Hi all,
I just installed openSUSE 10.0 on a spare machine to
try and do some development work. I did a checkout on
libpri, zaptel, and asterisk and everything compiled and
installed perfectly. My issue is with the zaptel script
placed in the rc.d directory to automatically initializ
the
On 15 Mar 2006, at 14:36, Stojan Sljivic - GDS wrote:
Hi,
I have downloaded an IAX softphone and tested the connection locally.
The sound is perfect.
How should I troubleshoot this IAX connection between these two
Asterisk
servers?
Is there some tool that can help in determining the cause
IIRC, it's something that is supported in the latest versions of SIP,
which Asterisk doesn't support yet.
On 3/15/06, Noah Miller [EMAIL PROTECTED] wrote:
Hi Giorgio -
we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to
show the called name on the calling polycom display
This is a function of the Phone itself. Asterisk has nothing to do with
it as it does not know anything about the call until after the SIP
device 'sends' it.
To my knowledge it is not posible. I don't even think a SIP standard is
available for this.
This 'feature' along with changing CallerID
The more I think about it, the crazier this supposed requirement sounds to me..
It would seem that any IVR for a bank, which allows account access 24/7/365
would be in violation, since we all know banks have no humans in them
off-hours... Many businesses have IVR type systems that operate
We are runnign into problems where our legacy PBX reaches a frame loss threshold and takes it's T1 card offline (the T1 card that interfaces with the Asterisk servers TE110P). During this time, the Asterisk server senses a Yellow alarm.
We've noticed that if we quit asterisk, stop zaptel, start
On Mar 15, 2006, at 4:20 AM, nik600 wrote:
hi
in my callcenter i start asterisk on server with asterisk_safe
command, after 4 days i can see that it is crashed 12 times, reporting
segmentation fault error...each time asterisk is correctly restarted
without loss of services but, is it normal?
The phone must have transported you to Australia... :)
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 10:05 AM
To: asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Double-ring tone
I upgraded my
On Mar 15, 2006, at 6:36 AM, Stojan Sljivic - GDS wrote:
Hi,
I have downloaded an IAX softphone and tested the connection locally.
The sound is perfect.
How should I troubleshoot this IAX connection between these two
Asterisk
servers?
Is there some tool that can help in determining the
That does not work with svn-trunk-r10536 from Feb 20th. Command is
accepted by asterisk, but nothing happens. The sip_notify.conf does
include the entry.
Aaron Daniel wrote:
In the sip_notify.conf file, there's a couple different events that will
cause different phones to reboot. One of them
On 3/15/06, Alexander Lopez [EMAIL PROTECTED] wrote:
This is a function of the Phone itself. Asterisk has nothing to do with
it as it does not know anything about the call until after the SIP
device 'sends' it.
To my knowledge it is not posible. I don't even think a SIP standard is
I'm using frequently the perl api within asterisk.
Now I'm looking for documentation for the perl commands.
Some perl commands I found on this URL:
http://www.voip-info.org/wiki/view/Asterisk+PHP
Does anybody got more documentation or where I can found some more
documentation about perl
You could use Asterisk to flip the caller id and dnis variables, before you
Dial the phone. :)
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Title: Re: [Asterisk-Users] OSHA requirement to "reach a live human" ??
I think the original poster has realized that he now has
ONE leg LONGER than the OTHER.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
JonesSent: Wednesday, March 15, 2006 12:28 PMTo:
Although this is a T1, this is NOT running a PRI. It is running with
2 DSS voice trunks using fxs groundstart. Is that not possible with this card?
At 08:53 PM 3/14/2006, chan \(Alpha Trilogies Networks\) wrote:
Please check your zaptel,
TE110P is the T1/E1, you should state the parameter
True. And why would OSHA differentiate between an IVR and a traditional
answering machine? Likewise, what about the other alternative - no
answer at all - how is this better than an IVR from an OSHA point of
view? Clearly people are allowed to have answering machines and clearly
not every
Not sure it's that weird :O
Douglas Garstang wrote:
The phone must have transported you to Australia... :)
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 10:05 AM
To: asterisk Users Mailing List - Non-Commercial Discussion
On 3/15/06, Geoff Manning [EMAIL PROTECTED] wrote:
We are runnign into problems where our legacy PBX reaches a frame loss
threshold and takes it's T1 card offline (the T1 card that interfaces with
the Asterisk servers TE110P). During this time, the Asterisk server senses a
Yellow alarm.
