TC wrote:> on a zap channel you can do for example
exten -> 1,2,Dial(Zap/1rN) where rN is
(r1=quick chip+normal ring; r2=British style ringing; r3=three short bursts;
r4=long ring).
Are there any options that would work with an ATA186 as the instrument?
Thx.
B.
_
Right, I was reading about them a few months back.
I was really excited there for a second.. 'WOW! three provinces away, ..I'm
famous!
So, how are you finding Asterisk? I may be building a toll bypass circuit
with it in a few weeks. While I'm talking about it, is there a gui with
asterisk?
_
on a zap channel you can do for example
exten -> 1,2,Dial(Zap/1rN) where rN is
(r1=quick chip+normal ring; r2=British style ringing; r3=three short bursts;
r4=long ring).
see also this patch to allow userdefined ring candence
http://www.marko.net/asterisk/archives/0212/0318.html
maybe CAM has ke
Hi!Steve,
Thank you for your reply. I think the change for Tor ISA card from T1 to
E1 might be as following:
T1E1
VCXO crystals 6.1756MHz8.192MHz
framer chips DS2151 DS2153
and the jumper E1-1 and E1-2 should be closed.
I don't know if what my
Perhaps I am associating you with the wrong company ... there is a Storm
Communication in the Ottawa area that is an ISP
It was recommended by users on a LUG mailing list I subscribe to but if I remember
correctly, they didn't provide service in my area
Storm3 Communications ([EMAIL PROTECTED])
As a sidenote ... can asterisk generate distinctive ringing for the analog extensions?
Jim Archer ([EMAIL PROTECTED]) wrote*:
>
>Hi All...
>
>Can Asterick detect distinctive ringing on a POTS line and answer with
>different configurations?
>
>Thanks...
>
>
In a quick search of the list archives, I found no mention of the recent
CERT advisory concerning vulnerabilities in some implementations of the
SIP protocol (i.e. whether or not * users were impacted by it, and if
so, to what extent and/or in what configurations), so I figured it would
be worthwhi
To the guy who wanted to know if the Tiger 560 card was a X100P - I don't
think so.
Here's the lspci -vv from a fairly new X100P:
00:09.0 Communication controller: Tiger Jet Network Inc. Model 300 128k
Subsystem: Unknown device 8085:0003
Control: I/O+ Mem+ BusMaster+ SpecCycle- Me
- Original Message -
From: "Brian Johnson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 04, 2003 11:46 AM
Subject: RE: [Asterisk-Users] OT: PRI costs in US
> I'm right here ... where are you?
Right about here..
>
> Just kidding, I'm in Stratford, Ontario, Canada
I'm
Hi All...
Can Asterick detect distinctive ringing on a POTS line and answer with
different configurations?
Thanks...
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
I tested both MySQL and PostgreSQL under various loads, Actually
got paid to do it. In the simple case where one process just
stuffs data into the database with INSERT MySQL was about 4x
faster in terms of INSERT/second
However if you have five or six processes that are doing lots
of transaction
Hi, Mark.
I'm posting a little patch that permits to set the
busycount value in dsp (via chan_zap), for systems that have "busydetect=yes" in
zapata.conf and have noticed sporadic hangup,
caused by incorrect cadence detection during normal conversation.
With this patch, it's possible to i
> You may know I work on a call center application using gnophone; I would
> like people to be able to login / logoff the asterisk server without
> opening closing gnophone. Right now, I have added a toggle button (Login
> / Logoff) to the interface, and the call to iax_reg() is delayed until
> the
On Tuesday 04 March 2003 02:35 pm, John Lange wrote:
> We have a S100U which we were using as our sole internal phone
> extension.
>
> Today it died.
>
> Picking up the phone now results in a brief dialtone followed by
> bursts of random static and crackling noises and then a second or
> two of fas
Is the tiger 560 the same as the X wildcard ?
Thanks,
Francisco
-Original
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Underwood
Sent: Tuesday, March 04,
2003 9:30 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: Torment
> Picking up the phone now results in a brief dialtone followed by bursts
> of random static and crackling noises and then a second or two of
> fast-busy followed by silence and more crackling noises.
