If you don't have any hardware for conferencing than you could use the
ztdummy from zaptel package. Check the archives. look for ztdummy
Martin
On Tue, 27 May 2003, Rahul Gupta wrote:
> Hello ,
>I am a newbie to * and have just been able to call
> a sip User Agent on a dif
I didn't know that one can run Ethernet over T100P.
Now what NIC card are you using ?
Martin
On Wed, 28 May 2003, Nick Eggleston wrote:
> Digium T100P
>
> On Wed, 28 May 2003, Martin Pycko wrote:
>
> > What network card are you using ? (model and vendor)
> >
>
It should detect the incoming call on a 2nd or 3rd ring.
I don't know about the limit though.
regards
Martin
On Wed, 28 May 2003, WipeOut . wrote:
> Hi Mark,
>
> I have tried a value of 10 and still had the problem.. How high can this value be
> set and what would the effect
FXO ports don't get DID numbers usually so they'll always go to 's'
Martin
On Wed, 28 May 2003, Jon Pounder wrote:
> When immediate is set on a port that is an fxo, what is the meaning of this ?
>
> Will it go immediately to the "s" extension of the
Check if you can enable "remote disconnection supervision" with your PBX.
If not you may try using the software detection when you specify
callprogress=yes
or
busydetect=yes
before the definition of your channel in zapata.conf
Martin
On Wed, 28 May 2003, Manuel Marin Ga
Nope.
usecallerid=no should work for it.
If not you might try to modify the code in chan_zap.c
Martin
On Wed, 28 May 2003, Jon Pounder wrote:
>
> my question was -> will immediate put an end to the extra 2 rings before
> pickup ?
> (I know they go to "s" eventually.)
Make sure that you don't have a R2 signalling. Since then you'll have
problems EuroISDN PRI is all right.
Martin
On Wed, 28 May 2003, Ricardo Saar Gemignani wrote:
> Hello
>
>I'm starting to learn about Asterisk and trying to install the first one.
> I
Try to add this line to zapata.conf
relaxdtmf=yes
before the "channel => " definition
Martin
On Wed, 28 May 2003, Nick Eggleston wrote:
> We've got an asterisk system hooked up to a number of telephones via a channel
> bank.
>
> [*]T100P---CAC(access bank)---Ph
You _CAN_ use a wildcard on the callerid matching. It goes through the
same code.
regards
Martin
On Thu, 29 May 2003, Jamie Carl wrote:
> I was just thinking that. Shouldn't this be a feature?
> I'm sure coding it would be a cut and past job. :)
>
> Another one f
Try running asterisk like this:
screen -d -m asterisk -vvvc
or
screen -d -m asterisk -c
or
screen -d -m asterisk -f
Martin
On Thu, 29 May 2003, Tjardick van der Kraan wrote:
> When we have the G.729 codec (ordered from digium) active in * we have the
> following problem:
>
>
Do you have your zap channel in asterisk when you type "zap show channels"
?
If not than make sure you have a proper config files (zaptel.conf &
zapata.conf)
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> Hi list,
>
> I have the follow configuration:
> ==
Lets say that your E1 channels are assinged to
context=incoming
channel => 1-15,17-31
Then in extensions.conf in context
[incoming]
exten => fax,1,Dial,Zap/1 ;if your Zap/1 port is FXS where the fax
;is attached
(all other extensions)
regards
Martin
Then propably your board stoped taking interrupts. Try changing the PCI
slot or IRQ. Make sure you don't run X-windows.
regards
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 11:41:01 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
>
>
What bandwidth do you have available for you connection (upsteram and
downstream)? Do you have any CIR for VSAT connection ?
Martin
On Thu, 29 May 2003, Jim Ockers wrote:
> Hi all,
>
> For some reason VSAT or Satellite Internet services are not mentioned
> (or searchable) in
Check whether "strace -xx cat /dev/zap/1" gives you any output
If it stops and waits than your board is not taking interrupts.
Is the board running on the separate IRQ ?(/proc/interrupts)
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 14:08:
So now that I finally relize that you're using T1 or E1 circuit
Do you have a ISDN PRI or an analog ciruit ?
