-users] problems with hylafax + iaxmodem
+asterisk1.8.5
On 5/09/2011 10:05 PM, Alessio wrote:
someone can help me to solve this problem?
thanks
--
From: Alessio ales...@asistar.it
Sent: Friday, September 02, 2011 5:10 PM
To: Lee Howard fax
someone can help me to solve this problem?
thanks
--
From: Alessio ales...@asistar.it
Sent: Friday, September 02, 2011 5:10 PM
To: Lee Howard fax...@howardsilvan.com
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +
asterisk1.8.5
Alessio wrote:
I have 2 computers in the lan, one is the Asterisk PBX and the other is
the server with hylafax and iaxmodem installed.
.
Sep 1 16:50:11 FAXServer FaxGetty[6225]: -- [4:RING
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] problems with hylafax + iaxmodem +
asterisk1.8.5
Alessio wrote:
I have 2 computers in the lan, one is the Asterisk PBX and the other is
the server with hylafax and iaxmodem installed.
.
Sep 1 16:50:11 FAXServer FaxGetty[6225
'SIP/0465940394-0002' status is
'CHANUNAVAIL'
--
From: Lee Howard fax...@howardsilvan.com
Sent: Friday, September 02, 2011 4:33 PM
To: Alessio ales...@asistar.it
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Hi!
I recently upgraded Asterisk from version 1.6.2 to 1.8.5
Now about every 10 minutes all SIP TRUNKS becomes UNRECHABLE for a few seconds
or minutes after become LAGGED and later become OK.
I have no idea of the cause of this problem.
With the version 1.6.2 all runs perfectly.
I can't say
Hi!
from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5.
I have 2 computers in the lan, one is the Asterisk PBX and the other is the
server with hylafax and iaxmodem installed.
In Asterisk I set up an IAX trunk in this way:
___
iax.conf
Hi!
I need help regarding the following problem:
when I receive a phone call to the PBX from the number 01234567890
rings the number 100, get up the phone, I transfer (assisted) to the number 100.
When the 100 number rings, the display shows the number of those who
transferred the call and not
the normal behavior of assisted transfers. Try a blind/non-assisted
transfer, that should show the original callerid.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 2:56 AM
To: Asterisk Users
So I can't do anything?
--
From: Kevin P. Fleming kpflem...@digium.com
Sent: Friday, July 29, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] call forwarding number from outside.
On 07/29/2011 10:41 AM, Danny
Subject: Re: [asterisk-users] call forwarding number from outside.
Upgrade to 1.8/10.0
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alessio
Sent: Friday, July 29, 2011 10:04 AM
To: asterisk-users
Hi!
I'm using ael language and I need to pick up a call from outside to an internal
number.
for example:
i'm 120
the phone 100 rings, it's a call from outside.
now I pick up the call with: *8100
and I would expect to answer the call but the response is Declined
the Puckup code is below:
_*8X!
I think I have solved with the following code:
_*8X! = {
PickUpChan(SIP/${EXTEN:2});
Hangup();
}
thanks
From: Alessio
Sent: Friday, July 22, 2011 11:27 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Pickup(${EXTEN:2}); not works from outside
Hi!
I'm using ael
the first usable version.
So, in the end, my opinion is that is just a matter of time.
Hope it helps, have a nice Christmas everyone!
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I migliori saluti,Scrivi a:
Alessio[EMAIL PROTECTED
migliori saluti, Scrivi a:
Alessio [EMAIL PROTECTED]
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,Alessio
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http://www.beronet.com/index.php?option=com_remositoryItemid=38func=selectfoldercat=1lang=en
just untar it and do make install, it will download and compile all
needed files for misdn - chan_misdn.
If you need further assistance you can contact me (in italian also) al
alessio AT interconnessioni
. (hint: a receptionist is first member of a queue and another person is the second ... receptionist goes for a pee and magically calls are rerouted to the backup operator after ringing to the first).
Hope you can find out something to share, maybe we can also launch a count us initiative :)Alessio
, Alessio.
~~Aaron
- Original Message -
From:
Alessio
Focardi
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Cc:
[EMAIL PROTECTED]
Sent: Thursday, June 29, 2006 1:31
PM
Subject: Re: [Asterisk-Users] Call Queue
NOT using RoundRobin ?!?
Welcome
Hi folks!Based upon your experience on the field what wifi sip phone would youreccomend ?A customer asked for a wireless * install and I'm looking for advice, tnxAlessio Focardi[[*] - Interconnessioni Italy
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Alessiomailto:[EMAIL PROTECTED
sound
CP interruptions) and to top it all it detects fake DTMF's all the time.
