-in SIP
client works.
Link: http://reviews.cnet.com/4520-6454_7-6358771-1.html?tag=txt
Thanks,
El Flynn
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Crazy Boy wrote:
Hi,
Here I am posting my problem. I am getting this problem since 8 days. I have studied documentation and looked previous posts in forums. But, I am unable to solve this problem. Please show me a solution. I am from India.
We have installed Asterisk with Digium 04B card
there might be something to do with your php.ini settings. The LMS
application requires you to have the Zlib extensions installed.
Best regards,
El Flynn
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Barry Fawthrop wrote:
Tried to install and get this
extension_dir does not exists
/usr/lib/php/extensions/no-debug-non-zts-20020429
when entering index.php
Barry,
Could you also check and make sure that PHP is running properly on the server?
Perhaps PHP is trying to load some extension,
Hello,
We've just released our Libero Management System application, a web-based
interface to configure and manage your Asterisk-based PABX. Designed for the
not-so-novice Asterisk administrator in mind.
LMS is simple to install, has minimal requirements (no external databases or
components
Alex Robar wrote:
Hi all,
I have a Sangoma A200 card with hardware echo cancellation. The card has 12
ports (10 of which are active; All FXO). Twice on this particular card I've
seen all ports simply stop receiving incoming calls. There is no other
indication of this, however. I am able to
voiplist wrote:
Anyone know how to keep an Analog Zap channel from answering?
I know I can answer it and send it to voicemail or do any number of
other things with it once it's answered.
I want to keep Asterisk from answering it, completely ignoring it
while still having the line connected for
Sharon Lim wrote:
Hi there,
I would like to ask, is it possible to group sip user? Means group A with
sip user 100,200 and group B with sip user 100,200?
thanks in advance.
in your dialplan, define the following variables:
GROUP_A=SIP/100SIP/200
GROUP_B=SIP/150SIP/200
and in your dial
Christopher Snell wrote:
Hi,
I posted earlier about Call parking breaks suddenly. I believe that
I have narrowed this down to a problem with DTMF detection and the
Sangoma A101 card that we use.
Earlier, DTMF detection was not working at all. Then, I set
'relaxdtmf=yes' in zapata.conf and it
Rizwan Hisham wrote:
Anybody who knows a good source of AGI tutorials on the net? plz share
How about the Asterisk Wiki?
Flynn
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bram kortleven wrote:
We are currently looking for a way to easily configure a 'auto attendant'
system on our asterisk pbx.
More in detail, I'm looking for a webbased (or something similar)
configuration generator, that has a feature like asking me how many 'menu
levels' I want, what text to
Rizwan Hisham wrote:
Hi,
i need to set the dtmf mode on my quintum tenor a400 gateway.
You might want to check the a400 manual on how to do that.
i cant dial
any extension thru my normal digital phone which is connected to asterisk
thru the quintum gateway. it always falls to 's' extension.
Jim Lynch wrote:
I attempted to install a new A200 module with one each FSX-2 and FXO-2
module. I connected an internal power connector to the board as
instructed, but when the system reboots, it just beeps at me. It
doesn't even let me get to a bios prompt. I removed both of the modules
M.Hockings wrote:
Even now, given that I don't know what caused the problem or what solved
the problem (for the time being). I might expect that powering the
system off may cause software errors due to partially written files but
I would NOT expect it to damage the hardware, particularly
Robert La Ferla wrote:
I want to create an extension say 8000 that prompts the user to enter
a number and then dial that entered number according to a set of
rules. The rules for dialing out are in different context (dial-
out-rules).
[some-context]
exten =
Michael Sampson wrote:
Basically I will have a call come in a PRI trunk and be routed out the
same PRI trunk. The point of this is so I can use asterisk to record the
call. Has anyone set up a system like this? I know how to get asterisk
to record a call from and extension, but not a call that
Khaled Chehab wrote:
I am using trixbox,I want ot disable and enable voicemail from command line
At [EMAIL PROTECTED] v 2.8 I was using this command and was working successfully
snip
But at trixbox its not working
Any ideas pleas
Did you try checking with the people who _wrote_
chan (Alpha Trilogies Networks) wrote:
Hi,
Does some one experience the Sangoma A20X-ec series card that cant detect
the hangup tone?
snip
[channels]
context = from-pstn3
switchtype = national
usecallerid=yes
hidecallerid=no
transfer=yes
echocancel = yes
echocancelwhenbridged = yes
M.Hockings wrote:
snip
Today the power went
out due to a mis-configuration on my part the UPS shut down before the
machine shut down. Now, I would not think this should be a problem but
the Digium card no longer responds. lspci does not show it either so I
presume it dead
While I don't
Douglas Garstang wrote:
General question.
