Can I get a copy of that PERL script?
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237 ICQ: 5662270
An update on this...
I was wrong. The wireless problem was an altogether different issue. the wj0011
firmware finally made my phone useable, after 6 months of problems.
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
hello list,
ineed to setup an asterisk system with 5 ISDN trunks. ifound C4 cards but they are very expensive.i found that if i use 5 AVM Fritz! cards itwould bevery cheap. i want to use 2 boxes. 3 in boxA +2 in boxB=5 isdn.
and i want,this two boxs to work as a single box so that one box can
Here's my suggestion. Do a dialplan thing where when all trunks on boxA are
busy, they are sent via IAX to boxB which sends them out via the ISDN
trunks... this way boxA will be your primary box and boxB is your spare box
that takes over if everything else is busy...
On Tuesday 13 September
No always possible to get a fixed IP address from your provider - in
Ireland anyway.
I've found a novel work-a-round: I have a server on the Internet in a
data centre that maps a real static address to the dynamic IP address of
the computer connected via. the ISP. I've got a script that
I've found a novel work-a-round: I have a server on the Internet
in a data centre that maps a real static address to the dynamic IP
address of the computer connected via. the ISP. I've got a script
that runs on the client ISP connected machine (its running Linux an
the script is in the
On 24/08/05, razza [EMAIL PROTECTED] wrote:
I have a standard BT home DSL, which means I cannot have a static IP
address,
Yes you can - depends on your ISP, but the underlying IPStream Home
products don't prevent the allocation of static addresses. It's all
down to who you buy your service
Hi,
currently I got the following problem:
The system contains a HFC-PCI card, it's running without problems under zaphfc.
Now this system also should send/receive faxes. Receiving faxes works with
rxfax (spandsp) but txfax is not comfortable enough and doesnt bring the
demanded results. So an
Marek Zachara wrote:
I have a new asterisk working in New Zeland and everything is working
well except when an incoming call to the PSTN hangs up, asterisk wont
hang up the zap trunk (X100P).
The New Zealand indications etc are in CVS. Give me a call on (03) 4555770
(Dunedin) if you are still
Hi all,
I'd like to use the w option of the meetme application.
>From tiki i read:
'w' wait until the marked user enters the conference
All other connected users will hear MusicOnHold until the marked
user enters.
The question is, how can i indicate the "marked user"?
thank's in
On Mon, Sep 12, 2005 at 06:00:25PM -0700, Matt wrote:
anyone knows how skype provide world wide call service to regular phones by
voip at such low rate?
is this by partnerships with various * isps?
Skype do deals with local telcos, they use G.729/SIP for SkypeIn/Out
between them and the local
On Tue, September 13, 2005 11:53, Accursio Avona said:
Hi all,
I'd like to use the w option of the meetme application.
From tiki i read:
'w' -- wait until the marked user enters the conference
* All other connected users will hear MusicOnHold until the marked
user enters.
Jens Wrote:
Who needs that when there's dyndns and similar free services which
are even supported by many routers? I have a dyndns hostname and my
router is configured to contact the dyndns site whenever the IP on
the public side changes. Works very well for my Asterisk setup at home.
I'm
On 13 Sep 2005, at 11:18, razza wrote:
Jens Wrote:
Who needs that when there's dyndns and similar free services which
are even supported by many routers? I have a dyndns hostname and my
router is configured to contact the dyndns site whenever the IP on
the public side changes. Works very
Really? I've found that a dynamic domain name -- IP linkup didn't work
with Asterisk - it seemed to caches the IP of the domain name and didn't
re-lookup the IP until after a reload.
Derek
Jens Vagelpohl wrote:
I've found a novel work-a-round: I have a server on the Internet in
a data
Skype prices is not that low. F.ex buying price today for Argentina Buenos
Aires is between 0,0050 and 0,0056 Euro and Skype charge 0,0170 euro.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent: den 13 september 2005 11:48
To:
Ray Wrote:
I'm sure if you use a DNS in SIP.CONF for your external IP this is
only
resolved when loaded?
Jens:
This might be true - for me there's only other Asterisk servers
connecting from the outside using IAX, and that works fine.
IAX is different! So back to sip, assuming if externip
We seem to be having problems at our office with lines locking up.