>
> I've tried the hardware attached to a couple of different machines as
> well as a couple of di
Asterisk cannot detect a fax until *after* the line is answered. You'll
need to answer the line and then dial if you want to use that feature.
Mark
On Tue, 4 Mar 2003, Darrell Eldridge wrote:
> I can't seem to make the fax detection work. Here's
> an excerpt from zapata.conf:
>
>signalling
Hi i am a newie, and would like to know if the "TIGER" 560 IS THE SAME
as the X100 Voip card from Digium ?
Is it the same manufacturer ?
Will the Tiger board work under linux ?? anyone..
Thanks,
Francisco Perez-Landaeta
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTE
When I posted my last message about the cellsocket, I hadn't thought to
try incoming calls. (I bought the cellsocket to use only for outgoing
calls.)
Unfortunately, I cannot get the cellsocket to work for inbound calls.
CallerID does not work even though I have a GSM phone. The cellsocket
answer
I agree. postgres could be too slow if used into a big
system and you can't afford a rather good machine.
mysql is very fast and simple.
and I don't see where's the problem into the extraction...
a simple division could be done in any language, with any
program...
matteo.
Il mar, 2003-03-04 al
I beleive there is some issue/bug in the low level fax detection logic in
dsp.c
see dtmf_detect
if you look in chan_zap.c around lines 3066-3088 & stick a DEBUG msg in
there
you prolly notice that that the FRAME type is not set to DTMF & the subclass
<> 'f'
Mark says Thr or Fri this week he may h
We have a S100U which we were using as our sole internal phone
extension.
Today it died.
Picking up the phone now results in a brief dialtone followed by bursts
of random static and crackling noises and then a second or two of
fast-busy followed by silence and more crackling noises.
I've tried
Try exten => s,1,Answer
Rich.
>I can't seem to make the fax detection work. Here's
>an excerpt from zapata.conf:
>
> signalling=fxs_ks
> group=0
>
> context => guestaccess
> channel => 47-48
>
>and from extensions.conf:
>
> [guestaccess]
> include => incomingmain
>
> [incomi
Fax detection doesn't work here either. It's probably a tone detection
thing and needs to be tweaked.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Darrell Eldridge
> Sent: Tuesday, March 04, 2003 3:00 PM
> To: [EMAIL PROTECTED]
Might you want to Wait for a few seconds, then Answer the
line, to give the tone detection a chance before passing a
call to your desk extensions?
-Tilghman
On Tuesday 04 March 2003 01:59 pm, Darrell Eldridge wrote:
> I can't seem to make the fax detection work. Here's
> an excerpt from zapata.c
I can't seem to make the fax detection work. Here's
an excerpt from zapata.conf:
signalling=fxs_ks
group=0
context => guestaccess
channel => 47-48
and from extensions.conf:
[guestaccess]
include => incomingmain
[incomingmain]
exten => s,1,Dial,Zap/1&Zap/9&Zap/10&
On Tuesday 04 March 2003 01:12 pm, Steven Critchfield wrote:
> On Tue, 2003-03-04 at 12:57, Matthew S. Hill wrote:
> > I am pushing all the cdr info to a MySQL database on a separate
> > machine. I have noticed that the duration times for all calls are
> > recorded in seconds, by Asterisk. Is there
Why would anyone use such a big axe for a small problem, a trigger to do simple math. Bah.
MySql is perfectly suitable for CDR logging on any practical phone system short of telco main office
eqpt.
Karl Putland wrote:
On Tue, 2003-03-04 at 11:57, Matthew S. Hill wrote:
I am pushing all the cdr
In your select statement, where the column list is, use 'round(duration/6)/10 as duration' instead
of just duration.
Also, it would be trivially easy to modify the source of cdr_mysql.c to change it although you would
then have to watch on updates from CVS.
Bill
Matthew S. Hill wrote:
I am pus
On Tue, 2003-03-04 at 11:57, Matthew S. Hill wrote:
> I am pushing all the cdr info to a MySQL database on a separate machine.
> I have noticed that the duration times for all calls are recorded in
> seconds, by Asterisk. Is there a way to set the recorded call duration
> to a decimal representa
On Tue, 2003-03-04 at 12:57, Matthew S. Hill wrote:
> I am pushing all the cdr info to a MySQL database on a separate machine.