If you install a Digium card in an Asterisk system, and install zaptel drivers,
do this give any benefit of echo cancellation? Our PSTN gateway is a separate
Audiocodes box, so the zaptel card wouldn't actually be connected to anything.
I'm wondering
Christopher Aloi wrote:
Hello -
I currently have 10 DID's coming into one Asterisk server, I seem to be
having some difficulty routing based on the DID dialed and am hoping
someone
on the list can assist me.
snip
Unless I'm misunderstanding you, how about trying this:
1. In your sip.conf:
Andrew Nowrot wrote:
Hi
I have a problem with Dial application. The dialplan looks like this:
;
exten = x,1,Dial(Sip/|30|L(6:3:1))
exten = x,2,Hangup()
exten = h,1,DadAGI()
;
The call is limited to 60 sec and after that time the conversation stops,
but
Khaled Chehab wrote:
How can I limit extension voicemail messages to 10 messages per user ?
If you look in the voicemail.conf.sample file in the source, you can find the
following lines:
; Maximum number of messages per folder. If not specified, a default value
; (100) is used.
Curt Shaffer wrote:
snip
the iax.conf config but the sound is ridiculous. The echo is horrible and
there is a screeching in the background on the receive end. Is there anyone
snip
You could also discourage your pet parrot from playing around with the phone...
haha its been a long day...
Jerry Geis wrote:
Hi,
There seems to be no solution for call progress on analog lines
and using outgoing spool call files . My wave file starts playing before
the person has answered the phone so the first part of the message is
missed.
Can the backgrounddetect app be used for this. I have
BILL GITONGA wrote:
I have two asterisk systems that share voicemail on an
NFS. I recently upgraded to Asterisk 1.2.9.1.
After the upgrade, the voicemail gets cut out after
about 5 seconds of recording. Any ideas on what might
be causing this?
What does it show on the CLI when this
Anderson Ling wrote:
Hi all,
I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in
my office. the out going calls symptom like when called party pickup the
phone but the calling party still hearing the ring tone from the IP phone.
Please light me up. it been many sleepless
Hello,
Does anyone know the maximum number of queues that can be defined in an Asterisk
system?
Thanks
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[EMAIL PROTECTED] wrote:
We have some laptop soundcards that are really bad and I would be glad
if you could share your experiences when changing to a USB headset
instead of using the built in soundcard in your computer.
Well, IMO if the soundcards are already crap to start out with, there's
[EMAIL PROTECTED] wrote:
I don't quite follow you? There are USB headsets that don't require a soundcard
at all. They have a built in soundcard which (I suppose) could be better than
the crap they build into most laptops.
well, slap me around and call my silly :) I haven't ever used one of
Sharon Lim wrote:
Are you looking for an web interface that write to asterisk config
files? if
yes, you can look at freepbx.org .
Hello,
just out of curiosity -- are you based in Malaysia?
Cheers,
Flynn
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chan (Alpha Trilogies Networks) wrote:
Hi,
Did someone experience that Asterisk OS 1.2.5 voicemail issues?
Problem description:
Some one call to the extensions 200,
After 10 sec ring then go to voicemail [EMAIL PROTECTED]
Announcement Please leave me a messages.blar blar..
When I completed
Kyle Sexton wrote:
TE410P:
- zttest will never report 100% for me across different motherboards
(Supermicro P8SCT, Dell 850)
- Crash/instability of about once per two weeks where I have to power cycle
the server, i.e. phone calls stop working and a reboot fixes it
TE406P:
- zttest runs
John Novack wrote:
Wavepad works well, without complaining about libraries, and you can
even edit. listen to the results and back out,if need be.
Harder to use for those who aren't sighted, though
John Novack
You could also use Audacity, which has a bunch of filters and effects that you
Hi,
I'm running two boxes side by side, identical specs and setup but with differing
dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same
folder for voicemail, exported via NFS from another file server.
Everything was working fine for an extended period of time,
Douglas Garstang wrote:
I just started poking around with writing a python module to interface to the
Manager API, and it suddenly hit me... how the heck are you supposed to program
this thing?
