Welcome to the club, same here with freebsd :(
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Jarek
Jarzebowski
Envoyé : vendredi 10 février 2006 23:01
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Asterisk 1.2.x + oh323 on Debian
I'm the coordinator for the OpenH323 project
The Ekiga team (previously known as GnomeMeeting) maintain
Debian-compatible snapshots of openh323 and pwlib. See the GnomeMeeting
(http://www.gnomemeeting.org) download page for more information.
Failing that, What versions of openh323/pwlib did you
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if
On 10/02/06, Jordan Novak [EMAIL PROTECTED] wrote:
I am using Asterisk to send Voicemail out as Email. I am running into a
problem I believe to be caused by the exchange server requiring SMTP
authentication. I cannot get the sys admin's to turn it off. Does anyone
know enough about sendmail
Hello List and Armin,
I have been trying to narrow down my problems with getting chan_capi to
function properly. It seems that anything above build 8015 causes a
segfault on dial or receive.
The problem almost seems sporadic, and is certainly related to sip or
iax channels.
As soon as I update
It claims to have carrier-grade algorithms - don't glibly translate that to carrier grade hardware, it's a PCI card...RobOn 2/8/06, Matt
[EMAIL PROTECTED] wrote:try sangoma carrier grade 104d hardware EC card. we're using it ourself.
Best RegardsMatt- Original Message -From: Anthony
An easy way to do that, if you do not neet to register on a gkp, its
doing a dial OH323/ipgateway:port
Did you try this?
Abdul Lateef escribió:
Hi all,
I have H.323 Gateway, and i want to make a peer to
route calls to this GW. But i don't know is oh323.conf
supporting to add peer type entry
Appreciate the thought on the handsets - but these lines will be
going into an apartment complex - that is why faxing must work on any
line and it must be analog.
The astribank will not be a valid solution with the number of them I
would require.
-Jon
Hans Witvliet wrote:
On Thu,
Hi all
I am using Asterisk 1.2.2 on frdora core 4. i have two
sip UA. if i put canreinvite=yes voice Ok on both
sides. and if i change canreinvite=no there is no
voice (media through asterisk)
one thing more if i try to use playback application
for playing some sound file it is also working
I'm replying to this mainly to add my comments to the archive and then all
the webcrawlers...
I found a deprecated command curl which I though had simply been converted
from an app to a function, was actually completely non-working. Anytime my
call hit a exten = s,1,set(CURL=curl()), the channel
On Sat, Feb 11, 2006 at 08:29:31AM +, Peter Bowyer wrote:
On 10/02/06, Jordan Novak [EMAIL PROTECTED] wrote:
I am using Asterisk to send Voicemail out as Email. I am running into a
problem I believe to be caused by the exchange server requiring SMTP
authentication. I cannot get the
maybe usefull for displaying CALLED party name when dialing
I'm remember, that this feature was planned to add to asterisk, any
progress?
PJ
New and Changed Information
Release 8.0(0) includes the following new and enhanced features:
•Remote-Party ID support has been added for
Upgrade to 1.2.4 - bug in 1.2.2 - see www.asterisk.org front page.
-Jon
Kamran Ahmad wrote:
Hi all
I am using Asterisk 1.2.2 on frdora core 4. i have two
sip UA. if i put canreinvite=yes voice Ok on both
sides. and if i change canreinvite=no there is no
voice (media through asterisk)
one
ello,
Our company is looking for an Asterisk platform to
offer DID service to our existing clients. We are
located in Dammam Saudi Arabia. Our local client base
is 97% business from small stores to large businesses
and production companies. We also have a global client
base but are more targeting
On Sat, 11 Feb 2006, Rob Lith wrote:
It claims to have carrier-grade algorithms - don't glibly translate that
to carrier grade hardware, it's a PCI card...
What echo canceller hardware do you recommend for an asterisk PC?
-Dan
___
--Bandwidth and
Anyone managed to get Qwest to enable disconnect supervision on analogue
lines?