What's the status of the span in zttool or in (/proc/zaptel/1).
Is it OK, RED, YELLOW ?
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 1
Didn't you just write a post before that it was running ?
The EBUSY means that you propably have asterisk running and the port is
busy or you have strace line on some other console
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 14:32:37 -0500 (CDT)
> M
So it means that the board is working all right but there is problem with
the telco or you're using diffrent signalling for your circuit.
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 15:06:12 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
I think they are hardcoded. But what do you exactly refer to by
"signalling bits" ?
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
> On Thu, 29 May 2003 15:26:25 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
>
> > So it means that the board is
Asterisk doesn't support it yet.
Martin
On Fri, 30 May 2003 [EMAIL PROTECTED] wrote:
> We are installing our second long distance ISDN T1 and I've been given the option of
> using the same
> D channel to control both long distance T1s.
>
> Does asterisk support signa
Comment out dmtfmode=inband or change it to something else.
With low-bandwidth voice codecs we don't have a good chance to decode
DTMFs, etc.
Martin
On Mon, 2 Jun 2003, Paul Cheng wrote:
> Hi,
>
> I searched the archives about this, but didn't find any references.
> When
Even after you reload the modules for the board ?
What about "ztcfg -vv" ?
Martin
On Mon, 2 Jun 2003, Tais M. Hansen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi,
>
> No matter what I configure my spans at (on a E400P) ztcfg -v always shows:
>
The transfer application generates the flash on analog interfaces.
It won't work w/SIP.
Martin
On Tue, 3 Jun 2003, John Todd wrote:
>
> OK, I'm stumped. I have no idea how one would use the Transfer
> application. Perhaps it is because I am an all-SIP environment, but
It might be done using the chan_local channel driver,
You could add this member in queue.conf
member => local/[EMAIL PROTECTED]
and in extensions.conf
[timeout]
exten => s,1,Wait,600
exten => s,2,Voicemail,b1000
I don't know if that'll work but it's worth checking.
what happened.
regards
Martin
On Sat, 7 Jun 2003, Omar Abhari wrote:
> I am not sure if anyone is having this same problem, but, with * as the
> IVR on 2 long distance T1's that we have, serving some 16000 customers,
> as they enter their phone numbers or any other group of digits, some
&
Or compile for PROC=i586 in asterisk/Makefile
Martin
On Sat, 7 Jun 2003, Gary wrote:
>
> try putting in modules.con
>
> noload => ??
>
> On Sat, 07 Jun 2003 00:04:56 -0400, hallian hallian wrote:
>
> >Hello all -
> >
> >This is my situ
It's easier to read a patch when you send make it with "diff -u"
or "cvs diff -u"
Martin
On Sat, 7 Jun 2003, John Congdon wrote:
> Another thing I am working on is to do the timeout people have been
> asking about.
> If they have been on hold for (X minutes) du
It's fixed now.
Martin
On Sun, 8 Jun 2003, Stephen Davies wrote:
>
>
> On Sat, 7 Jun 2003, Daryl Jones wrote:
>
> > I experienced the exact same symptoms but didn't have the confidence
> > to post my experience to this list because of my lack of experience
Hi,
I am just about to move out from my parents home and think about how I
will phone from now on. In Germany there is a provider (QSC) who
offers DSL (1024 down/256 up) with fastpath without volume or time
limits.
Does anybody know a comercial (or even semi-professional) provider who
lets
switchtype=national is National 2
switchtype=ni1 is National 1
The first is well tested the last was added recently
and is not propably tested well.
regards
Martin
On Sun, 8 Jun 2003 [EMAIL PROTECTED] wrote:
> Ok that's cool. I'll stick with the national because that's whats
-23
signalling=fxo_ks
group = 2
context = internal
callerid = <1000>
channel => 25-48
Notice that I skipped many other settings that you propably need to
familiarise yourself with (look in zapata.conf.sample)
regards
Martin
On Sun, 8 Jun 2003 [EMAIL PROTECTED] wrote:
> Can anyone ex
It doesn't happen on Nortel 350.