All the events seem to be dumped to all the open connections. If I send a command, such as a
Douglas Garstang wrote:
Yikes. Java. Yuck. I'll stick with Python... Thanks anyway.
I just worked it out... you can supply an actionid to the request to know what
reply to look for, although it will still be tricky filtering out the noise.
Well, with the Asterisk java code it's pretty much
Anton Krall wrote:
Guys, anybody has some info regarding the format that queue_log has and how
to interpret it.. I found some info on the wiki about the conditions of a
call but the first fields I still don't know what they are for, although I
can imagine one of them is a call identifier, etc.
Hi Sig,
I'm trying to compile the assman package, but some errors come up:
dceptcons:/usr/local/src/libassman-current # make
make -C libassman
make[1]: Entering directory `/usr/local/src/libassman-current/libassman'
cc -I../inc -Wall -c -o assman.o assman.c
In file included from assman.c:8:
Sig Lange wrote:
I am currently developing a asterisk ncurses interface using the manager
API. The project is currently awaiting sourceforge's approval but I have a
beta online at http://sig.lange.googlepages.com/assman . The projects real
home will be assman.sf.net. This project really consists
Alexander Lopez wrote:
That may be the best one yet, It is pulling the information out of
Asterisk's BackEnd.
:-)
From the looks of the project's screenshots, assman needs to be able to handle
a lot of shit coming out of the back end, for cases when a busy server is
generating a lot of
Jerry Geis wrote:
I was searching on voip-info.org for saydigits.
I see no indication it is not valid in 1.2.4 asterisk.
however, when trying to use it I get and error no application saydigits.
what is the correct way to echo back digits in asterisk 1.2.4?
I tried say digits 123 and saydigits
Mike Clark wrote:
snip
Have you called Sangoma's tech support number?
I just implemented the same card about two weeks ago and really didn't
have any installation issues using fc3 and trunk, however their
documentation is a little on the rough side. Install info seems to be
a little in one
Michael Kenjie Nukui wrote:
Hello,
i am trying to install sangoma a200d to my centOS server but i am receivig
this error message:
ZT_CHANCONFIG falied on channel 1: invalid argument (22)
How is your hardware set up? Do you have just the one A200 board, or do you have
additional Remoras
Wolfgang Borgon wrote:
A RAW file I created after converting from MP3 and WAV, sounded raspy.
Does anyone have any tips for creating the best quality voice recordings?
Generally you'd use a good-quality microphone for your recordings. The adage
Garbage in = garbage out couldn't be
Hi there,
I've got a client complaining about the dispositions in the CDR report we built
for them:
1. User calls an extension, which rings three SIP phones in the group. Entry in
extensions.conf:
exten = 100,1,Dial(SIP/200SIP/201SIP/202)
2. On three test calls, she dials extension 100
Kong wrote:
can i know where to start? SIP is such a big topic.
Try looking for SIP configuration (sip.conf) in the Wiki, it's got lots of
examples. Or you can also try looking it up on google.
Flynn
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Andy Goss wrote:
Oct 11 20:32:58 WARNING[3048]: file.c:415 ast_filehelper: Unable to open
fd on /var/spool/asterisk/voicemail/default/5926/INBOX/msg.wav
Oct 11 20:32:58 WARNING[3048]: file.c:793 ast_streamfile: Unable to open
/var/spool/asterisk/voicemail/default/5926/INBOX/msg (format
Rich Adamson wrote:
snip
One other item to check is to ensure the digium T1 card is on its own
dedicated interrupt. Use 'cat /proc/interrupts' from the system command
line.
It is on one interrupt, first thing I checked when the problem cropped up. One
thing I did notice was interrupt
Rich Adamson wrote:
It is on one interrupt, first thing I checked when the problem cropped up. One
thing I did notice was interrupt latency when doing a 'lspci -v'.. should that
number be 0? If so, does anyone know how to set that at boot time?
I played around a fair amount with the
oner asterisk wrote:
Hi all,
I would like to add indication tones ,
What I did is
enter data to zonedata.c and indications.conf
than compile zaptel. and restart asterisk.
But it's not working what else I should do ?
Regards,
Öner
did you check that the new tones are loaded in
Hi all,
I've got a TE410p connected via T1 to four channel banks (2 FXO and 2 FXS), no
PRIs.