-Dan
___
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To UNSUBSCRIBE or update options visit:
Hello Gurus, here's my problem:
I downloaded and installed [EMAIL PROTECTED] (the version sporting Asterisk
1.2.4) and now I've got the following problem on the CLI output console
(Alt+F9): Most text is fine, except something that looks to me like
parameters. I don't really know how to explain
I've got a Mobile-to-PBX gateway installed and I want the ability to dial
from my mobile phone into my PBX and next dial a land-line from the PBX so I
can make cheep mobile-to-land-line calls while on the go.
I've contemplated using the WaitExten application but it only seems to wait
for ONE
I know this one must be easy but I'm an newbye so please help.
In my extensions.conf I want to have a line like:
Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)
Ie: I want to dial all the XXX-es, but not the 9;
How do I do that? What do I write in place of ${SOMETHING}? Navigating the
wiki
Exten = 9XX,1,Dial(Zap/4/${EXTEN:1},40,r)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Cosmin
Prund
Envoyé : samedi 11 février 2006 12:19
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Dialing part of the extension
I know
On Saturday 11 Feb 2006 00:10, Kevin P. Fleming wrote:
Warren Burstein wrote:
How about if it would set a global variable before each disk write so
the SIGFSZ handler would know which file caused it?
Ha!
Signals are asynchronous. This global variable would to be
lock-protected, would
Thanks, it works!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Olivier.taylor
Sent: Saturday, February 11, 2006 1:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE : [Asterisk-Users] Dialing part of
hi
i'm trying to configure a TE205P on [EMAIL PROTECTED]
i've edited /etc/sysconfig/zaptel adding this line:
MODULES=$MODULES wct2xxp
now, when the system is loading, i can see that the wct2xxp module is
loaded correctly
but if i try the command:
/usr/local/sbin/genzaptelconf
i get:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
I've been playing with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee
TE406P/411P and if you need to go dedicated to hanlde all possible look at an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO CANCELLER (1U Version) ± $1295
RobOn 2/11/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sat, 11 Feb 2006, Rob Lith wrote: It claims to have
Yes, right now we are only using span 1 on the quad span card with
plans to pull in another T1 PRI when we get this echo problem solved.
The echo is only experienced when the call terminates to traditional
analog circuits both local and long distance. Calls to cell phones,
and other known
The Orion echo canceller is just ok. The
Tellabs units work just as well if you dont mind 10 mins of soldering.
I have the orion running with an adit 600
and a TE110P. Echo cancel is fairly good, but I have loads of problems with
DTMF digits.
-Darren
From:
Can I configure the console to ring on one sound card and the headset on
another sound card?
Best regards,
Anthony.
___
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To UNSUBSCRIBE or update options visit:
Andres wrote:
There is no username in the above To header. Check your DIAL command
because something is wrong here. Thats why you get a 404. The SPA
can't match the username.
Yes. I had not reverted to an early enough commit on the configuration
files and the usernames were still missing
StaggI don't think it's a matter of trying echocancel on and off, it is a matter of tuning your system to your local PSTN - this is a combination of trying the different echocan alogrithims (i.e. MG2), the echotrainign etc and setting your txgain - too loud outgoing audio will result in echo.
Darren Wright wrote:
The Orion echo canceller is just ok. The Tellabs units work just as
well if you don’t mind 10 mins of soldering.
I have the orion running with an adit 600 and a TE110P. Echo cancel is
fairly good, but I have loads of problems with DTMF digits.
And, are usually found
On Sat, Feb 11, 2006 at 10:17:37AM +0500, ast guy wrote:
Hi,
I have downloaded iaxcomm version iaxcomm-lin-1.0rc3, when I try to
execute it it gives following error.
# ./iaxcomm
Error wxWindows Fatal Error : Couldn't Initialize IAX Client .
any idea what's going wrong ?
No. But this is
Bob Goddard wrote:
Using fopen/fputs to ONLY append to a file, is quite frankly, stupid.
Change it to open/write and you will be able to trap via the write
return code and errno.
