a userbname necessary ? And if how can i dial
so?
Can somebody help please ?
Thanks,
Thomas.
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Potsdamer Str. 18 A
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Please, can somebody tell me how do a h323 call correctly with the dial app
?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Montag, 22. September 2003 18:26
An: Asterisk User
Betreff: [Asterisk-Users] how to dial a h323
or similar like this ?
Thanks for help,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 11:15
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination
Here is my oh323.conf ...
;
; Configuration file of OpenH323 channel driver
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is
to dial a h323 destination ?
Try to add gwprefix in oh323.conf after your alias. You must know that
you can configure * gw in gnugk.ini or in oh323.conf. I recommend you
put in your oh323.conf.
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Hi all,
can somebody explain this ?
Thanks,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
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14513 Teltow
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-23 at 09:55, Thomas Haeger wrote:
Hi all,
can somebody explain this ?
Do you have something like a |15 in the dial string?
Do you have logs to show what asterisk did?
--
Steven Critchfield [EMAIL PROTECTED]
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I have tried it with a timeout and without...
here the * output for the first side:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1,
IAX2/useranme:[EMAIL PROTECTED]/99033283077731) in new stack
-- Called thaeger:[EMAIL PROTECTED]/99033283077731
-- Call accepted by
I have tried it with a timeout and without...
here the * output for the first side:
-- Starting simple switch on 'Zap/3-1'
-- Executing Dial(Zap/3-1,
IAX2/useranme:[EMAIL PROTECTED]/99033283077731) in new stack
-- Called thaeger:[EMAIL PROTECTED]/99033283077731
-- Call accepted by
-23 at 09:55, Thomas Haeger wrote:
Hi all,
can somebody explain this ?
Do you have something like a |15 in the dial string?
Do you have logs to show what asterisk did?
--
Steven Critchfield [EMAIL PROTECTED]
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Hi Jim,
i had the same probs, and it seems to be bug/feature of i4l. I can not find
anything in the code that would bring these messages to the top of
ttyI:-(
Or is there somebody who knows it better ??? ;-)
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto
Hi,
Given the following setup
Cisco (7960, G729) -- Asterisk -- IAX2(SPEEX)
It seems that asterisk cannot do the conversion (calls are rejected) even if i have G729 licenses
Is there a parameter in IAX.conf that allows this conversion.
PS
if the call originates from FXS, i.e.
it be a bug on the other side (terminator switch) ?
Have anyone an idea ?
Thanks,
Thomas.
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Hi,
what the hell is this ?
Can somebody cancel this user from the list ???
Thanks,
Thomas.
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beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
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FAX:+49 (0) 3328 334779
Email
Hi all,
gave somebody an idea ?
I have not set a AbsoluteTimeout or smothing like this.
Regards,
Thomas.
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Hi,
On the same note, I am having a problem with G.729, having 4 * asterisk boxes 2 with 10 licenses and one with 2 licenses.
The licensesinstalls fine, but the codec doesn't work as supposed to be. In path thru situation, where a UA (grandstream phone) is talking to the * that is connected to
I totally agree with you. The codec is buggy and the license agreement from VoiceAge is - to put it in proper way- preposterous. However, I have to find a solution for customers with Cisco (79xx and budgetone) that don't want to use up all their network bandwidth. Until someone implements speex of
We had the same problem
I found it useful to turn down the rxgain and txgain to -14 on those channels
and make the T1 card the master clock source
zaptel.conf
span=1,1,0,esf,b8zs
#for the T100P
--- Note the 1 instead of 2 for the second parameter.
span=1,1,0,esf,b8zs
# For the T100PSteven
DEBUG[1116941120]: File chan_iax.c, Line 3553 (socket_read): Ooh, voice format changed to G729ADEBUG[1116941120]: File chan_iax.c, Line 3864 (socket_read): We don't do requested format G729A, falling back to peer capability 256NOTICE[1116941120]: File chan_iax.c, Line 3867 (socket_read):
a call from *1 over *2 to PSTN, i can hear an echo in my
analog phone,
even though echocancel and echocancelwhenbridged is on yes on both
sides.
Can somebody explain me what i'am doing wrong ?
Thanks,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing
loader.c, Line 347 (load_modules): Loading module
chan_modem.so failed!
Thanks for help,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email: [EMAIL
[contr1/8504]/22 (macro-stdexten s3 ) Up Dial
SIP/snom1|20|mt
2 active channel(s)
gw-bzo*CLI
Is this complete?