> I have noticed that the duration times for all calls are recorded in
> seconds, by Asterisk. Is there a way to set the recorded call duration
> to a decimal representa
I am pushing all the cdr info to a MySQL database on a separate machine.
I have noticed that the duration times for all calls are recorded in
seconds, by Asterisk. Is there a way to set the recorded call duration
to a decimal representation of minutes? ie 90 sec = 1.5 min. My
extraction process
I'm right here ... where are you?
Just kidding, I'm in Stratford, Ontario, Canada
As an aside, I looked at switching to your company for internet access from
Bell HSE but switched to istop.com instead.
We have a branch office in Kemptville (currently using Bell HSE - but only
until I get time to
On Tue, 2003-03-04 at 11:52, Chris Wetemans wrote:
> I was considering buying a devkit, bit it looks like it's not longer
> available.
> Will there be a replacement devkit?
> If not, where to buy a reasonable priced channel bank?
> Which channel bank was included in the devkit and where can it be b
I was considering buying a devkit, bit it looks like it's not longer
available.
Will there be a replacement devkit?
If not, where to buy a reasonable priced channel bank?
Which channel bank was included in the devkit and where can it be bought?
(btw I live in Holland)
Chris
_
have you any zaptel device in your box?
a zaptel device is required for timing
source for the conference (so meetme)
matteo
Il mar, 2003-03-04 alle 16:49, Rattana BIV ha scritto:
> Hi,
>
> I try the application MeeMe but i Have a problem when I call a conference.
> It show me : Unable to open ps
On Tuesday 04 March 2003 10:58 am, Steve Murphy wrote:
> I've proven to myself, via throw-away addresses, that individuals
> and robots sift thru archives and mailing list subscription lists,
> gathering addresses to sell on CD's to spammers. No list is immune,
> especially if it or its archives ar
On Tue, 2003-03-04 at 10:31, George Bean wrote:
> I have wanted to setup an Asterisk system for several months but have
> been unable to do so because I lacked the funds for a T1 card to
> connect my channel bank to the server. Recently, I acquired several
> Brooktrout T1/E1 interface (PRI-PCI48VC/
Hello, everyone.
I've proven to myself, via throw-away addresses, that individuals and
robots sift thru archives and mailing list subscription lists, gathering
addresses to sell on CD's to spammers. No list is immune, especially if
it or its archives are available over the web. I've been spammed
I have wanted to setup an Asterisk system for several months but have
been unable to do so because I lacked the funds for a T1 card to connect my
channel bank to the server. Recently, I acquired several Brooktrout T1/E1 interface
(PRI-PCI48VC/PRI-PCI64V-C) cards for the PCI bus. According t
Uncomment ztdummy from zaptel/Makefile
make clean ; make install
modprobe ztdummy.
Restart asterisk, all fixed.
Rattana BIV wrote:
Hi,
I try the application MeeMe but i Have a problem when I call a conference.
It show me : Unable to open pseudo channel
Does anyone can help me ?
regards
Rattana
But you can connect several asterisk boxes as one system.
regards
Martin
On Tue, 4 Mar 2003, Sphyrna wrote:
> NO, THE ASTERISK HAS A PRATICAL LIMIT OF 8 E1S CURRENTLY. THE I/O ERRORS
> STOP EVERYTHING
> - Original Message -
> From: "Florian Overkamp" <[EMAIL PROTECTED]>
> To: <[EMAIL PRO
On Mon, 3 Mar 2003 07:15:03 +0100
"Victor Sanchez" <[EMAIL PROTECTED]> wrote:
> when i have used modprobe tor2 i halt my PC.
>
> and i need to reset it.
What gcc You are using?
I found out that after building tor2 with new gcc-3.2
it halts the system
The solution is: in Makefile edit CC variabe
Hi,
I try the application MeeMe but i Have a problem when I call a conference.
