24|rTtWw',34,28,'ANSWERED',3,'','1126369355.6')
Sep 10 19:23:09 DEBUG[27367
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2005-09-10 19:22:35','\Soner Tari\ 21','21','19','ichat',
'SIP/21-efcb','Zap/5-2','Dial','Zap/5|24|rTtWw',34,28,'ANSWERED',3,'','1126369355.6')
Sep 10 19:23
: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2005-09-10 19:22:35','\Soner Tari\ 21','21','19','ichat',
'SIP/21-efcb','Zap/5-2','Dial','Zap/5|24|rTtWw',34,28,'ANSWERED',3,'','1126369355.6
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid)
VALUES ('2005-09-10 19:22:35','\Soner Tari\ 21','21','19','ichat',
'SIP/21-efcb','Zap/5-2','Dial','Zap/5|24|rTtWw',34,28,'ANSWERED',3,'','1126369355.6')
Sep 10 19:23
Canuck15,
No, I hadn't played with the gains. But I've now done so and no difference
unfortunately. Thanks for the suggestion though.
I have discovered that after Asterisk has answered the call and the remote
caller has hung up, if I lift the receiver on a phone connected to the
line
(in
Did you search the maillist archives for hybrid echo cancellation?
Hello
In the following setup:
call coming from a pstn line - into FXO card - asterisk - SIP phone
i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything
i
speak into SIP phone microphone i hear in its
Did you search the maillist archives for hybrid echo cancellation?
well, yes i googled a lot beforehand, came across the hybrid issue, but
from
what i unerstand, the hybrid is a piece of hardware that sits on the
X100P
card. I'm not sure what can be done about it - the card doesn't seem to
echotraining=yes
echotraining=800
This looks odd to me, I would use just:
echotraining=800
Gain setting are important of course. You could use ztmonitor for that.
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Asterisk-Users mailing
Gain setting are important of course. You could use ztmonitor for that.
the asterisk server is a racked machine with no sound card. so can't use
the
ztmonitor. If everything fails i'll dig it out and try this
You don't need a soundcard to use ztmonitor, what do you mean by that?
Marek, you
: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 07, 2005 4:59 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Motherboard and processor recommendations
On Wed, Sep 07, 2005 at 11:02:58PM +0300, Soner Tari wrote:
Hi All,
For sometime now I've been searching
Hi All,
For sometime now I've been searching the wiki and googling, but I think I'm
missing some of the very important answers. So I'll have to ask this to the
list.
I'm trying to decide on the right motherboard and processor. Here are my
questions:
1. Would I have problems with
I think you can do it in extensions.conf. I would use a special macro only
for the operator, where the call would be directed back to the operator
instead of voicemail, if the transferee is busy or unavailable. Sounds
simple to me.
- Original Message -
From: Paradise Dove [EMAIL
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488
web admin page you enter these registration values. When you reboot the
HT488 you should see it registering on Asterisk CLI.
What's left is a
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount in sip.conf. On the FXO section of HT488 web
admin page you enter these registration values. When you reboot the HT488
you should see it registering on Asterisk CLI.
What's left is a
Of course... Those are the basics to get HT488 working for the OP. In this
thread I am not trying to show how to create dialplans.
On Tue, 2005-08-30 at 17:11 +0300, Soner Tari wrote:
I use HT488, and I can make and receive FXO calls. It's actually quite
simple, you create a SIP acount
If nic is loaded using modprobe - you can set options for duplex -
depending on the nic...
See /etc/modules.conf
I assume you really meant /etc/modprobe.conf ;)
Rich, he really means /etc/modules.conf. He is using 2.4 kernel.
I know, because the reverse happened to me :), see this:
Are changes to the zapata.conf file read on the fly or do you have to
restart the asterisk process?
I've never seen any .conf changes activated without reload.
On CLI, try this:
reload chan_zap.so
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In my experience, rxgain and txgain settings is changing without full
restart, reload of chan_zap.so seems to be enough. Please let me know if you
think this is problematic. I don't know about other zapata.conf parameters.
- Original Message -
From: Samy Kamkar [EMAIL PROTECTED]
To:
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the
I'd suggest turning off echotraining on the FXS altogether, and perhaps
even
killing the echocanceller on FXS entirely. (you won't be getting
significant
echo from the FXS, and the FXO should be handling it anyway) --
echocancelwhenbridged might be an interesting thing to play with as well.
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the coarse-training
I have splitters on 2 of the 3 PSTN lines. As I mentioned in my previous
posts, the echo performance of my system is not so bad, but does anybody
know if ADSL splitters may cause echo? After all, splitters have some
circuitry, and my wild guess is that that may influence the characteristics
of
Hi,
I'm not the OP, but I had a similar problem, in my case fxotune ran
successfully for just one out of 3x FXO modules, but the coefficients were
all 0's. My kernel is 2.6.11 on CentOS 4.1.
So I'm curious if 2.6 kernel (instead of 2.4) has any input in this whole
echo issue, not just
Is it practical to 'assume' that in your case mentioned above that
#1 is not going to occur again (since I assume when you say 'line'
you are referring to an outside pstn line), and, #2 is in a mode
of fine-tuning the training when in fact you'd really like it to
start the coarse-training from
Hi,
I use Grandstream HT486 with Asterisk. I dial 9 to get an FXO line, then
hangup and wait 1-2 secs. Then, I dial immediately 9 again to get another
outbound channel and hangup again. Guess what happens, the two outbound
lines are connected indefinetly.