Some users are complaining that they hear clicks and pops on the FXS lines,
generally when they pick up the phone it's noisy. This happens only after a
while, e.g. after a fresh restart of
taran wrote:
i have one extension going straight to voicemail, while others that are
configured identically don't, so i don't think it's an overall config
problem. nor do i think it's a callerID problem. maybe it's an enduser
operation that i can't find documentation on?
snip
it would be
housi mueller wrote:
I am new to asterisk and would like to know if a configuration like shown on
the picture with asterisk is correct?
Thank you in advace
Housi Mueller
Looks good
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Wolfgang Pichler wrote:
Hi all,
at time i am trying to get a better idea of callgroups and pickupgroups
(especially within the SIP Channel)
A Pickupgroup is relative clear - everyone in the same pickupgroup may
pickup a call
And a callgroup does what ? - The same ?
example:
phones A, B
Malcolm Taylor wrote:
Can anyone point me in the direction of a utility which will let me
determine the length (in seconds) of a wav49 file created by Asterisk?
Many thanks,
Malcolm
if you're talking about the duration of a voicemail, you could do:
grep duration msg.txt
Michele O-Zone Pinassi wrote:
I've upgraded Asterisk from CVS, spandsp and app_txfax and app_rxfax but i'm
still unable to send/receive faxes :-(. I'm using amp_fax to send and this is
what i get from logs:
snip
Sep 6 11:06:13 VERBOSE[10750]: -- Executing System(Zap/1-1, tiff2ps
-2eaz
Rod Bacon wrote:
I am wanting to front-end a legacy PBX with an asterisk box. I have done
plenty of asterisk work over the last 6 months to PRI circuits, but not
with a PBX being involved.
I know I can use asterisk and digium cards in this manner, but do I need
separate cards for the PRI -
Stephen wrote:
Hi All,
I have configure my Asterisk as follow (using [EMAIL PROTECTED]):
[zaptel.conf]
span=1,1,0,ccs,hdb3,yellow
bchan=1-15,17-31
dchan=16
loadzone = uk
defaultzone=uk
try this in your zaptel.conf:
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
CRC required for
Joseph wrote:
Is it possible to do nested dial() command on one line,
Dial number, wait new seconds, dial another number etc.
or dial number and jump to another line to continue dialing.
D(ww) doesn't work as it sends DTMF but before the call is bridged, and
I need to send numbers after the
Joseph wrote:
Here is a session with D()
exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(ww218))
Executing Dial(SIP/11-3dec, SIP/[EMAIL PROTECTED]|30|D(ww218)) in new
stack
-- Called [EMAIL PROTECTED]
-- SIP/pstn-5665-713c is ringing
-- SIP/pstn-5665-713c answered
Hi all,
On the Wiki it says something about the motorola WiFi/GSM hybrid phone, the
Motorola CN620. Don't know whether that one ever made it to the market or not,
but I read a review on c|net about another upcoming model, the A910.
The A910 is Linux-based, and offers WiFi on top of GSM, GPRS
Ronald_Wiplinger wrote:
Brian West wrote:
If you use mp3nb from the sample configs you will have exactly 1 per
class.
Great!
Where can I read more details about it?
(musiconhold.conf)
in musiconhold.conf:
[classes]
default = mp3nb:/var/lib/asterisk/mohmp3
Flynn
peiyin wrote:
Dear all,
I want to create a php web front end to disconnect a SIP call (from a
particular sip phone) which is in progress. Any ideas how to do so?
Google for Flash Operator Panel. Or look in the Asterisk wiki for it.
Flynn
___
Noah Miller wrote:
snip
In addition to largely being a rehash of existing docs on the internet,
there are many editorial errors in the version that I have. Before I
was comfortable with the conf files, these editorial errors were very
confusing. The editions coming out now may have fixed
Bryce Chidester wrote:
Assuming you mean you have 30 analog POTS lines, the way to go about
this would be with a couple channel banks and a quad-T1 (I haven't seen
a two-port around, but that's all that is needed) card.
For the record, 30 individual analog lines is generally inefficient and
Dominique Kull wrote:
_Description_
We are looking for an expert LAMP (Linux, Apache, MySQL, Perl, and PHP)
developer
snip
I will coin a new phrase for this list:
LAMPEPA developer - a developer of solutions based on Linux, Apache, MySQL,
Perl/PHP and Asterisk
haha
Dave Morrow wrote:
Hi all. I have been using Asterisk for sometime now and have recently come across AMP
for the first time. I am wondering if someone could enlighten me a little as to the
advantages and disadvantages to using AMP as opposed to the do-it-yourself
Asterisk? Is this
Richard Cook wrote:
Hello,
Has anyone had issues with faxes showing up squished in the TIFF file?