Martin
On Mon, 9 Jun 2003, David Carr wrote:
> I have the same problem. I use an Aastra 480 phone and as long as I don't
> touch any of the ADSI soft-buttons then my keypad stays active and the
> downloaded script works great. But as soon as I
Try in /etc/zaptel.conf to add this line:
alaw=1-4
sine by default E&M is used in US and the ulaw codec is being used.
Martin
On Tue, 10 Jun 2003, Eduardo Goncalves wrote:
> Hi list,
>
> I have an E400P using only one span with 4 channels, using E&M immed
: 4670 4548 4614 4518 IO-APIC-level tor2
All the four CPU's should have IRQ's like in the example above.
Martin
On Mon, 9 Jun 2003, Alex Zarubin wrote:
> Hi,
>
> We are trying to validate Asterisk as a media gateway PRI <-> SIP with two
> T400
Did you do "ztcfg" after you added that line ?
Martin
On Tue, 10 Jun 2003, Eduardo Goncalves wrote:
> On Tue, 10 Jun 2003 09:37:22 -0500 (CDT)
> Martin Pycko <[EMAIL PROTECTED]> wrote:
>
> > Try in /etc/zaptel.conf to add this line:
> >
> > alaw=1-4
&g
Well normally the telco phone line is an FXS line so you need
an FXO port to connect to it (e.g. X100P). However if your line is FXO
then you need FXS ports and TDM400P should work
Martin
On Tue, 10 Jun 2003, James Sizemore wrote:
> Can I use a WILDCARD TDM400P to connect to
> four
Did you configure the circuit in /etc/asterisk/zapata.conf ?
What do you see when you do "pri intense debug span 1" ?
Do you see SABME being sent out by asterisk and no response ?
Martin
On Wed, 11 Jun 2003, Mark McKibbin wrote:
> Can anyone give us a clue on setting up a E100P we
Did you recompile zaptel for -D__SMP__ ?
Check the zaptel/Makefile
Martin
On Wed, 11 Jun 2003, Carlos Carús wrote:
> Hi!
>
> I have the chance to play with a couple of E400P cards, each installed
> in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI
> HDD wi
Do you see in /proc/interrupts that tor2 receives IRQs on both CPUs ?
Martin
On Wed, 11 Jun 2003, Carlos Carús wrote:
> Martin Pycko escribió:
>
> >Did you recompile zaptel for -D__SMP__ ?
> >Check the zaptel/Makefile
> >
> >Martin
> >
>
> Yes, I did
It should be good enough. The problem is propably in software
configuration
Martin
On Wed, 11 Jun 2003, [UTF-8] Carlos Carús wrote:
> Jared Smith escribió:
>
> >I have a funny feeling your crossover cable might be wrong... I'm not
> >sure about an E1 crossove
Why do you think so?
Local variables get lost only when the call gets hanged up.
Martin
On Wed, 11 Jun 2003, Paulo Mannheimer wrote:
> Hi,
>
> Seems that my local variable content get lost when I call an AGI
> program. Is this the correct functionality?
>
> Thanks,
>
Notice that you should refer to PHONE_NUM variable this way:
${PHONE_NUM}
Martin
On Wed, 11 Jun 2003, Mark Street wrote:
> I am having a problem understanding/visualizing the environment of AGI and how
> variables defined there can be used in my dial plan. I am so close I can
> tas
Mark did the commit so I guess he'll add it when he gets a chance.
Martin
On Thu, 12 Jun 2003, Michiel Betel wrote:
> I just downloaded the latetst CVS. A compile now complains about a missing
> srv.c & srv.h used in chan_sip.c. Can they be added?
>
> --
> Betel Con
Hi,
* Erik Lagerway wrote/schrieb:
>
> There is a provider in the US -> www.AddaLine.com, who just launched a
> SIP> service with some great rates for North America
>
> I have been using their service for months and I am extremely happy with
the
> service.
looks like Germany is again
Check the line 118 of extensions.conf ???