Patches to fix bugs are most welcome. Given that these files are written
using fprintf (because they are using
Hi,
i treid this
OH323/ipgateway:port
and working well for me. But i need to add some more
featurres, like some of my H323 GW supporting only
G.7231 codec and some one G.729 and others feature
like rtptimeout etc
So if i am direct dialing without these feautres, the
GW are not able to
I thought that if the VPM was detected then you didn't have any control as to
which algorithm was used.
I was under the impression that the algorithms were only used for the software
echo cancellation.
At this point I'll give anything a try.
Stagg Shelton
www.oneringnetworks.com
On Feb 10, 2006, at 10:25 PM, Steve Underwood wrote:
Matthew Fredrickson wrote:
On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:
Found it, going to go test it right now :) thanks!
So far reports have been positive on the echo, but its a slow day ;)
We're using cisco 7960 phones,
Hi Sam Tam,
i would be interested in these ATA that you can offer.
please provide me with more details about this option.
thank you very much,
Mickey Lazar
On 2/9/06, Sam Tam [EMAIL PROTECTED] wrote:
We have got some ATA for only $55 if you are interested?Sam-Original Message-From:
Matthew Fredrickson wrote:
On Feb 10, 2006, at 10:25 PM, Steve Underwood wrote:
Matthew Fredrickson wrote:
On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:
Found it, going to go test it right now :) thanks!
So far reports have been positive on the echo, but its a slow day ;)
We're
The -biz list is more appropriate for this.
Tele Cost Price Reducer wrote:
Hi Sam Tam,
i would be interested in these ATA that you can offer.
please provide me with more details about this option.
thank you very much,
Mickey Lazar
On 2/9/06, *Sam Tam* [EMAIL PROTECTED]
Has anybody has issues with the new Native MOH and IAX trunking when
placing a call on hold?
My scenario,
Call is placed on a Definity G3 via PRI to Asterisk. Gets trunked over
to another Asterisk system via IAX2. Call is answered by operator and
placed on hold. At that point, audio is
Doug Lytle wrote:
Has anybody has issues with the new Native MOH and IAX trunking when
placing a call on hold?
Actually, I meant to say, the call is parked.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only way you would need authenticated SMTP is for relaying. My
suggestion would be to not set up sendmail to use a smart host but have
it act as an internet mail server. It will lookup the mx records and
make the sending determinations based on
At 12:51 AM 02/10/2006, you wrote:
-- Executing GotoIf(Zap/29-1, 1 0?4:3) in new stack
-- Goto (macro-stdexten,s-NOANSWER,4)
Should look like:
GotoIf( $[1 0]?4:3 )
Ira
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.375 / Virus Database:
At 07:08 AM 02/10/2006, you wrote:
I mean, can I write
the following two lines in only one line?
exten= 12345/100,1,Hangup
exten= 12345/200,1,Hangup
I would think this would work:
exten= 12345/[1-2]00,1,Hangup
Ira
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version:
Miguel,
On 2/11/06, Miguel [EMAIL PROTECTED] wrote:
pbx.c:1688 pbx_extension_helper: No application 'MeetMe' for extension
i did a normal make, make install, did i miss something?
You need zaptel headers installed to build MeetMe application. And you
need zaptel devices (or ztdummy) to run
At 07:08 AM 02/10/2006, you wrote:
I mean, can I write the following two lines in only one line?
exten= 12345/100,1,Hangup
exten= 12345/200,1,Hangup
Oops, forgot the underscore!
I would think this would work:
exten= _12345/[1-2]00,1,Hangup
Ira
--
No virus found in this outgoing
I just bought a new IAXy box and am only achieving one way calling.
Both iax.conf and the IAXy support ulaw and gsm. When I try to call,
however i get this error:
Feb 11 15:20:32 NOTICE[7963]: channel.c:1893 ast_read: Dropping
incompatible voice frame on IAX2/iaxy-2 of format ilbc since our
At 03:18 AM 02/11/2006, you wrote:
Exten = 9XX,1,Dial(Zap/4/${SOMETHING},40,r)
Ie: I want to dial all the XXX-es, but not the 9;
Exten = 9XX,1,Dial(Zap/4/${EXTEN:1},40,r)
Ira
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.1.375 / Virus Database:
Stagg Shelton wrote:
It was Digium's opinion that perhaps the card had a VPM. We got a
replacement TE411P, I implemented it tonight and still the exact same
echo problem. At this point I feel like I can rule out failed hardware.