Any ideas what could be causing it?
We had some issues while getting fax-email and email-fax working. As far as I
can tell, it ended up being a wonky version of libtiff that was causing
dave cantera wrote:
hi,
is there anything going with VoiceXML in asterisk??? is this the list
to query regarding this or should I put this on the dev list?
thanks,
dave cantera
I don't think there's anything built-in to support VoiceXML, but you _can_ do
something like this:
1. get a
Libel Lawyer wrote:
This is the guy that has a ton of email addresses.
Almost as many as he has phone numbers.
google kvj
He doesn't like our president either:
Here's look at a MISERABLE FAILURE and I use facts:
garbage snipped
Er.. did you type in the wrong email address in the To: field?
Darren Wiebe wrote:
Good Day,
I'm finally getting around to officially announcing ASTPP. For the last
6 months, I've been working on converting ASTCC into a decent billing
package for asterisk.
The link in the original email opens a page that says
Download the latest version of the code
Anton Krall wrote:
Why disregard from MX? :)
You might want to check the archives, or Google for Vonage staff arrested in
Mexico, or something along those lines..
flynn
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Brett, Gary wrote:
Does anybody know of a WINDOWS application (preferably freeware) that will
simply playback asterisk GSM sound files, I don't want to record them, just
playback the ones that are currently there.
Any help would be greatly appreciated
You could try Audacity
Johann wrote:
What third party programs are available for parsing the queue_log file
and CDR file? I know about XC-AST, but management would prefer a php
based solution.
What have other admins done to retrieve detailed call information about
the queue system? Anyone develop their own that
Michael Jones wrote:
Hi All;
I'm a newbie so please be gentle.
I'm a new * user and am using it to control the 3 IP phones in my
house. I'm using asterisk because I enjoy the flexibility and I'm sort
of a tinkerer.
Here's my question: Everyone has used the dial by directory function
where
Mark Wormgoor wrote:
Hi,
Is it possible to set a variable for a context for all extensions? I
haven't been able to find it.
Try looking up the application SetVar:
demo*CLI show application SetVar
demo*CLI
-= Info about application 'SetVar' =-
[Synopsis]:
Set variable to value
[Description]:
snip
This is the sort of thing that AGI is great for. When I was first
starting with Asterisk I wrote an AGI script to ask the caller for their
Zip Code, then connect to weather.com, download the current weather
conditions for that zip code, massage the text, run it thru a text to
speech
El Flynn wrote:
snip
This is the sort of thing that AGI is great for. When I was first
starting with Asterisk I wrote an AGI script to ask the caller for
their Zip Code, then connect to weather.com, download the current
weather conditions for that zip code, massage the text, run it thru
Jim Lists wrote:
snip
I'm still left wondering if Asterisk supports multiple lines
at once? If I had one land line, voip line, and asterisk setup and 10
people called my number, would all 10 people be able to speak to their
appropriate party at the same time, or would the other 9 get a busy
Hello all,
For those of you who've attempted to use the Dialplanner, but could not receive
the exported dialplan, we sincerely apologize for the problem. There was an
internal misconfiguration on our mail server which stopped the dialplan from
being emailed.
We've since corrected the problem
Anton Krall wrote:
Guys.
I just had a weird problem. I have my Dial cmd configured with mwtWT as
parameters however, a call came in thru a zap channel and I answered on a
sip phone. I tried using # as configured on my features.conf file to
transfer the call but the transfer prompt never came in,
amna saleem wrote:
hi!
I was wondering if the i extension works ,i mean i have included
this in my extensions.conf ie
exten = i,1,Answer
exten = i,2,Playback(pbx-invalid)
exten = i,3,Hangup
but it doesn`t seem to work,i am getting no announcement when i dial
an invalid no. rather i get the invalid
James Bean wrote:
Hi,
Weird issue, 2 asterisk boxes running 1.0.7, when I call iax from box 1
to box 2 it works fine, when I dial from box 2 to box 1 I get a
On Box 1
Apr 11 17:25:39 WARNING[8969]: chan_iax2.c:5553 socket_read: Call
rejected by 192.168.69.1: No authority found
On Box 2
Apr 11
kritikus Araklidas wrote:
Hi:
Somebody know how to configure the Authentication cmd with DB (Mysql)
suport. its work with single password and password file, but i cannot
find information for use database in conjunction with DB.