Martin
On Thu, 12 Jun 2003, Derek Beaumont wrote:
> Whenever I issue the reload command, asterisk crashes. Below is the
> output I get from
> (gdb) bt
>
> Any help
His problem was that he had only one number assigned to the whole E1. So
telco didn't send any called number in SETUP. Adding immediate=yes
to zapata.conf helped here.
Martin
On 13 Jun 2003, Levent Guendogdu wrote:
> Hi Dave, hi all,
>
> I've the same problem for a few days n
Since you're using the sound card for testing you need to
change in the /etc/asterisk/alsa.conf or oss.conf
context=local
to
context=default
regards
Martin
On Fri, 13 Jun 2003, Moshe Yudkowsky wrote:
> I've built the latest CVS of asterisk -- not the zaptel or libpri
> direc
I think when you exceed the txgain or rxgain settings than the echo
canceller might turn off.
You can find if the pending call has echo canceller turned on when you do
"zap show channel " on the CLI.
Martin
On Fri, 13 Jun 2003, John Congdon wrote:
> Does anyone know what this mea
I'd valgrind asterisk or just start by removing everything else
other than e.g. one interface definition in extensions.conf:
[channels]
signalling=...
channel => a-b
Martin
On Fri, 13 Jun 2003, Derek Beaumont wrote:
> I am still getting a segmentation fault when I try to relo
You may need to copy the files in /var/spool/asterisk/outgoing every
second or half a second.
Martin
On Fri, 13 Jun 2003, Thomas Haeger wrote:
> OK, sorry for my deficient description...
>
>
> Scenario is as follwes:
>
> One 4 BRI card ->
>
> ttyI0 - ttyI7 fo
Well I just checked the zaptel.c not guessed and it looks
like this message pops in when the fax/modem transmit the echo canceller
disable tones.
regards
Martin
On Fri, 13 Jun 2003, Martin Pycko wrote:
> I think when you exceed the txgain or rxgain settings than the echo
> canceller migh
You have a RED alarm on the link.
Check also " head /proc/zaptel/1"
So either you have wrong framing, no CRC4, or a diffrent timing or the
circuit is disconnected.
Martin
On Fri, 13 Jun 2003, Jorge wrote:
> Hi,
>
> I have an E100P card.
>
> My zaptel.conf is:
> spa
You're missing that then the IAX call will be started between ast1 and
ast2 and you'll get connected to ast2 Zap/1
Martin
On 13 Jun 2003, Eric Wieling wrote:
> As I understand it (and my understanding is obviously incorrect) the
> switch => statement sells the Asterisk
But he didn't think about agent. Just a regular SIP phone.
It should be in general like the original author of this thread thinks.
Besides it's easy to test so why not to test it :)
Martin
On Fri, 13 Jun 2003, TC wrote:
> >Hi.
> >
> >I was wondering how can I make i
The idea of switch is for every box to know what it can reach locally. And
then to do the 'switch' to remote boxes if the called number can't be find
locally.
Martin
On 13 Jun 2003, Eric Wieling wrote:
> Cool.
>
> Now if I am on ast-1 and want to call 2200, which is a Za
Over what interfaces ? (voip, analog t1, pri ?)
In general when you want to send it over T1 to the telco and further on to
PSTN than it might not be possible since you're allowed most of the times
to send the callerid that is one of your assigned DID numbers.
regards
Martin
On Fri, 13 Jun
Check "show application setcallerid"
Martin
On Fri, 13 Jun 2003, Derek Beaumont wrote:
> I only want to do this internally, from the reception phone to another
> phone attached to my asterisk box.
> I am using X100P and TDM400P.
>
> -Derek
>
>
>
> >>O
I think it's per context.
Martin
On Fri, 13 Jun 2003, Andy Powell wrote:
>
> So is that one switch statement per installation or one per context.
> ie can i have multiple switch statements each one applicable to a
> different section in extensions.conf
>
> Andy
>
Then I guess in zapata.conf before the definition of the
callerid=asreceived
channel => 1;FXO port
Martin
On Fri, 13 Jun 2003, Derek Beaumont wrote:
> I don't understand how or where I would use setcallerid.
