At the moment asterisk can get the callerid from the From: field.
regards
Martin
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
Hi
I have defined my SIP phones like this ...
[Sip1]
username=gs1
callerid= Full name 1001
etc etc
Now, when I do this in a given
exten = _0X,1,Dial,Zap/g1/0${EXTEN:1}
Martin
On Wed, 9 Jul 2003, Petr Michálek wrote:
Hi!
Is there simple way how to add prefix to dialed number?
I need change 0X. to 0X.
Regards
Petr Michálek
___
Asterisk-Users mailing list
[EMAIL
You forgot about _ in front of 0X
Martin
On Thu, 10 Jul 2003, BK [address only for mailing lists] wrote:
On Thursday, July 10, 2003, at 06:37 AM, Petr Michálek wrote:
Is there simple way how to add prefix to dialed number?
I need change 0X. to 0X.
how about this
exten =
IAX2 uses hardcoded 4569 port so it's not looking for port keyword.
Nothing to worry about.
Martin
On Thu, 10 Jul 2003, Richard Scobie wrote:
When starting *, I get the following when the chan_iax2.so loads:
[chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
== Manager registered action
That got implemented recently ...
Martin
On Tue, 8 Jul 2003, carlos del mayor wrote:
Hi Rattana,
That kind of transfer is not yet implemented in *. The
way it will be indicated is:
exten =111,dial,Zap/1,20,T
The T indicate that transfer is permitted for calling
party, but as I've said,
Well first of all if you set up DigitTimeout to 5 seconds so asterisk is
going to wait up to 5 seconds to retrieve the digits specially when you
have a match of _X. that is at least to digits but with the timeout of 5
you could imagine that asterisk will intercept all digits.
How about having a
Did asterisk register with both accounts ?
sip show registry
Can you post what happens on the console along with 'sip debug' ?
Martin
On Tue, 8 Jul 2003, Derek Beaumont wrote:
Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a
Busy is n+1 if n+101 doesn't exist.
Martin
On 8 Jul 2003, Steven Critchfield wrote:
On Tue, 2003-07-08 at 10:10, Derek Beaumont wrote:
Has anybody out there tried to use two different iconnecthere accounts
with Asterisk?
What I want to do is use a second account if the first is busy.
I
How about that:
exten = _91NXXNXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED]SIP/${EXTEN:[EMAIL
PROTECTED]
Martin
On Tue, 8 Jul 2003, Derek Beaumont wrote:
Asterisk has registered with both accounts:
sip show registry
Host Username Refresh State
213.137.73.178:5060
On your place I would check separately if you can use both accounts. I
think that one of your accounts in disabled ...
Martin
On Tue, 8 Jul 2003, Derek Beaumont wrote:
First off, sorry for using a mail client without the in-reply-to
function.
Second: I still can't make two calls using
You plug a channel bank to a T1 in your PC connected either over T100P or
T400P.
regards
Martin
On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote:
Hello everybody
My doubt is about configuration. Can I use a channel bank like zplex-10 or
adtran, plug on it an T1, 24 POTS, an ethernet
The system at the moment can run some program/script but there is no way
to retrieve the results. Although you could have tried with
${ENV(VARIABLE)}
Martin
On Mon, 7 Jul 2003, WipeOut . wrote:
Can the system command be used to retrieve a variable from a mysql database using
the mysql
overlapdial=yes in zapata.conf
for those channels that you want the overlapdialing be activated.
By default only incoming overlap dialing is enabled.
regards
Martin
On 7 Jul 2003, Thilo Salmon wrote:
Hi,
I am lost trying to figure out how to enable overlap dialing for calls
coming in and
Come on,
exten = *98,1,Dial,Zap/g1/BYEXTENSION
should work since it's old sytax.
It's more propable that you have that *98 in a diffrent context
than assigned for those channels or you don't have the group 1 defined
properly
Martin
On 7 Jul 2003, Steven Critchfield wrote:
On Mon,
But in this case you should have said:
What exten for picking *the oldest* or *the first* call that got there
Martin
On Fri, 4 Jul 2003, Matteo Brancaleoni wrote:
It's an offence ;) ?
yes I know that.
But in that case I needed to be able to pick the older
call in the parking lot, w/o
Lets say that you're going to receive 20 diffrent DIDs
1000 - 1019
[incoming]
exten = _X.,1,SetVar,SAVE_DID=${EXTEN}
exten = _X.,2,Goto,${EXTEN}|s|1
[1000]
exten = s,1,blabla
[1019]
exten = s,1,..
Martin
On Fri, 4 Jul 2003, Steven Kawuma wrote:
Hi all,
I'm trying to write a set
CDR is used for that.
Martin
On Fri, 4 Jul 2003, destan wrote:
Hi all,
Does Asterisk report or keep a database of the duration of SIP Calls? Is CDR
used for this? If there are people using this facility, how accurate is it?
Thanks in advance,
Umut
show application authenticate ?