Try this settings for echo cancel: in my setup they work wery well
(most of the times)
[g1]
echocancel=256
echotraining=no
jitterbuffer=4000
jitterbuffer_upper_threshold=0
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Alessio
On 5/2/06, Gidean Chan [EMAIL PROTECTED] wrote:
Can anyone tell me how to make it
work?
I have asterisk 1.10.006 and hylafax in
the same linux server.
2 x100p on PCI slots connected with 2
PSTN lines.In my opinion you have two options:1) setup iaxmodem for hylafax and use asterisk as pbx
be absolutely great variable=0 led off, 1 led on, 2 led blink ... Alessio Focardi
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On 4/4/06, Pimjai Wesnarat [EMAIL PROTECTED] wrote:
I'm able to recieve fax with pure SpanDSP 0.0.2 + Asterisk successfullybut I have problems with some fax machine so I wanted to try usingHylaFax to recieve Fax instead of SpanDSP hoping that it'll solve my
problem. I'm trying to connect Asterisk
, or possibly dropping frames.You might want to check with Digium support to verify
Let me know the result.Cheers!RemcoOn Tue, 4 Apr 2006, Alessio Focardi wrote: Hi, I have an asterisk installation with 2 E1 cards Software version is
Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax
Hi,I have an asterisk installation with 2 E1 cardsSoftware version isAsterisk 1.2.6Libpri 1.2.2Zaptel 1.2.5I'm having problem with fax transmission, let me explain better mysetup:
My fist TE110P E1 card is connected to the telco linethe second TE110P E1 one to an Nexspan PBXso the server is
about that since it may seem rude.
Give misdn a test, it works better every day!
Saluti, mandami un messaggio privato ad alessio AT interconnessioni PUNTO it se vuoi continuare la discussione.
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Alessio[EMAIL PROTECTED
: cable disconnected or
no line) results in congestion for ${DIALSTATUS}, but message is too
generic for my use.
Any suggestion will be greatly appreciated, tnx!
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Alessio mailto:[EMAIL PROTECTED
server
while having a conversation between 2 phones: call should stay up.
Hope it helps!
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Alessiomailto:[EMAIL PROTECTED]
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where in the webradio dir there was just a dummy mp3 file
I would like to reproduce this using native mp3 ... any idea ?
Tnx !
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Hello richard,
Wednesday, November 23, 2005, 4:54:54 PM, you wrote:
rC Alessio, Sergio
So an upgrade is of course necessary.
rC i have upgraded the vigor. Bad news... i am not able
rC to register the draytek anymore. But using a XLite on
rC my pc behind the Vigor works now fine (no one way
and it was working, have you
put the private ip address of the asterisk box in the vigor setup ?
Can you ping the private address of the vigor from the asterisk box
and viceversa ?
Hope it helps !
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Alessiomailto:[EMAIL PROTECTED
Hello richard,
Wednesday, November 23, 2005, 12:34:33 PM, you wrote:
rC Hi Alessio
rC [SIPphone]--[Asterisk]--[Firewall]---[VPN]---[DrayTek]--[Analog-phone]
I tried a similar setup some times ago and it was
working, have you
put the private ip address of the asterisk box in
the vigor
Congratulations from Italy now back to work for 1.3 ! :)
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(in xlite is named transmit silence).
Hope it helps!
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? :)
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to create the relative extensions in the incoming
context ... just s will not work anymore.
Hope it helps!
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Alessiomailto:[EMAIL PROTECTED]
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,
Alessio mailto:[EMAIL PROTECTED]
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Hi,
is there an Agentlogout procedure opposite of the one we get with Agentlogin ?
I tried simply having another agent log from the same extension, but when I try
Show agents
10 (Alessio) available at '[EMAIL PROTECTED]' (musiconhold is
'default')
51 (Giuliano) available
!
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The original poster's statement about not even receiving any
s proof thathe was certified is kind of amazing.
s I wouldn't be too upset about it either because it is probably
s anhonest mistake, but I would be firm on demanding that you get
s what youpaid for.
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Alessio
,
Alessio mailto:[EMAIL PROTECTED]
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CVS head, let's hope the patch gets into 1.2!
Have you got any idea on how to setup call pickup pressing the blinking
button on snom phones?
Tnx!
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Alessiomailto:[EMAIL PROTECTED]
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) while calling
put the hint of the called to X (blinking light, cant remember which
state it is ) while phone is ringing, then to 1 if call is answered.
Unfortunately I dont know how to accomplish this
Regards!
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regards,
Alessiomailto:[EMAIL PROTECTED]
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Hi,
anyone can write down a working example of a regex fuction ?
I'm using this syntax
Gotoif($[${REGEX(/B/ | A)}=1]?20)
But function always return 1, even if I write
Gotoif($[${REGEX()}=1]?20)
Tnx for any help !