> I have tried to do
> exten=>400,1,Setcallerid,asreceived
>
/etc/asterisk/manager.conf
Martin
On Sun, 15 Jun 2003, Alvaro Parres wrote:
> Hi,
>
>Any of you know where to define the user and password for gastman.???
>
>
> PLEAS HELP ME!
>
> Alvaro Parres
>
>
> ___
>
Try to explicitly add this line
,1,SetCallerid,("somename" <12345>)
,2,Dial,Zap/g1/${phonenumber}
regards
Martin
On Tue, 17 Jun 2003, Tom De Wispelaere wrote:
> Hey all,
>
> I have a E1 setup with a E400P digium card. Everything works just great
> except for the calle
Did you cvs update zaptel and recompiled ?
Martin
On Tue, 17 Jun 2003, K. C. Li wrote:
> On Tue, 17 Jun 2003, Mark Spencer wrote:
>
> > I'm in Paris right now and can't test this change, but I've been
> > researching the DAA and there are a few international s
Of course you can use Gotoif with expressions.
Gotoif,$[${VAR} > 12]?1|4:1|5
Martin
On Tue, 17 Jun 2003, Anthony Minessale wrote:
>
> I just made my first 2 modules for asterisk (The 1st one is depriciated already).
>
> I was annoyed that i couldn't get GotoIf to take any e
Just do the "pri debug span 1" and see for yourself that asterisk sends
that. You might however send it without one digit or something ... or
maybe your telco doesn't support it. Just give then a call.
Martin
On Tue, 17 Jun 2003, Tom De Wispelaere wrote:
> Hey Martin, t
Do you have '-z' option with the definition of random in musiconhold.conf
? actually I just did see the options of mpg123 and it has to be an
uppercase Z:
-Z
Martin
On Tue, 17 Jun 2003, Rafael Gonzalez Lomeña wrote:
> Hello to all,
>
> I want to put incoming calls in a
We have it done at Digium so it can be done.
Just record your name I guess with voicemail but I'm not entirely sure
about that you can record that in voicemail.
Martin
On Tue, 17 Jun 2003, Derek Beaumont wrote:
> I'm wondering if I can do the following:
>
> Caller acti
Describe that a little bit.
The call came on what interface and then you dialed what interface
and how did you park it ? You pressed a flash button or '#' key ?
Martin
On Tue, 17 Jun 2003, John Congdon wrote:
> Has this been solved? When I park a call, the caller hears a second
ction (less false hangups)
although I tested the algorithm without
-DBUSYDETECT_TONEONLY nor -DBUSYDETECT_COMPARE_TONE_AND_SILENCE
with busycount = 10 and after 1 hour of conversation I didn't have any
false hangups.
regards
Martin
___
Asterisk-Users
Did you try to use 'w' as a digit before dialing the number like this:
exten => _X.,1,Dial,Zap/1/w${NUMBER}
You could also try to put 'w' inbetween the digits.
regards
Martin
On Tue, 17 Jun 2003, John Laur wrote:
>
>
> Quite frequently, outgoing calls fr
Well it was in
#error You can't
^
sorry about that.
Martin
On Wed, 18 Jun 2003, The Traveller wrote:
> Yo Martin,
>
> On Tue, Jun 17, 2003 at 17:03:15 -0500, Martin Pycko wrote:
>
> > Hello,
> >
> > I've commited the new busydetect ro
That's what you get when you modify your code and that modification is in
conflict with the CVS.
Martin
On Wed, 18 Jun 2003, John Congdon wrote:
> O -fPIC-c -o chan_agent.o chan_agent.c
> chan_agent.c: In function `login_exec':
> chan_agent.c:595: parse erro
You can use 'at' utility to copy a file that you prepare
one you execute AGI script. Look at asterisk/sample.call.
Martin
On Wed, 18 Jun 2003, Xisco Mateu wrote:
> Hi all,
>
> Now I'm working with a E400P, and I don't now if it's possible to do the following.
You can call setmusiconhold app and as an argument call class silence,
off, or whatever non-existant class and it works now.