Thomas
mgcp.conf:
[general]
port = 2727
bindaddr = 0.0.0.0
disallow=all
allow=alaw
inbanddtmf=0
transfer = yes
threewaycalling=yes
musiconhold=1
-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109
loadzone = fr
defaultzone=fr
Thanks for your help.
Regards,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328
running ?
Best regards,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Scott
Stingel
Gesendet: Dienstag, 14. Oktober 2003 02:28
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] PRI/E1: machine freeze/dies after a few
calls
Hi Thomas
i guess it is a matter of
naming conventions.
Has someone an idea?
Thank you in advance,
Thomas Wienecke
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Am Freitag, 17. Oktober 2003 19:51 schrieb Lal, Deepak (Contractor):
i am willing to assist also.
mostly on weekends, i m afraid, but willing.
Thomas W.
Count me in too.
-Original Message-
From: sip [mailto:[EMAIL PROTECTED]
Sent: Friday, October 17, 2003 1:56 PM
To: [EMAIL
with this ?
Thank you very much.
Thomas.
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Here with a snom200/SIP and ATA-186/MGCP everything works fine
(i dial *8 to pick up a call).
-Thomas
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Have you another ISDN card in your system ?
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von C M
Gesendet: Donnerstag, 23. Oktober 2003 14:06
An: [EMAIL PROTECTED]
Betreff: [Asterisk-Users] wcfxs error
hi guys, i got a TDM400P FXS card an everything
Hi all (Michael),
how it is possible to get the ip address of the calling party ?
(i know by using h323... but there're a few unknown segfaults...) and so i
want to use oh323, but i have to get the ip from the caller to permit or
deny the call with AGI.
Is it possible at all ?
Thanks,
Thomas
the wcusb driver on a VIA EPIA 5000
machine.
Any ideas ?
Thanks,
Thomas.
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Ok,
meanwhile i loaded the tor2 driver with insmod (previous loaded slhc,
ppp_generic,zaptel)
this works, but when i execute ztcfg the machine is freezed!
Now any ideas ?
Thanks,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
out the span,bchan and dchan values and then ztcfg works.
But this is naturally not wished ;-)
Any ideas (agian...) ?
Regards,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Mittwoch, 29. Oktober 2003 14:05
with loading tor2 and wcusb
did you check that there are no irq conflicts ?
On Wednesday 29 October 2003 3:13 pm, Thomas Haeger wrote:
Hi i'am again,
here my zapata.conf:
span=1,0,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
Hi all,
can somebody tell me where i can get the g.723 codec for * ?
Thanks.
Regards,
Thomas.
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, 2003-11-03 at 14:28, Thomas Haeger wrote:
Hi all,
can somebody tell me where i can get the g.723 codec for * ?
http://store.yahoo.com/asteriskpbx/asteriskg729.html
$10 per channel. I looked into the licensing costs for another product,
and this is damn cheap.
Cheers,
Gavin
Thanks Steve,
there is no special reason for me for using g.723.
I will take g.729. It seems to be easier :-)
Regards,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Steve
Underwood
Gesendet: Montag, 3. November 2003 17:14
An: [EMAIL
there is a codec
g729a listed also the g729b is not installed.
what is the difference between g729a built in * and the puchased g729b
codec?
Thanks for help.
Regards,
Thomas.
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of an a
or a b).
I thought this codec could take calls with g729.a codec but this seems not
to be so.
If my fiction is right, how can i take calls with g.729.a codec ?
Thanks,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet
to ...
fxoks=1
fxsks=2
... if I changed to kewel-start in zapata.conf ?
I assumed so, and went ahead and did so. Still no dial-tone though.
Thanks,
Thomas
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== Manager registered action ZapDNDon
== Manager registered action ZapDNDoff
== Manager registered action ZapShowChannels
Thanks again!
Thomas
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On Sun, Nov 14, 2004 at 03:16:13PM +0100, Soren Rathje wrote:
Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe
the correct pair to use should be 2+3.
It's the middle pair. I assume that's 2+3 on an RJ connector ?
___
= chan_oss.so
So I assume now that it's not capable of making dialtone ?
Regards,
Thomas
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On Sun, Nov 14, 2004 at 03:35:06PM +0100, Soren Rathje wrote:
Just for verification, do you have any green led's lit on the back of your
card ??
