Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-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: &quo
someone can help me to solve this problem?
thanks
--
From: "Alessio"
Sent: Friday, September 02, 2011 5:10 PM
To: "Lee Howard"
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asteris
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
**
-- Executing [06456789@IncomingFAX:2] Dial("SIP/06456789-0003",
"IAX2/iaxmodem") in new stack
-- Hungup 'IAX2/iaxmodem-2218'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/0465940394-0002'
risk-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]
Sep 1 16:50:11 FAXServer FaxGetty
risk-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]
Sep 1 16:50:11 FAXServer FaxGe
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
[iaxmode
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@lists.digium.com
So I can't do anything?
--
From: "Kevin P. Fleming"
Sent: Friday, July 29, 2011 4:48 PM
To:
Subject: Re: [asterisk-users] call forwarding number from outside.
On 07/29/2011 10:41 AM, Danny Nicholas wrote:
Couple of questions -
This "magic tr
utside.
That`s 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 A
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 t
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 u
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! =
r, i think, to release 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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Alessio
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,Alessio
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ronet
web site
http://www.beronet.com/index.php?option=com_remository&Itemid=38&func=selectfolder&cat=1&lang=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 co
his is by design), then
why on earth was the RRMemory strategy created??
Thanks for your response, Alessio.
~~Aaron
- Original Message -
From:
Alessio
Focardi
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Cc:
[EMAIL PROTECTED]
Sent: Thursday, June 29,
h a "count us" initiative :)Alessio FocardiOn 6/29/06,
Aaron Paxson <[EMAIL PROTECTED]> wrote:
I have setup several Calling Queues, each setup
with RoundRobin strategy. When I call the queue, the first
member/agent phone rings. Great! I call it again, the second
member/agent
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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Alessio
other side is allays complaining about 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_
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 pb
k with meetme rooms, since no extension is created opening a conference ... anyone has used the patch with meetme and is willing to share ?Tnx for the support!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Alessio
FocardiSent: Thursday, April 06, 2006 4:34 AMTo:
as
ntroducing
jitter, 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> Li
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
absolutely great variable=0 led off, 1 led on, 2 led blink ... Alessio Focardi
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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 basical
lling more 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 [
: 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
ug the ethernet cable from the asterisk server
while having a conversation between 2 phones: call should stay up.
Hope it helps!
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k/sounds/pbx/webradio,http://grace.fast-serv.com:9206/
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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Alessiomailto:[EMAIL PROTECTED]
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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 work
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 priva
up some times ago 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
Congratulations from Italy now back to work for 1.3 ! :)
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Hope it helps!
P.S.
italiano ? :)
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ence suppression in your
softphone (in xlite is named "transmit silence").
Hope it helps!
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Asterisk
be able to
see on the cli the called number.
Then you will have to create the relative extensions in the incoming
context ... just "s" will not work anymore.
Hope it helps!
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Hi,
anyone has a working example of this new function ?
that's all that I have found
-= Info about function 'REGEX' =-
[Syntax]
REGEX("" )
[Synopsis]
Regular Expression: Returns 1 if data matches regular expression.
[Description]
Not available
Tnx!
-
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
?
Tnx!
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!
s> 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
any help
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are working using 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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__
aller to 1 (steady ligt) 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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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
l again you get the 300
ringing ... this looks more rrmemory than roundrobin, there is
something wrong in my setup maybe ?
Tnx !
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Ast
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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Al
more intuitive/instructive
AG> transfer process.
All I'm asking is a native function that can be used regardless of the
UA, if you got such functions integrated in the phone, better yet, is
up to you to choose then.
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the way: anyone got experience in attended trasfer with snom ? :)
>>
>>Alessio Focardi
PF> Oh, you mean the completely natural feeling "put them on hold, dial
PF> new party, tell them you have a transfer, hit transfer"? I want some of
PF> whatever kool-aid the person who t
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>
MP> I honestly think that transfers is one thing that
pbx's users are expecting call transfer 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
is happening ?
I dont have access to callman logs, so I can only report what is
happening on my side.
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t; Asterisk-Users mailing list
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r any help!
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FC3/misc/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:
>> Hell
cfxs wctdm 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.
&g
own 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
r any help!
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me ?
I'm struggling to get it working with the BRISTUFFED version of *
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b 17 15:17:29 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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knoww if someone has done a merger, or can help me in such
task ?
Tnx !
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other than wait for realtime to begin stable ? :)
Tnx !
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le
it does not work anymore,${IPPHONES} 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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gt; its database driven?
So 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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dialplan.
Am I getting it all wrong ?
Tnx for any suggestion!
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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
>>
>&g
ant 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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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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: have you got any message on screen ?
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, then do so.
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
--
Best
re or less).
Can someone more skilled than me describe what 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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riment there.
Hope it helps !
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does not
result created
Any idea of what I'm getting wrong ?
tnx !
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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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e you using the "bristuffed" version of asterisk ?
http://www.junghanns.net/asterisk/
Exactly what is the problem you are experiencing ?
Regards !
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Asterisk-
rrect 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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king for help, when
is morning here most of the members of the list are still sleeping
AS> Anyway, Wait() is your 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.
-- Executing VoiceMailMain("Zap/8-1", "@domain") in new 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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-- Executing VoiceMailMain("Zap/8-1", "@domain") in new 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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ringing
Ring tone to PRI 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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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 t
asterisk
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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To
and how it
can be controlled ?
Tnx !
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on: 128 taps unless TDM bridged, currently OFF
MM> PRI Flags:
MM> Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7351
MM> zap_show_channel: Failed to get conference info on channel 1
MM> Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7357
MM> zap_show_channel: Failed to get confmute info
W> From: [EMAIL PROTECTED]
RTW> [mailto:[EMAIL 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>
CTED]
RTW> [mailto:[EMAIL 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,
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Best regards,
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,
Al
ed with hdparm, setting dma mode 3 and other
parameters ... still nothing !
Tnx for the help !
--
Best regards,
Alessiomailto:[EMAIL PROTECTED]
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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
something over
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,
Alessio
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