Martin
On Wed, 18 Jun 2003, TC wrote:
> Yea, I have faked that with a silent mp3,
> but to do it right it should also be a config flag in the agent.conf file
>
You could always have
exten => asterisk,1,VoicemailMain
Martin
On Wed, 18 Jun 2003, Test wrote:
> Does anyone know if this was implemented? If not then where should I look to
> try and make the mod?
>
> Thanks
> Tan
>
>
>
> - Original Message -
> Fr
This works for me.
Martin
#!/usr/bin/perl -w
use Socket;
use IO::Handle;
socket(SOCK, AF_INET, SOCK_STREAM, getprotobyname('tcp'))
or die "Cannot create a socket: $!\n";
connect(SOCK, sockaddr_in(5038, inet_aton('localhost')))
or die "Cannot c
Anyone in New Zealand using AsteriskPBX? If
so, what hardware are you using to connection to Telecom's lines?
How old is your zaptel code ?
Mark recently increased some timer for that.
Martin
On Wed, 18 Jun 2003, Sam Bingner wrote:
> I have an extension setup with voicemail, for incoming calls on an X100P
> card. It quite often will record about 15 seconds of dialtone... I'm
> guessing
nce
time */
try the same value as in wcfxo.c
recompile/reload and test
regards
Martin
On Thu, 19 Jun 2003, Sam Bingner wrote:
> Zaptel was the version from about 4 days ago when I sent this message, I
> updated again yesterday night
>
> Sam
>
> Quoting Martin Pycko <[EMAIL PROTE
s designed to be working only in US.
Martin
On Thu, 19 Jun 2003, Dan Fernandez wrote:
> I have an X100P and when I place calls to the PSTN which are not answered, the
> Billsec field of the CDR still logs the seconds that the phone rang.
>
> Can someone please confirm that t
You need to change the FREQs for the events. I don't know exactly how the
code works. There was someone on the list that claimed to have the UK
callprogress working.
regards
Martin
On Fri, 20 Jun 2003, Tan Aks wrote:
> Isn't there any way to make callprogress work for people in Eu
cdr_mysql.conf
On Fri, 20 Jun 2003, carlos del mayor wrote:
> hi
> I want to do a database to save the cdr with a billing finality. I've created the
> database in mysql (thanks for the table and all that!) but I'm not sure of how to
> 'connect' asterisk to that database in order to save there t
Well if you have lots of /dev/timer opened than you have to edit your
asterisk/Makefile and comment out ZAPTEL_OPTIMIZATIONS or something like
that.
Martin
On Fri, 20 Jun 2003, Derek Beaumont wrote:
> What is the recommended version of mpg123?
> I am running
se. It would help if you would send 'sip debug' along
with 'pri debug ...'
Martin
On Sat, 21 Jun 2003, Daryl Jones wrote:
> I'm running a pretty successful Asterisk system and recently moved our
> PRI to a T100P board. The PRI was previously connected to a Cisco
asterisk -vvvcn | tee /tmp/log
CLI> sip debug
CLI> stop now
or
script
asterisk -vvvcn
CLI> sip debug
CLI> stop now
shell> exit
Martin
On Sun, 22 Jun 2003, destan wrote:
> Hi everybody,
> I want to read to debug messages and try to interpret them but they happen
> too f
The hardware we are planning to use
is:
Micronet SP5050 FXO Gateway
http://www.micronet.com.tw/Products/VoIP/SP5050.asp
Micronet SP5100 IP Phone
http://www.micronet.com.tw/Products/VoIP/SP5100.asp
We are hoping to use this hardware along with
AsteriskPBX to replace our aging PBX system.
Well how did you solve your previous problem then ?
Martin
On Mon, 23 Jun 2003, Thomas Haeger wrote:
> The problem before is solved. But now gives another problem ...
>
>
>
> == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI)
> == Registere
You need to find out which way your SIP gateway wants to receive the
DTMFs. There are three ways to do that. Read sip.conf.sample.