Martin
On Fri, 4 Jul 2003, Kim C. Callis wrote:
As I was working on my extensions.conf file, I started to segment
calling privileges. For the everyday workers, I don't free reign to LD
access unless it is business related. So I was wondering if there was a
You have one board on the same IRQ as VGA. Try to replace the board in a
diffrent PCI slot or maybe you can do the IRQ job in the BIOS.
Martin
On Fri, 4 Jul 2003 [EMAIL PROTECTED] wrote:
Debian / kernel 2.4.18.
Found in /proc/pci ::
Bus 0, device 0, function 0:Host bridge: Intel
Also you can have the wrong pridialplan in zapata.conf. Look at pri debug
span 1 for q931 trace.
Martin
On Fri, 4 Jul 2003, Stefano Finetti wrote:
Greetings.
Today I've installed a fresh new E100P on a EuroISDN PRI.
It seems to work well, accepting calls, but, when I start *, I have
the
That means that asterisk is sending SIP messages but gets no response from
the device.
Martin
On Thu, 3 Jul 2003 [EMAIL PROTECTED] wrote:
Hello All!
There is description of my problem with Asteriks below.
Asteriks CLI says:
File chan_sip.c, Line 415 (retrans_pkt): Maximum retries exceeded
You Answer on analog channels and then you need to have
a fax extension in the current context.
regards
Martin
On Wed, 2 Jul 2003, Joe Antkowiak wrote:
How do you tell asterisk to detect for fax tones?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
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 Jul 2003, Serge
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
--- Louis-David Mitterrand
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 Tue, 1 Jul 2003
What configuration of hardware/software are you running. I just checked
picking up with *8/transfer on zaptel/SIP and it works on our Digium PBX.
I placed a call from SIP to Zap, picked it up with Zap (*8) and transfered
to Zap
and also
place a call from Zap to Zap, picked it up with SIP-Snom200
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 coming from telco.
To
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 with 256Mbytes RAM.
CPU
Try to change something in zaptel/tonezone.c around these lines
/* Bring it down -8 dbm */
gain = pow(10.0, (LEVEL - 3.14) / 20.0) * 65536.0 / 2.0;
td-fac1 = 2.0 * cos(2.0 * M_PI * (freq1 / 8000.0)) *
32768.0;
td-init_v2_1 = sin(-4.0
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko
Sent: Thursday, June 19, 2003 11:57 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoiceMail recording dialtone
Well experiment yourself with the code.
in wcfxo.c
/* Don't accept a ring
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 general
list, but I can't
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)
== Registered channel type 'Tor' (Zapata Telephony
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:
span=1,1,0,ccs,hdb4,crc4,yellow
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 fast, how can I log these guys to
is
it that is needed to make it work?
T
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 19, 2003 11:36 PM
Subject: Re: [Asterisk-Users] Billsec on CDR
It has to do with the fact that with analog channels like FXO
we don't have
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 0.59r
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 that it picks up
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 PROTECTED]:
How old is your zaptel code
It has to do with the fact that with analog channels like FXO
we don't have a way to tell whether the call has been answered or not.
So after the interfaces sends the called number we assume that the
call got answered. This happens unless you have callprogress=yes
in zapata.conf. But it's designed
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
for each
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: $!\n;
connect(SOCK, sockaddr_in(5038, inet_aton('localhost')))
or die Cannot connect to the manager port\n;
Try to explicitly add this line
,1,SetCallerid,(somename 12345)
,2,Dial,Zap/g1/${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 callerid. When i make an outgoing call
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 settings I can
change, so I've changed
stack
-- Executing SetCallerID(Zap/6-1, somename 0) in new stack
-- Executing Dial(Zap/6-1, Zap/g2/X) in new stack
I still dont get any CLID on the receiving phone however :(
Maybe i should contact the telco about this ?
Regards,
Tom
On Tue, 17 Jun 2003, Martin Pycko
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 queue and that user hear
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 activates the Directory
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 of
music on
Hello,
I've commited the new busydetect routine to CVS.
You need to cvs update asterisk of course and then choose it
in asterisk/Makefile and recompile asterisk.
All you X100P users that had the problems
with false hangups or the card not being able to detect the busy tone
please check that.
In
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 from the X100P cards here will not dial
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 now on an E1 with an
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 channel_no on the CLI.
Martin
On Fri, 13 Jun 2003, John Congdon wrote:
Does anyone know what this means? It
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 for outgoing
second 4 BRI card -
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 might turn
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 incoming calls to wait if the phone
then I have to define 2200 on ast-1 and can't put it in the
master/central dial plan on ast-2?
On Fri, 2003-06-13 at 12:34, Martin Pycko wrote:
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
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 2003,
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
Over what interfaces ? (voip, analog t1, pri ?)
In
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
On 13/06/2003 at 13:28 Martin Pycko
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
but that doesn't seem to
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 Consultancy
Abelenlaan 19
1185 RT
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 is appreciated.
***
*CLI
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 :-(
--
Carlos Carús
Ingeniero de Sistemas
[EMAIL
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,
Paulo H. Mannheimer
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
taste it. I just
Are you sure that you compiled zaptel for __SMP__ ?
Edit your zaptel/Makefile.