It mainly seems to occur when we have someone call into a Fixed Cellular
Trunk, at which point they use DISA to select an outgoing PSTN line or
the other way round, when a call comes in and is transferred out to the
cell phone.
On Monday 12 Sep 2005 21:53, Colin Anderson wrote:
The one I like:
http://www.rhetorical.com/cgi-bin/demo.cgi
is toast. I think they went broke or got aquired by someone. Also, is there
a Festival voice that sounds as good as Rhetorical or the AT T stuff? The
According the UK Companies
Colin Anderson wrote:
The one I like:
http://www.rhetorical.com/cgi-bin/demo.cgi
is toast. I think they went broke or got aquired by someone. Also, is
there a Festival voice that sounds as good as Rhetorical or the AT
T
stuff?
According the UK Companies House, they are still going.
Alex,
Context solved my problem. Thank you so much.
HappyDavid.
On 9/13/05, Alex Ongena [EMAIL PROTECTED] wrote:
On Mon, 2005-09-12 at 23:34 +0800, VoIP Newbie wrote: Below are what I have in extension.conf.
Is this the complete file ? exten = s,1,Goto(1234,1) ^^^is used to jump to
Hi everyone,
I decided to have a look at SIP NAT again and I've been at it for a
[quite a] few hours but typically nothing is working for me. Actually
I'm not sure if SIP and NAT can ever work but some emails on this list
do suggest that someone has got it working, once, maybe.
I'm
When Asterisk uses an ISDN interface, it periodically sends to CLI
messages such as:
== Primary D-Channel on span 1 down
[...]
== Primary D-Channel on span 1 up
Is there a simple programmatic way of capturing them for monitoring
purposes?
Enzo
___
We need to find out how to provide world wide international low rates by
using * , which company should we contact for a simple one stop shopping,
instead of looking for each country to find a * partner?
look for local teleco for international rate will be still more expensive
than skpe's low
This is fine but I need to be able to modify my sip.conf (externip =
w.x.y.z) and reload sip, does anyone know of a script/app which does an
nslookup and modifies the conf file, then reloads sip?
What I did was to have the ip checker write a one line file called
externip.conf containing the
Hello,
I'm using asterisk-1.0.9 and wanted to compile it with database support in
voicemail (postgres).
I made the following changes in the /apps/Makefile:
USE_MYSQL_VM_INTERFACE=0
USE_POSTGRES_VM_INTERFACE=1
Without theese changes asterisk compiles perfectly, but after setting the
postgres
Mensaje citado por: Derek Conniffe [EMAIL PROTECTED]:
Hi,
Does anyone know how to get NAT SIP working where the SIP phone is
behind a NAT server talking to a publicly accessible * server?
Have you tried sip-conntrack-nat for netfilter?. May be could help you.
Get pom-ng from
Hi,
Can some one tell me what is the meaning of all the fields of show queue
callcenter? for example in my system it gives:
callcenter has 0 calls (max unlimited) in 'roundrobin' strategy (33s
holdtime), C:429, A:12, SL:0.0% within 0s
How is the holdtime calculated? what is A and SL?
Derek,
You said -
Needless to say when I don't have any NAT settings on the SIP phone I
don't get any registration with the * server (this confuses me too - I'm
not sure why I only get registration when I set the * server to be the
outbound proxy? Maybe its because the SIP phone sends its
Hi,
I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and
wondered if anyone is able to offer any advice.
In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk
box. e.g: HiCom user dials access code and can call Asterisk extension or
establish SIP
I have 1.2.0 beta 1 running and it works fine. My x100p returns caller id
with no problems. When I test the CVSHEAD callerid fails with checksum and
len 0 errors.
I can run with the cvshead of zaptel and libpri with beta1 but only the
beta1 source works for caller id. Any source after beta1
-Forwarded Message-
From: IEG [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Fwd: Re: [Asterisk-Users] civil emergency comms: Asterisk
+ HAM]
Date: Tue, 13 Sep 2005 03:04:42 -0700
The answer is a multiplexed terminal node controller (TNC) This was the
very thought behind trunked
this is normal for p2mp lines, expecially in Italy,
where the Dchan is allowed to go down when no layer3
activity is on the BRI.
Matteo.