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Alessio mailto:[EMAIL PROTECTED
get the 300
ringing ... this looks more rrmemory than roundrobin, there is
something wrong in my setup maybe ?
Tnx !
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or they will stick to ULAW they used for first part
of the call ?
A quick test showed that they will use ULAW ... can I work around this
or am I getting something wrong ?
Tnx for any help !
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Alessio mailto:[EMAIL PROTECTED
Hi,
I get this message after password request in voicemail app:
Unable to create lock file: No such file or directory
Anyone got a clue about fixing that problem ?
I can't understand what directory or file we are talking about ..
Tnx for any help!
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Alessio
I think that's mostly right, but it should also be a native
xfer function working the same way regarding of the user agent, some
sort of common ground we can trust for installation with mixed
devices.
By the way: anyone got experience in attended trasfer with snom ? :)
Alessio Focardi
PF Oh
in the phone, better yet, is
up to you to choose then.
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to work, is there a
way to reproduce this behavior in asterisk ?
For what I can see it's not possible and you will have to select two
codes, one for blind and one for attended tranfers
What do you think about it ?
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Alessio mailto:[EMAIL PROTECTED
Hello Michael,
Wednesday, July 20, 2005, 11:54:40 AM, you wrote:
MP Alessio Focardi wrote:
Hi,
I'm experimenting attended calls tranfers and I'm a little bit
confused.
MP SNIP
MP I honestly think that transfers is one thing that Asterisk should
MP improve a LOT to be able to stand up
is
happening on my side.
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!
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!
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parameter (see dmesg)
FATAL: Error running install command for wctdm
What relates wcfxs to the wctdm that I was using previously ?
Maybe deleting wctdm
DO On Fri, 18 Mar 2005 15:17:57 +0100, Alessio Focardi
DO [EMAIL PROTECTED] wrote:
Hi,
I was using a TDM400P with cvs version
is called instead.
Any idea ?
TNX !
DO On Fri, 18 Mar 2005 15:32:02 +0100, Alessio Focardi
DO [EMAIL PROTECTED] wrote:
Hello Dana,
Friday, March 18, 2005, 3:23:36 PM, you wrote:
DO If you have any FXS ports, use wcfxs.
No, only green modules.
But this is what I get when loading driver
/wctdm.ko': No such
file or directory
That does not look normal to me, I have built another kernel to try to
make this behavior go away, still no luck
Tnx anyway ...
SG On Fri, 18 Mar 2005 16:02:23 +0100, Alessio Focardi
SG [EMAIL PROTECTED] wrote:
Hello Dana,
Friday, March 18, 2005, 3
:29 WARNING[19097]: loader.c:509 load_modules: Loading module
res_config_mysql.so failed!
libmysqlclient is present on the system, should I edit something to point *
to the right directory for it or something like ?
Tnx for any help!
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struggling to get it working with the BRISTUFFED version of *
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a merger, or can help me in such
task ?
Tnx !
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for realtime to begin stable ? :)
Tnx !
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} is not solved as _3XX.
If I change in table ${IPPHONES} with _3XX all returns normal.
So my conclusion is that variables can not be used as extensions in
realtime contexts, the actually work for all the other usual purposes
anyway.
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for your IP Phones ? let me change
a variable and we are set!
It seems that this is not supported, am I getting somethig wrong in
the syntax? There is another way to accomplish that ?
Tnx!
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Hello Dave,
Monday, January 17, 2005, 12:50:13 PM, you wrote:
DC On Mon, 2005-01-17 at 12:30 +0100, Alessio Focardi wrote:
Hi,
I tried set up a global var for an extension, like this
[globals]
IPPHONES=_3XX
[sip]
exten=${IPPHONES},1,Answer
What I would like to do is to make
for any suggestion!
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this is not a bug, it's a feature! :)
Seriously, anyone verified my problem and it's willing to share a
solution if there is any ?
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Hi,
I'm testing realtime right now, it does not seem to me that realtime
contexts can be included in normal context, like this
[sip]
include=sip-dial
exten=i,1,Hangup
[sip-dial]
switch=Realtime/sip-dial
Am I getting it wrong ?
Tnx !
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on screen ?
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Any idea of what I'm getting wrong ?
tnx !
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To my experience you will need bri_cpe as signaling (point2point),
immediate=no (if you have more than one numer on the ISDN),
pritrustusercid = no.
Regarding the pridialplan and prilocaldialplan I suggest to make some
experiment there.
Hope it helps !
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Alessio
are the significant
changes this addon have brought in * and what are the differences
between realtime and config in sql ?
Tnx for the support !
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.
MB RealTime does NOT force you to use itself.
Sure, I'm testing it right now ... looks VERY nice, writing a gui or
automating some common task now looks a lot easier!