0: 75283844 75241320 75286285 75247088IO-APIC-edge timer
1: 1 0 1 1IO-APIC-edge keyboard
2: 0 0 0 0 XT-PIC
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
sine by default EM is used in US and the ulaw codec
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 hit listen through
You configure zaptel.conf since zaptel interface drives the boards. In
general you can configure the hardware part (dchannels, clear channels -
dchannel needs to be run in HDLC mode, circuit timing, framing, etc)
Zapata.conf belongs to asterisk and you need to tell asterisk which
channels to use
I think that most of the companies that have IVR calling cards
applications experience similar problem. It's because of the cheap phones
some customers use or because of the noise on the line. So when they press
a DTMF digit the generated tone is interrupted by some short noise on
the line and the
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 situation! I have a PC with no PCI slot and no modem! But I would
Do you really have the channels in asterisk ?
zap show channels
Is the alarm on the E1 circuit ?
Martin
On Thu, 5 Jun 2003, Jay Banda wrote:
Hello All.
Does anyone have experience with the Valiant Comms vcl30 channel
and the Digium E100P in asterisk ? We have the vcl30 channel bank,
It does use sendmail. Which app are you using ?
voicemail or voicemail2 ?
Martin
On Thu, 5 Jun 2003, Derek Beaumont wrote:
I used to have email notification working with my voicemail services but
it stopped working when I installed the new version of asterisk.
I have not changed my
It should be exten=sip,1,Dial(SIP/sipphone)||t
Martin
On 6 Jun 2003, Dave Wolven wrote:
Hi Don't know if someone answered this yet...
when calling the dialapp append the |t to it
exten=sip,1,Dial(SIP/sipphone)|t
This will allow you to hit # and then the callparking extension.
Thanks
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.
regards
Martin
On 4 Jun
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
I don't see what purpose
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 I make an outbound SIP
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:
SPAN x: D4/ AMI Build-out: 0 db
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:
running * in standard
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:
===
extension.conf:
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
On Thu,
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:
Do you have your zap channel
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 this list's archives.
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 14:08:01 -0500 (CDT)
Martin
(CDT)
Martin Pycko [EMAIL PROTECTED] wrote:
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
The command strace -xx cat /dev/zap/1 didn't
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)
Martin Pycko
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:
Didn't you just
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 working all right but there is problem
What network card are you using ? (model and vendor)
Martin
On Tue, 27 May 2003, Nick Eggleston wrote:
We are using the zaptel driver to deliver a combined voice/data T1 circuit.
The data channel-group is using the cisco hdlc protocol (on the linux side)
and connects with a cisco router on
1) you need two accounts in iconnecthere
2) you need to register with two accounts
3) then simply receive the call using one and send it over another account
Martin
On Wed, 28 May 2003, pradeep kumar wrote:
Hi All.
I am trying to setup asterisk so that I can place two outbound calls via
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 different machine thru
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 context when the line
first rings, or something
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.)
At 10:40 AM 5/28/2003 -0500,
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've a doubt. Here in Brazil the
You configure them as usual
zaptel.conf
fxoks=1-n #(n - how many cards you have)
Then you can just plug a single phone line to each of them
and then in zttool which one will go into OK from RED state.
regards
Martin
On Thu, 3 Apr 2003, Jim Archer wrote:
Hi All...
If I have more than 1
Can you use Playback instead ?
Playback doesn't use mpg123.
regards
Martin
On Thu, 3 Apr 2003, Tamas Levente wrote:
Hey,
I've installed 0.59r mpg123 on a redhat box. I set the extension up for the
mp3player. I called and it was playing the file back,but it was full of drops. like
sound -
I woudln't write that if it wouldn't support mp3.
On Thu, 3 Apr 2003, Tamas Levente wrote:
And does playback support mp3?
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, April 03, 2003 5:58 PM
Subject: Re: [Asterisk-Users] MP3player
Some people run fax over IAX using ulaw codec on the local LAN.
Martin
On Thu, 3 Apr 2003, Brian J. Schrock wrote:
From what I have heard packetizing fax does not work well, does not
matter if it is IAX or SIP. I think that was straight from digium tech
support.
On Wednesday, April 2,
asterisk -vvvcg (use g option to generate the coredump file)
than gdb asterisk core.pid
bt
Also you might send a log of pri intense debug span number
regards
Martin
On Wed, 2 Apr 2003, Alex Zarubin wrote:
Configuration:
Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown
P4 2.5
cvs update -r 1.x channels/chan_sip.c
make install
where 'x' is from 1 to 30
version 1.30 is dated 2003-04-02
if not sure check rcs2log -v |more
regards
Martin
On Tue, 1 Apr 2003, Russ Beaupre, P.E. wrote:
A while ago SIP transfer via the # key on a call to a cell phone via
iconnect was
then download the sources and compile it ...
On Wed, 2 Apr 2003, it wrote:
I installed the cygwin yesterday. But it seems that the cygwin does not have
the cvs command.
$ cvs
bash: cvs: command not found
Regards
john
- Original Message -
From: Michael Bielicki [EMAIL
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