Il giorno mar, 13/09/2005 alle 19.50 +0800, Enzo Michelangeli ha
scritto:
When Asterisk uses an ISDN interface, it periodically sends to CLI
messages such
Curious if the perl scropt overcomes the 100 char limitation?
-Original Message-
From: Rod Bacon [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 13, 2005 12:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] OT: Online TTS
Hi ! :)
I'm having trouble with my MusicOnHold... I'm using standard files
(fpm-*.mp3), which are provided in * package (so I assume they're not
variable bitrate encoded... anyway, I tried re-encoding them and no
changes).
The music is played, but with micro-cuts every 2/3 seconds... it's
Hi all,
I tried to set the calleridname of an incoming call to get different
incoming labels displayed for different incoming numbers.
This does work for hidden number-calls so I can set the displayed CIDName
on my cisco7960 from CID withheld to abc CID withheld
If the incoming CID isn't hidden
Carlos Alperin wrote:
I don't get this?
Is included on 1.0.9 or is not? I know that a lot of people was trying it,
but just to be clear, is T.38 passthrough included on 1.0.9?
Thanks,
Carlos Alperin
No. We are not even sure at this point if T38 will make it into 1.2.
-Matthew
You probably want to use 'database put' for changing incoming CID
http://voip-info.org/tiki-index.php?page=database%20put
*CLI database put cidname 111222 test user
Updated database successfully
*CLI database show cidname
/cidname/111222 : test user
so now
Just setup the stls4 card to work in NT mode and connect the siemens pbx to
the asterisk with a crossover cable. Then you will be able to make calls to
from the hicom to the asterisk machine you do not need to have an nt box to
make the connection. With the nt box you can power an ISDN phone if
Is there a way to pass variables/arguments to the h extension ?
for example :
[default]
exten = _1098933X.,1,NoOp(CARRIER TWT-TIM, EXTEN: ${EXTEN}},
SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN})
exten = _1098933X.,2,SetVar(_PROVA=bla)
[lot of stuff, agi, goto, tricks and magic that
This should be of interest to us, asterisk buffs:
http://www.linuxdevices.com/news/NS6761928882.html-- We hold [...] that all men are created equal; that they areendowed [...] with certain inalienable rights; that among
these are life, liberty, and the pursuit of happiness-- Thomas Jefferson
Hello,
I would like to install the current beta1, but I can't find a working
bristuff package. On 'junghanns.net' I found two versions, one for use
with the stable (1.0.x) tree and one for the CVS. As far as I
understood, the CVS version should be the correct one. But when I try to
use the
I don't think using database is the solution for prepending a shortname to
the cidname based on the dialed incoming extension
The cidname isn't static...
You probably want to use 'database put' for changing incoming CID
http://voip-info.org/tiki-index.php?page=database%20put
*CLI database
running 1.2 on both servers
from A to B to a 7960
the 7960 receives callerid as
NAME
usernameofsipuser
i tried setting callerid etc before doing the dial
A to B is via iax B to 7960 is via sip of course
on B right before i dial 7960 i noop calleridnum and name and both populated ok
is
All of sudden my FXS module is not working.
I have a TDM card with one FXS and one FXO, FXO module seems working fine.
I also noticed the LED is not on for my FXS module while it is on for my FXO
module.
Sep 13 12:11:44 WARNING[9870]: chan_zap.c:887 zt_open: Unable to specify
channel 1: No such
Rich Adamson wrote:
Running fc3 with current cvs-head...
Is there a nice way to rotate the /var/log/asterisk/messages file without
shutting down asterisk?
I'm currently rotating the log files via cron, however my script requires
asterisk to be shut down, which also kills any outstanding cli
Hi all,
I'm having some problems getting TDMoE setup for the 1st time. I have a
TE405P installed in the main server with an ethernet cross-connection
to the secondary machine.
(Yes, I know about IAX2 but I want to use TDMoE to simulate using T1s.)
I'm using -HEAD from yesterday.
On the
On 11:36, Tue 13 Sep 05, Derek Conniffe wrote:
Really? I've found that a dynamic domain name -- IP linkup didn't work
with Asterisk - it seemed to caches the IP of the domain name and didn't
re-lookup the IP until after a reload.
Then why go with all the hassle of perl daemons and stuff ?