Yes, and I have tested with a different telephone and cable that I know
works.
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Asterisk-Users
reset.
Sorry to be so dumb, but how would I do that ? I only have one FXS
module. Or is it possible to simulate a call from the *CLI console ?
Thanks,
Thomas
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hook state to 0 (00)
kernel: Setting FXS hook state to 0 (00)
I don't like the look of that NO BATTERY message. What do you think
Soren ?
-Thomas
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On Sun, Nov 14, 2004 at 04:36:21PM +0100, Soren Rathje wrote:
NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not
receive power from the line, i.e. it is not plugged into the wall socket.
(if I read the source correctly)
ok. I connected it to the PABX and I got this so I
on either linux or windows I'm happy
to try. What would you suggest as the easiest softpone to install ?
Thanks,
Thomas
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are ok.
I'm going to get hold of the supplier and see if he can test the module
for me.
Thanks so much for your help Soren. I have *really* appreciated it!
Regards,
Thomas
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On Mon, Nov 15, 2004 at 07:32:29AM -0500, Jason p wrote:
check to make sure you have a ip address added to teh skinny.conf
file.. if your even using skinny.
Yup, that's it. Thanks Jason.
Regards,
Thomas
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I found. I just put the IP of the ethernet card in
there and the error went away.
Regards,
Thomas
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Asterisk to call a script that uses a SMB winpopup (or other
method) out to a specified computer sitting next to the phone?
Thanks very much for any ideas, or knowledge of something already in
existence.
Thomas Hutton
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marks), and messes up the echo.
You can set this up per extension, of course, naming the file differently per extension to avoid any problems... also it might be smart to use a different working directory. Just don't name your scratch file something really dumb like extensions.conf.
Thomas Hutton
Hi Duane,
You asked Why dump to a file? - I don't know if this is possible or
not, but can you send a ctrld to the smbclient -M command? I believe
the way you wrote the command it will just hang, no?
Thomas Hutton
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firefox browser with evolution. (probably
an even greater sin than top posting)
Thomas Hutton
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hi all.
ive got a problem implementing my own small office asterisk solution.
i want to use
- a hfc-card via mISDN in NT-mode to serve my siemens gigaset 3035 isdn
phone
- an avm b1 to connect to pstn
- sip, iax etc.
working:
- chan_capi via the b1 works fine, i can dial in and get the demo
-
hi christiaan.
Christiaan Brink schrieb:
I'm currently busy on a similar application with a hfc-card. However, my
needs is to interface the ISDN card in NE-mode with the operator. If tried
using Hisax but ran into a problem with the voice quality being bad in one
directions.
How did you manage
hi list.
after my unsuccessfull experiences with mISDN i tried again to implement
a zaphfc based solution.
problem is: sound on calls via capi is stuttering/broken and therefore
unuseable.
my conf:
- cel 1300, 256mb ram
- avm b1 via capi connected to my outgoing ISDN
- acer surf pci via zaphafc
Martin List-Petersen schrieb:
On lør, 2004-11-20 at 14:55, Tim Robinson wrote:
Hi
I recommend you abandon the old card with the CAPI drivers and purchase
a second HFC card.
Not sure if I would call that card old or think it's the cause for the
stutter. The B1 is a active card. That
Thomas Jagoditsch schrieb:
next step i will try what tim recommends - its cheap hardware vs.
expensive time anyway ;-)
i just have to wait for the second card.
hi all and thanks for your input so far.
second card arrived today and after all the important work i had the
urgent need to try
Tim Robinson schrieb:
Please post your zapata.conf and your zaptel.conf - not easy to advise
with out this info.
Ihave to confess to abandoning 2.6 kernels in favour of the older 2.4
kernels. that was then, maybe it is better now
btw. if you run ztcfg more than once you will have
hi stefan.
Stefan Märkle schrieb:
For me this doesn't look zt-related , do you have a pty/console/permissions
problem?
Try starting ztcfg with output sent to nirvana or to a file e.g.
ztcfg /dev/null 21
I wonder whether the segfault happens once more.
thx for you input.
unfortunatly its
Thomas Jagoditsch schrieb:
yeah, would be my next try to use 2.4. . no other idea in sight at
this time.
thx all for your help, got it running with kernel 2.4.27 now :-)
had some troubles identifiying which of the two hfc cards was the
internal/external but after some hours i found out. seems
notice is that one of the endponts registers itself on port 1025
as opposed to 4569 for some strange reason. I don't see any errors
about binding to port 4569, so I'm wondering what's the deal. Any
ideas? Thanks very much in advance!