I will test it, However it is still PolyCom Specific.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nathan Bowyer
Sent: Wednesday, March 15, 2006 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
in my callcenter i start asterisk on server with asterisk_safe
command, after 4 days i can see that it is crashed 12 times, reporting
segmentation fault error...each time asterisk is correctly restarted
without loss of services but, is it normal?
No. I am only a small experimental home
That's good to know Francdois,
I was not aware of that. I talked to a Sangoma tech about that possibility
and they told me that although their hardware echo can is just an added
module, there are some minor differences on the board it mounts to as well
so it's not a possibility to upgrade a
I have a custom sip reboot message I am transmitting to a Snom 200 to reboot
it. The Snom gets it, it says 200 OK but it doesn't reboot. I have turned
off the challenge-reboot option in the Snom. When I modify the Event:
directive from reboot to snom-reboot the phone yields Bad Event and
when it
Best way is to have a PRI interface to your PBX, I don't have any experience with NEC, but with our nortel system this is what we did. You program your PBX to send extension 123 out the PRI, asterisk sees the call and routes it accordingly.
On 3/15/06, John Padovano [EMAIL PROTECTED] wrote:
I've been asked to look at a tender for a switch, and one of the
capabilities the customer is looking for is support for Q.932. They
have a number of exchanges and are looking, in the future, to support
things like remote and aggregated operators.
Can anyone point me to an idiot's guide to Q.932
I am trying to use ChanIsAvail to detect the best route for a call. I am
testing by dialing an extension that is then forwarded to the DID.
Normally it will be an incoming PSTN call that is forwarded.
When I try it, I get put on hold for a few seconds and miss the
beginning of the recorded
Forgot on the Snom 200 it won't reboot if under the Memory tab in the web
interface, Connections 0 then remote reboot is not possible. Manually
cycling the power allows the phone to be rebooted by Sipsak remotely.
HOWTO: Reboot a Snom with Sipsak
Checklist:
1. Under Advanced in the web
Why? If you flip the callerid and dnis variables, it should work with any phone.
-Original Message-
From: Alexander Lopez [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 11:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] how to
This is a function of the Phone itself. Asterisk has nothing to do with
it as it does not know anything about the call until after the SIP
device 'sends' it.
This is not just a function of the phone. The phone has no idea what the
caller id of the receiving end of the call will be.
Im developing an ip telephony project and i need some help in order to
choose the better PCI card, the options at the moment are digium, sangoma
and voicetronix, the strongest ones are digium and sangoma but i dont know
how justify the election
If you want people to answer, maybe you should be
When a friend calls, I would like for him to enter a 4 digit password
in order to access to a sub-menu, if no password is entered, then the
welcome msg is said ...
Any hints on how to do that ??
In your incoming-rtc context, define an extension (let's say 1234)
exten =
I'm using this patches for TE406 to SIP ATA but it didn't work???
2006/3/15, Paolo Prandini [EMAIL PROTECTED]:
I found a bug in the latest T38 passthrough patches, the effectis that a non-SIP call after being put on hold is then lost, no
resume is possible.The fix is to be applied in the
Hi
I have a project where I need to connect an
asterisk server to an avaya definity prologix PBX in order to add ip telephony
to a organization, actually there's no E1 ports available
forconnections,I want to use a 2 port card (still dont
knowwichdigium or sangoma), thenone E1 cable will
That's probably what is happening on my end. Any suggestions on how to
fix this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron
Daniel
Sent: Tuesday, March 14, 2006 7:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I have several servers using them, but I only needed to download them
directly from you just once. I replicated the bits myself. The magic of
these advanced technologies... ;)
I could go download them a few more times if it would make you feel better.