I don't recommend anyone use free dyndns via router support. If you reboot
your router more than once or twice in a month or have a power outage or
whatever dyndns stops updating the IP automatically and will cancel your
account for too much activity. You won't know it for a few weeks until they
Try this after your done rotating your log:
asterisk -rx reload
This is what I use..
..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-Original Message-
From: [EMAIL PROTECTED]
In article [EMAIL PROTECTED],
Simone Cittadini [EMAIL PROTECTED] wrote:
Is there a way to pass variables/arguments to the h extension ?
for example :
[default]
exten = _1098933X.,1,NoOp(CARRIER TWT-TIM, EXTEN: ${EXTEN}},
SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN})
exten =
Hi all,
I'd like to use the w option of the meetme application.
From tiki i read:
'w' — wait until the marked user enters the conference
* All other connected users will hear MusicOnHold until the marked
user enters.
The question is, how can i indicate the marked user?
thank's in
On 12:22, Sun 11 Sep 05, Paul wrote:
Rich Adamson wrote:
Running fc3 with current cvs-head...
Is there a nice way to rotate the /var/log/asterisk/messages file without
shutting down asterisk?
I'm currently rotating the log files via cron, however my script requires
asterisk to be shut
Hi Ray,
It would be great to find a solution which doesn't need modification of
the firewall setup (like if it was a customers firewall rather than your
own).
There is two things I'm wondering about: -
1) Can a Outbound SIP Proxy be a server out on the Internet (i.e. not
in the local
On Mon, Sep 12, 2005 at 08:48:31PM -0600, Gabriel Gunderson wrote:
Upgrade! Upgrade! Upgrade! At the very least, upgrade to
the last 1.0.x. Even better would be to use the CVS-HEAD
or the latest 1.2beta release of libpri, zaptel, and asterisk.
There are new features that help
Hi Ray,
I was wondering if the qualify option is used [in sip.conf] to keep a
connection (from the SIP phone inside the firewall to the Asterisk
server outside the firewall) open then would the firewall not allow two
way communication without incoming port mapping/NAT (providing that the
Hi Michiel,
The problem is the Asterisk server on the far away side (not the local
office behind the firewall/pppd) server. When a call comes in to the
far away (Datacentre) server it needs to be able to make an IAX2
connection back to the office server to carry the call inbound so the
Brian C. Fertig wrote:
Try this after your done rotating your log:
asterisk -rx reload
Or just a logger reload... I thought that was already said though.
Kevin
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing
What are you talking about?
In addition to having the dyndns account you have to update thier
servers with your info regardless of how many times your link goes up or
down.
There are many software packages that will do this for you and in some
distro's the app is included. Hell, even my
Accursio Avona wrote:
Hi all,
I'd like to use the w option of the meetme application.
From tiki i read:
'w' — wait until the marked user enters the conference
* All other connected users will hear MusicOnHold until the marked
user enters.
The question is, how can i indicate the
Hi All,
I need help to create one IVR Menu, when a say "Welcome to PBX Corp..." , press 1 to Sales, press 2 to Help Desk or wait to operator.
What function should I use for call transfer exten SIP to exten SIP. eg I call to extension 190 and after answer, I do one transfer to another exten SIP.
The problem turned out to be two phones sharing the same registration.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, September 12, 2005 8:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I have something similar to do SMS voicemail
notifications... I do not use any underscores when I set
the variable and it works fine in the 'h' extension.On 9/13/05, Simone Cittadini [EMAIL PROTECTED]
wrote:Is there a way to pass variables/arguments to the h extension ?
for example
This is the output:
-- Executing NoOp(Zap/4-1, We Are Here) in new stack
-- Executing Cut(Zap/4-1, myexten=ds|/|2) in new stack
-- Executing NoOp(Zap/4-1, 205|30|tr) in new stack
-- Executing Cut(Zap/4-1, mine=myexten|||1) in new stack
-- Executing NoOp(Zap/4-1, ) in new stack
I've got an * box that was performing happily for over a year now with
X100P and TDM400P FXO connections to the POTS. But recently the
landlord has changed the phone service in the building and now my phone
lines are through analog ports in his unspecified Nortel PBX.
The problem is *
Could you send me a copy of your script?