Thomas Hutton
Hello-
I've got some audio CDs that I'd like to use for MOH.
What's the best way to do this? I don't care if it's mp3 or some other
format - whatever will work best.
What applications (osx or linux) are best? Optimal settings?
Thanks-
Tom
___
Can somebody suggest theeasiestway to only allow outgoing long distance calls to countries x, y, and z?
Since Broadvoice allows free long distance to a bunch of countries I would like to take advantage of that, but block all other long distance calls.
Thanks,
Tom
Do you Yahoo!?
Yahoo! Mail -
I'm using Directory, but it would be nice if, after playing a mailboxes
recorded name, it repeated the extension as well.
For example, somebody enters the directory, and types the first three
letters of the last name: BRO
The directory app would play the recorded name Charlie Brown, and then
Hi Folks
Since updated to 1.0.1/2 I got a prob with the hotkey to
access voicemailmain.
According to the wiki
0 jumps to extension oand* to a
0 isn't working, I get vm-sorry followed by HangUp :(
* is working and I get access.
So I changed the dialplan to get my voicemail managed.
On Sun, Dec 19, 2004 at 12:21:28AM -0600, Matthew Boehm wrote:
I'm having a similar problem. Do you have operator=yes in your
voicemail.conf under [general]?
Argh, thats it, solved!
Thanks a lot :)
...cut
--
Tho/\/\as
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Hallo Jakob Strebel
On Mon, 22 Mar 2004 22:56:09 +0100 you wrote:
Thomas,
I restarted the hack and found that I did not edit all the files in
src.drv. My mistake sorry.
No prob.
...cut
So I fixed this, changed all _fcpci_ to _f2pci_. Now I am running into a
new problems.
IF I
Hallo Jakob Strebel
On Tue, 23 Mar 2004 09:07:20 +0100 you wrote:
Thomas,
Thank you for your help. Which Kernel version do you run?
I have debian 2.4.24, may be I could use your patched object code?
I wrote: I do not use this
I use the b1.
Could you please send it to me if it makes sense
Hallo Ignace CARIA
On Tue, 23 Mar 2004 09:35:27 +0100 you wrote:
Hello everybody,
I try to connect directly to Asterisk an ISDN DECT base station.
Here is the scheme:
ISDN Line--ISDN CARD(CAPI)--+Asterisk+--ISDN
CARD(???)-DECT Base
jc wrote:
I have a simple * setup with a couple of SNOM200 installed. I can make
IP calls and internal calls fine. But, I cant hear any of the
asterisk sound files on playback. Any ideas
are there any errors on the CLI? I had that problem too but it did
actually throw out errors on the
Hi,
I just got started with Asterisk. Installation was OK, no errors.
But how do I activate the IAX and SIP channels now? I loaded the
modules, but nothing happened, there's no connection to the
relavant ports.
Anything I forgot to do...?
Thanks in advance!
thomas
---
Thomas Schroeter
))
== Registered application 'SIPDtmfMode'
But I cannot connect to port 5060. Why not???
Regards,
thomas
PS: I xed the IP.
---
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)
Thank you
Thomas
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that most of you will tell me that Asterisk is better but I
would really like to know why ...
Thank you in advance.
Thomas
signature.asc
Description: This is a digitally signed message part
to handle the load and be stable ...
Any testimonial of people having implemented that setting and being
happy with it would be greatly appreciated.
Regards,
Thomas Mangin
signature.asc
Description: This is a digitally signed message part
Scott.
Please let us know if you are getting an answer off-list or if you
figure it out as I may experience the same issue soon, and I would
appreciate the information
Thank you in advance.
Thomas
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/thomas, my extension is 1505
[thomas]
type=friend
host=dynamic
dtmfmode=inband
; your dtmf mode may be right for your phone ... No idea.
username=thomas
secret=supersecret
callerid=Thomas Mangin 1505
context=default
mailbox=1505
;auth=md5
;reinvite=no
;canreinvite=no
;qualify=1000
;defaultip
The terminating telco is doing an SCP dip to thier local SCP's and the
database probably does not have that name mapped to this number.
First thing to do is make sure the generic name ISUP optional paramter is
set in the outgoing IAM / ISDN setup from your GW.