Martin
On Mon, 23 Jun 2003, Dave Alan Caruana wrote:
> hi there,
> I have an installed & working Asterisk server,
> which I am using to connect to
THat's not it.
in zapata.conf you *also* need to have
signalling=pri_cpe or pri_net
Martin
On Mon, 23 Jun 2003, Michael Bielicki wrote:
> On Monday 23 June 2003 2:58 pm, Anton Yurchenko wrote:
> > Hello,
> >
> > I have an E100P, and in the zaptel.conf I have:
> &
If you use brackets () then you need to call it like this
PGSQL(blabla(bla)bla)
That should work
regards
Martin
On Tue, 24 Jun 2003, Thomas Haeger wrote:
> Hi all again,
>
> if i make a query with
> ...
> exten => _X.,2,PGSQL,"Query resultid ${connid} SELECT getdest(&#x
I think that if you put
exten => _X.,1,DIal,Zap
it'll improve the matching dramatically
Martin
On Wed, 25 Jun 2003, John Todd wrote:
>
> My synapses are rather fried after a long few days of debugging other
> problems, so perhaps I'm being lazy in sending this to the
td->init_v3_2 = sin(-2.0 * M_PI * (freq2 / 8000.0)) *
gain;
Martin
On Wed, 25 Jun 2003, Surfer Dude wrote:
>
> I am wondering if anyone could help me figure out how to turn down the volume on all
> the dial tones, indications, etc.. and especially the call-waiting CHIRP!
>
>
Unless your telco signals hangup with a dialtone . it should help.
The thing is that most propably your X100P hangs up and then picks up the
line due to something ... that was my original idea.
Martin
On Wed, 25 Jun 2003, Sam Bingner wrote:
> I don't understand how that would af
It's either EuroISDN or E&M w/E1. And for incoming calls you'll get what
you need if the telco sends it on the channels.
Martin
On Thu, 26 Jun 2003, Scott Stingel wrote:
> Hello-
>
> I know this is a basic question, but before I start down the road of using
> Asteri
Try to put
noload => chan_oss.so
in modules.conf
also do you use mpg123 with musiconhold ?
Martin
On Fri, 27 Jun 2003, Dave Alan Caruana wrote:
> hi there..
> I have an asterisk installation with a PRI-E1 card
> running EuroISDN, installed on a 1GHz Intel Celeron
> box wit
ISDN PRI E1 is enough to receive DID and CallerID (ANI).
Martin
On Mon, 30 Jun 2003, Surajee Ratnayake wrote:
> hi everybody,
>
> my question is specific to ISDN signalling,
> in my set up, i want to get cli and dnis into my asterisk box, and i am going to use
> ISDN PRI E1s co
You need to look at "show application meetme" in the asterisk CLI
but for it to work you need to have some kind of zaptel hardware or
emulate it with zttdummy (but for that you need to have usb-uhci like USB
controller)
and then
exten => 1000,1,Meetme,1000
Martin
On Tue, 1 Ju
To pick up a call that rings someone elses phone that is in the callgroup
as your pickupgroup.
Martin
On Tue, 1 Jul 2003, carlos del mayor wrote:
> Well, I suposse is a very basic question but,,,for
> what is used: callgroup=1 and pickupgroup=1 ?
> thanks!
> c.mayor
>
&g
You have to call Dial with ||m option to have music-on-hold while
transfering
Read the "show application dial"
Martin
On 1 Jul 2003, Fabrice Tereszkiewicz wrote:
> Hello,
>
> I'm trying to do something really easy : transfer a PSTN call to a H323
> client. This wor
The meetmecount app is supposed to tell you the number of participants in
a certain conf number. However it does not create the var variable.
The error about "wrong use of LEN(" was do to the fact that your var
variable does not exist and it was a bug. It's fixed now.
Martin
On
(*8) and
transfered with blind transfer to Zap.
It worked fine.
regards
Martin
On Tue, 1 Jul 2003, Chad Sawyer wrote:
> I have a small problem,
>
> Whenever we pickup a call using *8 then try to transfer it via flash or # transfer
> the call is dropped. Any ideas? Whe have all ca
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