All kidding aside, I don't think I ever
Thanks for all the replies. Funny, Alexander! It's not for OSHA; it's
for a private company. The only thing that made me wonder was that it is
a system to be used by their *employees* so it might fall under
different rules than a system for customers, and would be more in OSHA's
purview.
I've beaten myself bloody dealing with this one... No luck so far. In
summary, incoming calls from Gizmo establish, but neither get nor send
sound. Outbound calls to Gizmo work fine (well a bit choppy but work)
My thought is that the SIP connection is being made fine, but the RTP
is getting
Okay, so as it turns out, this is *sort* of working. If I call from Extension
A to Extension B, and B places A on hold, A hears the hold music. However, if
A places B on hold, B does not hear the hold music. Is there a way to specify
the hold music if the originating party places a call on
Sorry, send this part from an unregistered account
I know this is going to a duh statement to a lot of people, but just
in case... when the non-audio gizmo connection rolls to voicemail, on
the cli I get:
app.c:645 ast_play_and_record: No audio available on
Hi, to all,
i am new in the list and i am interest to deploy a sistem with asterisk i
have a PC with a Suse Linux 8.2 and also i have Dialogic VFX card with 4
analog port.
From where a can get Dialogic Driver for linux.
From ware a mast beging to resolve the problem the project to implement
In some work places the term 'live-person' may not require a warm body.
Just reference some of the Dilbert comic strips! :-)
Snip
P.S. the ironic part is they want me to just allow the user
to record a question/comment to satisfy the supposed reach a
live human
requirement, which
Forgive me if this question has been asked/answered in another post.
And let me reiterate what other users have frequently said - Asterisk
is
great, and I really appreciate all the work you folks have put into
it.
How have some of you gone about integrating Asterisk with a legacy
office
Hi
Im also new but you should know very well all the interfaces you are going to
connect the sistem, the number of users you'll have (hardware requeriments),
know a lot about the soft/hardphones you'll use and download the asterisk
handbook or the big one (i don't remember the name)
Good
Aren't you bothered by the fact that the sound file quality goes up and down as
different sound files are played? It's quite obvious to hear the difference
between a ulaw file and a gsm file.
-Original Message-
From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15,
Hi,
I think there should be only one timing source, but you have 3 here...
Zaptel.conf
span=1,1,0,ccs,hdb3,crc4
span=2,0,0,ccs,hdb3,crc4
span=3,0,0,ccs,hdb3,crc4
span=4,0,0,ccs,hdb3,crc4
Not sure if this is causing the problem though.
Andy
On 3/15/06, James Sturges [EMAIL PROTECTED]
On 3/15/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Why? If you flip the callerid and dnis variables, it should work with any
phone.
How so? the caller will still only see the extension. Please get sober
before you repeat it again.
-Original Message-
From: Alexander Lopez
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
zttool always shows
Sync Source: Internally clocked
surely this cannot be correct.
___
--Bandwidth and
What about telnet and the reset command?
On 3/15/06, Rich Adamson [EMAIL PROTECTED] wrote:
That does not work with svn-trunk-r10536 from Feb 20th. Command is
accepted by asterisk, but nothing happens. The sip_notify.conf does
include the entry.
Aaron Daniel wrote:
In the sip_notify.conf
I've got exactly the same problem with echo, where the mic feeds into the
speakers. I'm looking at purchasing the Tellabs 2572 64ms T1 echo
cancellation card to see if it will help.
Anyone have any experience with this hardware and how it deals with echo?
I've read on the wiki that it's supposed
Douglas Garstang wrote:
Aren't you bothered by the fact that the sound file quality goes up
and down as different sound files are played? It's quite obvious to
hear the difference between a ulaw file and a gsm file.