You could also store with an SS7
SCP=Service control point (database that houses name to number)
SCP DIP = Query to an SCP via the SS7 network
ISUP = SS7 signaling for call setup and teardown (equivalent of
invite,ringing,ok,bye)
IAM = Initial address message (equal to the SIP invite )
LNP= Local number portability (uses the SS7
conf files if
possible. This is pretty new stuff for me... Thanks again!
--
Ryan
On Apr 6, 2004, at 7:59 PM, Kyle Thomas wrote:
SCP=Service control point (database that houses name to number)
SCP DIP = Query to an SCP via the SS7 network
ISUP = SS7 signaling for call setup
Jain, Sonal wrote:
Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment.
Thanks,
Entering reload in the console should do if you edit the extensions.conf
and some other files.
Stephen Karrington wrote:
I can't read German. Can you outline the cost for me? Thanks.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
Fax -
Paul Tyreman wrote:
Does anyone know if you can put two of the X100P cards in to the same
machine and have access to two landlines ?
I just need to know if it's worth buying two or not !
Thanks, Paul.
I run 4 X100P's in our asterisk box. Just make sure you give each card
it's own IRQ.
San Singhania wrote:
Hello,
I have just managed to get my 1st * server up and running and have a
lot of issues with theX100P analog card. Would really appreciate
anyone trying
to help me on the following :
1. The receive and transmit is too soft. So i increased the txgain and
rxgain. The
.
Is this a known problem...??
Regards,
thomas
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Tel./ Fax: 0700 - 22 55 22 66 // www.rapetho.com
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Matt Riddell wrote:
Are you using the more than one manager application? I.E. op_panel etc...
Reason I ask, is that I had two copies of op_panel running and lost a couple
of calls, but with just one running things have ben fine...
Matt Riddell
- Original Message -
From: [EMAIL
Eric Wieling wrote:
[EMAIL PROTECTED] wrote:
Yes turning off echo cancelling would be fatal. We have some serious
echo going on here that I can not seem to track done. I am assuming
it is just this old building. Maybe we can go ISDN or so but the
dropped calls are rather bad.
Rnadom
Matt Riddell wrote:
- Original Message -
From: Thomas Gallaway
-big snip -
| I am just running 1 instance of the op_panel. But today I noticed that 2
| calls got ended after
| 2 minutes 34 seconds. I will disable the management thing just for testes
There is no IE ( information element) in the isdn setup for this
indicator. Of course with ISUP(SS7) FGD trunks it is delivered in the OLI ISUP
parameter
On Tue, 20 Apr 2004, James Sharp wrote:
Quickie: Does anyone out there have experience with PRI delivery of ANI II
information?
Hallo Altus Snyman
On Wed, 21 Apr 2004 09:54:42 +0200 you wrote:
Do you have a copy for me,the page seems to be closed and it redirect me
to http://swpat.ffii.org/ and I cant read that
Thanks
Try this:
http://www.linphone.org/linphone.php?lang=usrubrique=1
On Wed, 2004-04-21 at 09:16,
is available to answer at this time
There is one exception: One specific number is dialled, and in the client I hear the
real outside dialtone, but it is definitely not the number I wanted to dial! So it
dials
ONE number, but the wrong one...?!?
Who can help...?
Regards,
Thomas
into a nother PCI slot or playing arround with the
IRQ's?
-- Thomas
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Zach Chambers wrote:
Ever tried to use the latest fedora kernel. Also what type of chipset
do you have? Might be an chipset incompatiblity. Did you ever try
swapping arround the pci card into a nother PCI slot or playing
arround with the IRQ's?
-- Thomas
Thomas, I could not get the zaptel
Thanks for your help. If there are any changes I should do just let me
know. I can just call t-mobile
customer support and see if I get disconnected.
-- Thomas
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[EMAIL PROTECTED] wrote:
lovely, :-)
Is it just me or where there allready 3 virus sent to this list today?
Maybe time for denim to disallow attachments? :-)
-- Thomas
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Hi,
ztdummy says the following:
VoiceBOX:/usr/src/zaptel# modprobe ztdummy
/lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_unregister
/lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_transmit
/lib/modules/2.4.18/misc/ztdummy.o: unresolved symbol zt_receive
I have the uhci_usb modules etc. installed.
Make sure this is a module and *not* part of the Kernel.
I have usb-uhci and usbcore as modules. What about PPP support?
Is that a problem? Should I also install it as a module?
regards,
thomas
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