It show me : Unable to open pseudo channel
Does anyone can help me ?
regards
Rattana
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listi
NO, THE ASTERISK HAS A PRATICAL LIMIT OF 8 E1S CURRENTLY. THE I/O ERRORS
STOP EVERYTHING
- Original Message -
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 04, 2003 8:35 AM
Subject: Re: [Asterisk-Users] 32 E1 or 64 E1 Configuration ?
> At 20:16
>I've got asterisk working as a PSTN gateway between two sites. The
>IP connection is pretty good and bandwidth exceeds 640Kbps all the
>time. I'm using X100P hardware on 300Mhz and 333MHz systems. One of
>them has a sound card, but the slower system does not have a sound
>card. Both are connec
I recently purchased a Cellsocket, which is a cradle that holds some
older Nokia GSM/PCS phones and converts them to an FXS interface. My
test phone is a Nokia 5190 on the T-Mobile GSM network.
After going through the ordeal of unlocking the phone (T-Mobile provided
unlock codes that didn't work)
I cannot seem to make the directory work properly ( as I see it ).
When I call directory with this setup and press the 3 letters, it locates the person
and plays their
name. When I hit '1' to call it hangs up without any fanfare or message on the
console ( verbose
999 ) It doesn't play goodbye
Hi,
I've got asterisk working as a PSTN gateway between two sites. The
IP connection is pretty good and bandwidth exceeds 640Kbps all the
time. I'm using X100P hardware on 300Mhz and 333MHz systems. One of
them has a sound card, but the slower system does not have a sound
card. Both are connec
On Tue, Mar 04, 2003 at 02:35:04PM +0100, Florian Overkamp wrote:
> At 20:16 4-3-2003 +0900, you wrote:
> >Is it possible to support 32 or 64 E1 in a linux box with Wildcard E400P
> >board ?
> >I'd like to make large scale PPS system.
>
> Hmm, 8 to 16 PCI slots ? That'll be a challenge. I think t
Ok. With loops=30 I got some "autoanswer",
but more rarely . Let's say that on 300 autogenerated
calls the issue happens half than without the patch.
I've raised the loops to 50, and on 300 testcalls
I never had a autoanswer. So that seems to works.
Matteo
> -Messaggio originale-
> Da:
> I have reverted back to the March 2nd CVS in the meantime, but can fire
> up the broken version just as easily.
Can you get me some sort of trace of the call? Mainly the output from
"sip debug" of an incoming call would be great. Thanks!
Mark
___
A
At 20:16 4-3-2003 +0900, you wrote:
Is it possible to support 32 or 64 E1 in a linux box with Wildcard E400P
board ?
I'd like to make large scale PPS system.
Hmm, 8 to 16 PCI slots ? That'll be a challenge. I think the current
practical maximum would be 12 or maybe 16 ?
Florian
Hi John,
[EMAIL PROTECTED] wrote:
> Hi!Steve,
> Glad to receive your message. Could you tell me if the card can work
> properly in Asterisk? and what is the different bettween the Tormenta
> ISA E1 card and the ISA T1 card (I refer to the hardware). Thanks.
> john
The differences between the T1
Is it possible to support 32 or
64 E1 in a linux box with Wildcard E400P board ?
I'd like to make large scale
PPS system.
Thanks in
advance.
See Young
Oh.
DIGITALWAVE,INC.
Earlier today, after updating from CVS this evening, I noticed that no
incoming calls had come in from my iconnecthere account since the
update.
Being that this is an odd thing, I tried to call, and after some
checking, determined that Asterisk was disconnecting the incoming calls
during the nego
Hemant Kumar wrote:
I am also looking for HylaFax and * together.
What I did was:
Hylafax<-->t38modem<-->chan_oh323<-->Asterisk<-->T100P<-->PSTN
But it didn't work.
Nope, chan_oh323 doesn't support T.38, at least not yet.
Michael.
If people in the list are interested and start working on this
I found it it's the option t in an extension
example : exten =>s,1,Dial,CAPI/25:B21|10|t
- Message d'origine -
De : "Rattana BIV" <[EMAIL PROTECTED]>
À : <[EMAIL PROTECTED]>
Envoyé : lundi 3 mars 2003 11:03
Objet : [Asterisk-Users] Transfert calling
> Hi,
>
> It is possible to do a tra
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