%- SNIP! -%
Wilson Pickett wrote:
What I did was to have the ip checker write a one line file called
externip.conf containing the line:
externip = nnn.nnn.nnn.nnn ; this is the new ip address
then in sip.conf,
#include externip.conf ; replace
When your landlord switched the phone service, he more than likely put on new service that doesn't supply remote disconnect supervision which was what was causing disconnects to be detected correctly before. They will need to activate this for you again in order for things to begin working again
Hi,
I was wondering if there is already an application or a simple
mechanism to convert the dialed extension into digits if letters were used.
I don't know if there is a name for that, I mean the letters on the phone
keypad: ABC=2, DEF=3, ...
So when I call e.g. JOE, the extension 563 shall be
I am currently attempting to get Asterisk working with our existing NEC IPK
192 system. The * box has a Wildcard X100P clone card in it and is hooked
to an analog port on the NEC system. The analog port is shared between an
elevator phone and a credit card machine. I have one SIP extension
On 9/11/05, Josip Gracin [EMAIL PROTECTED] wrote:
Hello!Is it legal to use RedirectAction to redirect a call that is waiting ina queue?
It works for me.
The idea is to have an external application manage a queue via managerAPI.The queuewould merely collect calls and play moh.
This is the same
On 18:13, Tue 13 Sep 05, Derek Conniffe wrote:
Hi Michiel,
The problem is the Asterisk server on the far away side (not the local
office behind the firewall/pppd) server. When a call comes in to the
far away (Datacentre) server it needs to be able to make an IAX2
connection back to the
Why do you need to write an application for this? Why don't you just
make joe extension 563??
--
Tom
On 9/13/05, Armin Schindler [EMAIL PROTECTED] wrote:
Hi,
I was wondering if there is already an application or a simple
mechanism to convert the dialed extension into digits if letters were
Does anyone have any sample configuration files for the Polycom IP500
phones that have been cleaned from the samples that come with the 1.5.2
firmware.
I'm wondering how much is necessary, and how much will just work by
default. I am really only setting the line authentication , and
Cody Lerum wrote:
Can I just pull unchanged lines out?
No. Many of the 'defaults' are only defaults because they are in the
sample configuration files, and if you upload new files that don't have
the defaults, the features will not work the same way (or at all). I
have personally seen Call
Hi All,
I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions.
But when I installed AMP and created new extensions, I'm not able to
call those extensions. I get the message that the extension is busy and
it is forwarded to
I am getting quite frustrated today, so please bear with me.
I just installed Fedora Core 4 (was running RedHat 9 with a working Asterisk)
now my Fedora does not appear to be recognizing my X100P (clone) at all.
Hardware browser just shows them as unknown device. driver: hisax
So, of
On Tue, 13 Sep 2005, Tom Hayden wrote:
Why do you need to write an application for this? Why don't you just
make joe extension 563??
If I would need this for just 'JOE', than I could do that. But for
about 100 names, it's easier to have a mechism for it.
Armin
--
Tom
On 9/13/05, Armin
Hi List,
Have you got experience with this product?
http://www.voipsupply.com/product_info.php?products_id=885
From its description, it looks like the ideal appliance to set up some
double play ISP data / telephony offer and I was wondering if anybody
was using it and what it was worth.
Hi All,
Asterisk is Up and running. I want to access this PC over internet. So
I registered at www.dyndns.com for dynamic IP-address mapping. I had
enabled the IP-forwarding (HTTP port 80) on the DSL Modem to point to
the PC running asterisk.
When I access from internet, I see the configuration
hisax seems to be a loadable module for an ISDN card. if:
# lsmod | grep hisax
prints any output, try
# rmmod hisax; modprobe zaptel
?
hth
Mojo
Shawn Porter wrote:
I am getting quite frustrated today, so please bear with me.
I just installed Fedora Core 4 (was running RedHat 9 with a
You probably need to turn off some sort of remote administration option
on your dsl modem, or change it to a port other than 80 if it allows.
Mojo
Vamsi Pottangi wrote:
Hi All,
Asterisk is Up and running. I want to access this PC over internet. So I
registered at www.dyndns.com
To clarify, if the modem doesn't allow anything but port 80 for its
administration and you can't turn it off, can you forward some other
exterior port (90?) to interior port 80 on the * box?
good luck!
Mojo with Horan Company, LLC wrote:
You probably need to turn off some sort of remote
In article [EMAIL PROTECTED],
Michael George [EMAIL PROTECTED] wrote:
Hello, all!