The cause of this issue is that I
It sounds like web_hostname in your /var/www/html/panel/op_server.cfg is set
to your external ip. If you change it to your internal ip, I think you'll
have the opposite of what you describe. I couldn't find a decent solution to
this dilemma. Any one?
- Original Message -
From: Jaco
In theory, using a URL instead of an IP address in web_hostname should solve
this issue. But last time I tried it didn't work (you may need to set DNS
records of your ADSL modems). I should try again.
- Original Message -
From: Jaco vd Westhuizen [EMAIL PROTECTED]
To:
Because 800$2000$ (interms of hardware).
(For 12 Ports: The min. price of T1(500$)+channel bank($1,500)=2000$.
While the min. price of 3 TDM400P with 24 modules is just=800$.
Your solution is still 400$ more expensive for 24 ports.)
Sorry but to have 24 ports you need 6x TDM cards, which adds
Since you say TDM04B, I guess you are talking about noise on FXO ports. So,
my guess is that your opermode setting for zaptel driver is wrong? This may
be true especially if the noise can be avoided or considerably reduced by
playing with rx/txgain. At least that was the experience I had with
Hi All,
I am experiencing a very strange problem. I call the FXO channels (Zap/2 and
3) almost at the same time, and then hang both up. The operator extension is
Zap/6, and after the greeting message Zap/6 starts to ring (there is no
disconnect supervision here, and I disabled the busy detect
And btw, I have CentOS 4.1. Could this be related with 2.6 kernel?
Hi All,
I am experiencing a very strange problem. I call the FXO channels (Zap/2
and 3) almost at the same time, and then hang both up. The operator
extension is Zap/6, and after the greeting message Zap/6 starts to ring
Hi Ceyhan,
We are specialized on Asterisk. Please check http://kulustur.com
Please also see
http://www.voip-info.org/tiki-index.php?page=Asterisk+consultants
for other consultants in Turkey, especially EMEA and Europe sections.
I can try to answer your questions via email. But, people in this
It looks from here like you've rebooted the system after checkout, but your
system was not configured to load zaptel drivers at boot time.
Have you forgot to do 'make config' while in /usr/src/zaptel ?
Hope this helps
- Original Message -
From: David Romero [EMAIL PROTECTED]
To:
See this:
http://lists.digium.com/pipermail/asterisk-users/2005-June/110524.html
Free for non-commercial use.
- Original Message -
From: Erdem HAK [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, June 10, 2005 11:53 AM
Subject: [Asterisk-Users] G.729AB codec support
I'm saying free for non-commercial use, you're saying Intel is free for
non-commercial use. And I point to the Intel code. And there is no fee for
the licence for non-commercial use.
So what is completely wrong about my post?
- Original Message -
From: Zoa [EMAIL PROTECTED]
To:
Did you install G729 codec and changed sip.conf accordingly? Or is it
just announcements?
On Fri, 2005-06-03 at 21:59 +0800, Nathaniel Angelo A. Torres (247talk)
wrote:
Hi, Im using an Asterisk Server and a Cisco AS5350. They are
interconnected via Sip. When I tried using G.729 codec, all
]
[mailto:[EMAIL PROTECTED] On Behalf Of Soner Tari
Sent: Friday, June 03, 2005 11:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] G.729 with RVA
Did you install G729 codec and changed sip.conf accordingly? Or is it
just announcements?
On Fri, 2005
I am in Turkey. I imagine this is due to incorrect zone information, but I
can't seem to be able to find the correct values for Turkey. I tried
guessing them with no luck.
DTMF tones in Turkey are the same as the standards everywhere. The other
signalling tones are different (such as dial
I did the modification that Rich explains in his email on March 23rd below.
I believe it works for me, because before this mod I was getting Ouch, part
reset... errors at least once a week, rendering * unsuitable for production
systems. After this mod, the system is running flawlessly for
If your problem is the same as mine then you need
to use busydetect. In Turkey, we don't have
polarity reversal, and signalling tones are quite different.
But just enabling busydetect in zaptel.conf did not
help me, it may work for you though. I had to change relevant compile options
and
refer to Martin's algorithm. Can you provide more details please?
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup
detection failing
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi all,
I am having problems with hangup detection on fxo line.
In Turkey, we don't have polarity reversal, so hanguponpolarityswitch option
in zapata.conf does not help.
The callprogress in dsp.c is just for the US, and the way it does call
progress is quite unsuitable for the tones in Turkey,
After ztcfg, /var/log/messages reads
Module 3: Installed -- AUTO FXO (FCC mode)
How can I change this FCC mode to something else?
Soner
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
kind of problems would I face? Or
should this e-mail be sent to the developers' list?
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 02, 2005 5:39 PM
Subject
Sent: Wednesday, March 02, 2005 8:47 PM
Subject: Re: [Asterisk-Users] How to change fxo_mode
Just add this to /etc/modprobe.conf:
options wctdm opermode=TURKEY
Julian J. M.
On Wed, 2 Mar 2005 18:15:24 +0200, Soner Tari [EMAIL PROTECTED] wrote:
Sorry for littering the maillist, I've found it myself
on your
location.
Telco standards vary widely across this globe we call Earth.
Lyle
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 4:49 PM
Subject: [Asterisk
Oh btw, my version:
Asterisk CVS-HEAD-02/27/05-17:01:45 built by [EMAIL PROTECTED] on a i686 running
Linux
- Original Message -
From: Soner Tari [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, March 01, 2005
Hi all,
My * is connected to the same line as my ADSL, of course with an attached
splitter. The problem is that * cannot detect remote side hangups and
continues to service the Zap FXO channel. I tried ks, ls, and gs, without
success, and I think I've read most of the relevant documentation and
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