I contacted Digium support and now they are telling me it's
It's part of ASTPP. It is in astpp -head ready for testing.
Darren Wiebe
[EMAIL PROTECTED]
Sam Tam wrote:
When will it be ready ?
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: Saturday, February 11, 2006 9:38 AM
To:
/etc/asterisk/musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing MusicOnHold(Zap/1-1, ) in new stack
-- Started music on hold, class
Hi folks,
Got an oddity that a user has raised to do with busy signalling and
inparticular when calling from a mobile phone. It seems that the
behaviour when calling * is slightly different to the norm i.e. If I
call an engaged landline number directly from my mobile then the mobile
gives
Johann Steinwendtner [EMAIL PROTECTED] writes:
I can only guess, but I think I can remember that the creflen needs
to be 2 octets for qsig. Check what the Alcatel switch sends in the
setup message to *.
Thanks, I will have a look at that.
Anyway, why do use QSIG ? Does name display work on
The genzaptelconf doesn't work with E1/T1 cards in my experience.
You will have to configure it by hand.
PsulH
- Original Message -
From: nik600 [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, February 11,
I can definitely vouch for Sangomas cards with hardware echo
cancel. Ive been installing Asterisk boxes for about 6
months now using Digium TDM cards and Sipura SPA-3000s in
small installations. This past month I installed in a small
office with 3 pots lines. The echo was very bad and of
On 13:44, Sat 11 Feb 06, Sean Cook wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The only way you would need authenticated SMTP is for relaying. My
suggestion would be to not set up sendmail to use a smart host but have
it act as an internet mail server. It will lookup the mx
FYI,
If you want to learn more about why ${EXTEN:1} works, check out the
Asterisk TFOT book, chapters 4 and 5. Page 95 of chapter 5 deals
specifically with the ${EXTEN} variable and the syntax of adding :1
(or :2, :3, etc.) - good stuff to know.
Check it out:
Perhaps there's a happy medium: sprintf()?
I am curious to know if putting the output into a char array with
sprintf() (to preserve the output formatting) and then writing it with
write(). How much additional overhead would this take? Hard to know
without trying it.
Is anyone in a position to
Tim Connolly wrote:
/etc/asterisk/musiconhold.conf:
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing MusicOnHold(Zap/1-1, ) in new stack
-- Started
On Sat, Feb 11, 2006 at 12:30:51PM +0200, Tzafrir Cohen wrote:
On Sat, Feb 11, 2006 at 08:29:31AM +, Peter Bowyer wrote:
Install MSMTP as your local MTA (replacing sendmail). Configure
Asterisk to use the local MTA, and configure MSMTP to forward to the
Exchange server with
I vouch for Sangoma's too.
We use sangoma 104d EC card, echoes gone, works well. As to te411p, we have
not tried yet, we don't know.
Best Regards
Matt
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, February 11, 2006 2:53 PM
Subject:
On Sat, Feb 11, 2006 at 03:26:26PM -0800, Michael Collins wrote:
FYI,
If you want to learn more about why ${EXTEN:1} works, check out the
Asterisk TFOT book, chapters 4 and 5. Page 95 of chapter 5 deals
specifically with the ${EXTEN} variable and the syntax of adding :1
(or :2, :3, etc.) -
On 16:48, Sat 11 Feb 06, Matt wrote:
I vouch for Sangoma's too.
We use sangoma 104d EC card, echoes gone, works well.
I second that, the sangoma cards are awesome.
--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key:
Michael Collins wrote:
I am curious to know if putting the output into a char array with
sprintf() (to preserve the output formatting) and then writing it with
write(). How much additional overhead would this take? Hard to know
without trying it.
Please... if you don't have the skills to
Hi
Not exactly a asterisk specific question and what more I'm a newby. I
apologize.
The story: I was given the task of transitioning my company's PBX (20
people) from a normal old digital PBX to something newer. I chose to use
Asterisk. For the project i was given a Dell 850 for the task. My
Hi!