The link for anyone interested:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime
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Hi,
since I run asterisk as root with a CLI open on TTY12 I was wondering
if the ! (shell) command can be disabled from the config, for safety
reasons it seems me usefully.
Tnx for any help !
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for the purpose.
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, ZapHFC HIAX and the need to
H install Capi !
H Please suggest best and easiest approach ?
I'm pretty satisfied with bristuff package from
http://www.junghanns.net/asterisk/
It downloads, patch and compile asterisk for HFC cards, also TE mode is
supported.
Good luck!
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the bristuffed version of asterisk ?
http://www.junghanns.net/asterisk/
Exactly what is the problem you are experiencing ?
Regards !
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ip any idea about such behaviour ?
Tnx !
P.S.
Fw version is snom190-SIP 3.46
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licence
yet.
Anyone has tried and is willing to share his impressions ?
TNX !
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stack
As you can see there variable CALLERID is empty, why ?
I tried also with CALLERIDNUM, same result.
Tnx for any help .
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friend I think, because sometimes caller id
AS information is not immediately sent. I'd wait one or two seconds
AS before dialing out again.
Will try this at once, tnx !
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CALL (no answer yet)
PRI call answer on called extension pickup
if extension is busy
Busy tone to PRI CALL (no answer at all)
Hoping for help ... tnx !
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stack
As you can see there variable CALLERID is empty, why ?
I tried also with CALLERIDNUM, same result.
Tnx for any help .
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Hello Jason,
Friday, August 27, 2004, 12:18:23 PM, you wrote:
JW On Thu, 26 Aug 2004 17:31:46 +0200, Alessio Focardi
JW [EMAIL PROTECTED] wrote:
Also dialing out works like a charm, the only problem is that calling
out asterisk is displayed on the called phone instead of the sip address
box.
I googled around but I have find nothing usefoul by now ... any guess?
Tnx !
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) and with my Grandstream ATA sip device (less easy it seems)?
Tnx !
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MM 10x
MM Maurizio
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PROTECTED] On Behalf Of Alessio
RTW Focardi
RTW Sent: 30 June 2004 10:12
RTW To: Tomaz
RTW Subject: Re: [Asterisk-Users] zaphfc - hfc pci based ISDN card :
RTW point2point DDI
RTW Hello Tomaz,
RTW Wednesday, June 30, 2004, 10:58:56 AM, you wrote:
T hello,
T anyone has worknig ISDN hfc-pci card
PROTECTED] On Behalf Of Alessio
RTW Focardi
RTW Sent: 30 June 2004 10:28
RTW To: Robinson Tim-W10277
RTW Subject: Re[2]: [Asterisk-Users] zaphfc - hfc pci based ISDN card :
RTW point2point DDI
RTW Hello Robinson,
RTW Wednesday, June 30, 2004, 11:19:35 AM, you wrote:
RTW We are using the HFC card
Sorry for the stupid question:
What's the purpose of defining a peer as trunk in iax.conf ?
The question is also valid generally speaking (for other channel
types), for instance: why define a Zap group as trunk in
extension.conf ?
Tnx for any help !
--
Best regards,
Alessio
Hi,
has anyone succesfully installed such scenario ?
I'm having problem with Award bios mb pc's... it do works with others,
what's your idea ?
Tnx !
--
Best regards,
Alessio mailto:[EMAIL PROTECTED]
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another console, like, for instance a find / then slips away again.
I suspect an Irq problem, what do you think ? What kind of problems
have you found with dell's ?
Tnx for the help !
--
Best regards,
Alessiomailto:[EMAIL PROTECTED
Hello Robinson,
Thursday, June 17, 2004, 1:19:12 PM, you wrote:
RTW Hi Alessio
RTW Yes, the problems you report do seem similar to the issues
RTW I had. I found on the Dells that the audio prompts were very
RTW choppy and played slower than normal. Occasionally there would
RTW be 'bursts' oav
parameters ... still nothing !
Tnx for the help !
--
Best regards,
Alessiomailto:[EMAIL PROTECTED]
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Hi,
I'm pretty new to asterisk so excuse the stupid question:
what is the purpose of defining channels as trunks ?
I noticed that you can define Zap groups and IAX connections as trunk,
but the purpose is not clear to me ...
Tnx !
--
Best regards,
Alessio mailto
Hello Brent,
Wednesday, June 9, 2004, 7:13:52 PM, you wrote:
BF On Wed, 9 Jun 2004, Alessio Focardi wrote:
Asterisk with one HFC isdn card, using the zaptel driver bristuff
All works ok, but voice coming in/out of the isdn card is out of sync,
squelky and disrupted, UNTIL I PUT SOME LOAD
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