Any help will be appreciated.
Unless I'm mistaken (haven't been
Matt Roth wrote:
Preferably, I would like an out-of-the-box solution, but custom-coding
is an option as long as the necessary data is available from Asterisk.
If anyone could point me in the right direction, it would be greatly
appreciated.
You're right in that most of the things you're asking
Rich Adamson wrote:
No, that's a service, or at least I think it is, the sales garbage obscures
what it really is so who knows.
What I need is a little box that diverts calls if the PBX goes down.
FYI, the topic has been discussed previously on the list, and the
problem that you're trying to
Kong wrote:
Is there any application that actually work like Background, but instead
of playing a specified file, it plays the streaming music from music on
hold?
the reason i am asking this because i come across a dialplan that goes
this way,
if a person gets to an extension that is busy, it
Hi all,
Just wondering if anyone's come across this issue, and what might be a fix for
it:
We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The
phone can do proper supervised transfer, but _only_ once. If the user attempts
to transfer a second time, it won't work.
any
Hi,
Is anyone aware of an IAX client that's made for the Windows CE/Pocket PC
platform? Or even the Palm platform for that matter.
Thanks.
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Matthew Boehm wrote:
From macintouch.com:
Apple is distributing an open-source Asterisk install package for Mac OS X:
A complete IP-PBX in software.
SNIP
If anyone's interested, Benjamin Kowarsch from Sunrise Telephone systems Ltd is
doing that. Check it out at http://www.sunrise-tel.com
You
Mateo Meier wrote:
What do you mean ?
My etc/asterisk/musiconhold.conf looks like that:
[EMAIL PROTECTED] root]# more /etc/asterisk/musiconhold.conf
;
; Music on hold class definitions
;
[classes]
default = quietmp3:/var/lib/asterisk/mohmp3
;loud = mp3:/var/lib/asterisk/mohmp3
;random =
Hi all,
I'm just wondering about these VoIP services -- do you have to sign up one
account -per- client that will be using the service? I've got multiple
extensions behind my Asterisk box, and I want to be able to allow all my staff
to place calls via the provider.
So if I sign up for one
Jay Wilton wrote:
Hello,
Is it possible to use a timeout in a queue and have the
option of pressing a button to remain on hold? I have been
using:
[qbert]
1,1,Queue(qsales|t|||180)
1,2,Voicemail(u22)
[qout-sales] ;dtmf-out context from queues.conf /[qbert]
*,1,goto(qbert|1|1)
Problem - I
[EMAIL PROTECTED] wrote:
Hello,
I have looking into the TDM series of wildcards.
All these card are for linux kernel 2.4.
If I were to use FC3 which is based on kernel 2.6, will
I have any compatibility issues.
Thanks
I'm not sure about Fedora, but we're running SuSE 9.1 with the 2.6
Altus Snyman wrote:
Good day all
I downloaded bristuff RFC3 and asterisk,zaptel,libpri versions 1.0.3
This is to install my quad bri card
All installed well
I coped over some old config files.All 4 ports are available,so that
gives 8 open lines for incoming or outgoing,correct me of I'm rond
The
Edgar de Leon wrote:
Hello, i got a question,
i need to create a group extension, to make calls to 6 sw phones, but i
need to know if asterisk can do help me to get a unique number and check
what extension has received less calls than the others, and pass the new
call. We got a call center and
jurgen wrote:
snip
Problem is, Asterisk times out and disconnects after 10 seconds,
stopping the recording.
If I run something else in the context, say the infamous Monkey
Sounds, everything's fine, and the call just keeps going, annoying the
people on the line with monkey sounds. For some reason,
Jason Brown wrote:
So I have a problem. A customer of mine wants a PBX, owns an office
building. I want to sell him on asterisk. He has 4 tenants. I am using
my asterisk box to simulate it. My asterisk box has a TDM400P card, not
a PRI card. Don't know if it makes any difference.
snip
Just a
Jon Gabrielson wrote:
When there are no zap channels available, I signal congestion
using the following:
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Playtones(congestion)
exten = _9NXX,3,Congestion
The congestion sound plays correctly, but the ringing continues
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