I am playing around with Asterisk and have a problem :-)
(Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4)
I have a sip-phone at my desk and an ISDN-phone (independent of the
Asterisk-server) in my living room, when I'm not at my desk, the
sip-phone is switched off. I would like to be
Try build 8015. I know its odd, but this is just like the problem I am
having...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Florian
Heer
Sent: Saturday, February 11, 2006 9:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
[EMAIL PROTECTED] wrote:
Try build 8015. I know its odd, but this is just like the problem I am
having...
Uhm... sorry if I seem a bit uninformed, but how do I get that version?
Regards, Florian.
___
--Bandwidth and Colocation provided by
Iam trying to run the latest stable Asterix version 1.2.4. on 64 bit amd procesor.
Things are working but the playback sounds that I hear when tring to connect over IAX are of very high frequency.
i.e a sentece which shoudl finish in 4 secs finishes in much lesser time. Where can be the problem?
On Sat, 11 Feb 2006, Rob Lith wrote:
TE406P/411P and if you need to go dedicated to hanlde all possible look at
an external dedicated canceller like www.oriontelecom.com VCL-E1 ECHO
CANCELLER (1U Version) ? $1295
Is the orion echo canceller a higher quality EC than tellabs?
Could be, but I don't see why one would spend more just becuase the
answer is yes. The tellabs will do the job perfectly (at least in my
experience) and can be picked up for less than $100.00 on eBay. They
have proven to last for 20 years (the older models being sold on eBay
were manufactured in
I don't know about the Tellabs cancellers in particular, but I think any
echo canceller built in the 80s will be a fairly poor performer. Many
improvements in EC occurred in the early 90s, in response to the
problems of earlier cancellers. Also, most older cancellers only cancel
fairly short
Hi everybody,
I have an Asterisk box and I want to install just ARI on it for
monitoring the calls. Installing [EMAIL PROTECTED] utilizes too much resources
and
memory and also takes away freedom of configuration asterisk. I like
using asterisk on its CLI. But just for recorded calls I need to
The most common Tellabs EC are not the ones from the 80s, I was just
pointing out that the quality was good, since they still last now 20
years later.
On 2/11/06, Steve Underwood [EMAIL PROTECTED] wrote:
I don't know about the Tellabs cancellers in particular, but I think any
echo canceller
On Sun, 12 Feb 2006, Steve Underwood wrote:
I don't know about the Tellabs cancellers in particular, but I think any echo
canceller built in the 80s will be a fairly poor performer. Many improvements
in EC occurred in the early 90s, in response to the problems of earlier
cancellers. Also, most
Eh. Not for $1000 more, and I've got both in production. Customer service
was an issue.
-Darren
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Saturday, February 11, 2006 10:19 PM
To: Asterisk Users
On 2/12/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Sun, 12 Feb 2006, Steve Underwood wrote:
I don't know about the Tellabs cancellers in particular, but I think any
echo
canceller built in the 80s will be a fairly poor performer. Many
improvements
in EC occurred in the early
Darren, how was customer service an issue? I mean once you got one to
work, it just plug and forget.
On 2/12/06, Darren Wright [EMAIL PROTECTED] wrote:
Eh. Not for $1000 more, and I've got both in production. Customer service
was an issue.
-Darren
-Original Message-
From:
WELL!
The Orion guys agreed to send me one as a demo for 30 days. I'm doing 1
install / week now, so it was a good business opportunity for them. I had
issues with DTMF during the test phase, and the tech guys were not terribly
helpful. 3 weeks into the test (a week early) collections calls
I just bought a new IAXy box and am only achieving one way calling.
Both iax.conf and the IAXy support ulaw and gsm. When I try to call,
Does the IAXy now support anything but ulaw or alaw? The original one didn't.
___
--Bandwidth and Colocation
Kevin,
I agree with your assessment of the preference of using fprintf()
instead of sprintf() + write() + maybe malloc(). After hearing your
candid explanation it makes perfect sense not to pursue this. Ive
only been playing with * for two months, so Im still gathering my
bearings. As
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