I'm looking at the wiki page and info on the mailing list and I'm getting
conflicting info...
I am using the manager API from the telnet CLI and I am testing creating calls
with it. I login with events: on
I have the same problem.
I've been having a bit of trouble getting the cards to work with
asterisk, and I thought perhaps you might know what I might be doing
wrong. I installed them in a linux box, and when I check to see if the
OS has recognized them it looks fine:
They show up as HSP56
How do I disable chan_skinny and chan_oss?
I think chan_skinny is associated with Cisco hardware, since I don't
have any I don't need this channel.
I just want to get rid of those warning messages at start up.
--
#Joseph
___
--Bandwidth and
On Tue, 13 Sep 2005 15:21:02 -0400, Shawn Porter wrote
I am getting quite frustrated today, so please bear with
me.
I just installed Fedora Core 4 (was running RedHat 9 with a
working
Asterisk)
now my Fedora does not appear to be recognizing my X100P (clone) at
all.
Hardware
Howard Leadmon wrote:
Hello all,
I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow
something is not quite right with my vMail setup. I would have sworn this was
all working, but maybe I was just dreaming.
Anyway here is what is happening, say I am on extension
My TDM400 on fc4 was working great..
all of sudden ..i am having the same issue ..you guys are having
all i tried to run asterisk as non-root user.. and I was able to run it as
non-root
and was able to receive and send call using asterisk..
I am not sure.. what thing I did wrong and coz all the
I have to design a dialplan for mulitple contexts (multiple companies)
and I'm not sure how to go about it and I thought someone may offer
help. Here is some background. There are three separate companies,
let's say A, B and C. Each has their own context and each has their own
set of numbers
Hi all,
Somebody already carried through the integration enters the DAC Nortel S1 with ITG and Asterisk?
Regards.
PJSantos
Yahoo! Messenger com voz: PROMOÇÃO VOCÊ PODE LEVAR UMA VIAGEM NA CONVERSA. Participe!___
--Bandwidth and Colocation
Hi,
We are terminating around 60 channels on one of our Asterisk boxes,
which the client sends in H323 mode.
Client (MERA) -- H323 -- Asterisk -- IAX -- Asterisk
The problem we face is that at random intervals the H323 process (as part
of Asterisk) dies and can no longer accept new calls
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
Any ideas?
Thanks,
Andy
Hi list, I'm hoping that I'm being stupid, and someone can tell me
what's going on, but for the life of me I can't figure it out. (it's
been a long day, and I'm now in the last 3 weeks of organising my
wedding, so I hope this makes sense ;) )
When at my desk, accessing (for example) my
Same problem here.
./asterisk stop;./zaptel restart;./asterisk start
seems to get it working.
Question: will ztcfg -vv alone get it working?
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen.
hi,
modules.conf
noload = chan_oss.so
noload = chan_skinny.so
On Tue, 2005-09-13 at 14:15 -0600, Joseph wrote:
How do I disable chan_skinny and chan_oss?
I think chan_skinny is associated with Cisco hardware, since I don't
have any I don't need this channel.
I just want to get rid of
Hello!
Anybody has tested the MTA-V102 with asterisk?
Thanks.
JS.
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To
List users,
It's been a while since I've posted here, but I've been hard at work
pushing toward our large scale Asterisk goal and keeping up with this
list can be a full time job by itself (I have19,543 unread list messages!!).
This Friday, September 16th 2005, my team will be at the MCI
Title: First PRI Installed - WOOT
Today I got my first PRI installed. It literally took less than 5 minutes and the circuit was up and we were making calls. The T100P is performing excellent. The Linux/Asterisk box is running well and the quality is great. The line is from MCI and they did a
PSI System Admin-Message-ID: [EMAIL PROTECTED]
Hi list members,
I'm sure this question has been posted before but I am still unable to find
the answer. I have a TDM 400P line card and I would like to set it up to
IGNORE the distinctive ring pattern that I have for a fax machine.
Many thanks
Andy Howell wrote:
I have a weird problem in which my digium card stops answering. After
running for a couple days, incoming calls are not seen. Running asterisk
-r shows no incoming calls. Restarting Asterisk does not help. After a
reboot it is fine.
This problem was fixed in CVS (HEAD and
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