Douglas,
I know that you have had a hard time grasping this, but not EVERY
Hi All,
I need to use - mms://61.112.173.60:81/ as souce for MOH, i cant find
anything about using that souce format in wiki.
If you have some info please advice.
Miklos
IPFONE TELEFONIA IP
Rua Caio Graco 735 São Paulo SP
IPBX - +55 11 3488-3800
http://www.ipfone.com.br
[EMAIL
I run Asterisk partnered up to the 280 (424 for us). We have a 6 cabinet
installation of the Toshiba so I understand you dilemma. There are some
quirks with the 280 that make it a challenge to use Asterisk with, but
it's do-able. Keep in mind, you can move to the CTX or the CIX and still
keep
This is a known (but very old and fixed) bug. See
http://bugs.digium.com/view.php?id=4186
bails wrote:
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
zttool always shows
Sync Source: Internally
Lee,
Does (and always has) worked just peachy for me.
Jason
--
Message: 26
Date: Tue, 24 Jan 2006 14:26:55 -
From: Lee Archer [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk with SuSe 10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Aren't you bothered by the fact that the sound file quality goes up and down
as different sound files are played? It's quite obvious to hear the difference
between a ulaw file and a gsm file.
Aside from a few company specific greetings, the slinear package has all the
sound files we need.
-
Trying out SVN-oej-t38passthrough-r12677 on a server that also needs to
pass some calls to another using IAX and attempts to use the Dial
command results in multiple messages Out of idle IAX2 threads for I/O,
pausing!.
Since this server needs to support IAX I'll have to back out this
version
On Wednesday 15 March 2006 15:45, bails wrote:
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
zttool always shows
Sync Source: Internally clocked
surely this cannot be correct.
Is the LED green? If so, you should *not* be getting
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent from
softphones through Asterisk, and based on them, build the destination
PSTN
What ATA's are you using?
I've notice occassional occurance of the same messages, and they seem
to be comming from only certain type of ATA's.
I'm suspecting it's ATA related, but I don't have enough evidence to
prove so yet.
Andy
On 3/14/06, Dan Morin [EMAIL PROTECTED] wrote:
The past two
This is just an idea. I personally love the idea of lower cpu
utilization, even more so than better sound quality. So take all your
gsm files, and convert them to ulaw, alaw, g729, etc. Now, when
someome calls in they'll always get the same quality sound files
(i.e., crappy), but cpu usage will
On 3/15/06, Robert P. McKenzie [EMAIL PROTECTED] wrote:
A user of mine has discovered that when you call into asterisk and get the
IVR menu with options 1-5 available, if you
dial 1 then immediatly dial 2 it will connect you to 2 and not 1. I expect
this is due to the digit timeouts and
Charles Marcus wrote:
[...]
So, how much work are we talking about to get our current system to play
nice with Asterisk? Will we lose any functionality? Gain any? Do you
know of any technical how-to's that my phone guy would be able to answer
these questions from? Are you available to
In case this is useful to someone...
Initially running * 1.0.7 and the default canceller, about 1 in 20 E1
PRI calls still had echo, sometimes quite bad.
Updated to * 1.0.9, and backported KB1 from 1.2 HEAD to this build -
No noticable change over the previous version, but we ran with it
anyway
looks like your nmap only scanned for tcp connections. Try the -u switch.
netstat shows that udp 5060 is accepting connections.
Your iptables ruleset gives me a headache to look at and is quite
redundant. Wouldn't it be better to just disallow all packets at the
beginning and
then open the
I have an asterisk server that specific users have delete=yes set in the
voicemail.conf file. They are occasionally still recieving the
voicemails in their voicemail inbox as well as their email inbox. Is
this a known issue, if so is there any work around?
You can compile zaptel with a watchdog feature that will restart your
interfaces should they start to act up. Just edit the zconfig.h file
that comes with the zaptel source code and uncomment this line:
/* #define CONFIG_ZAPTEL_WATCHDOG */
___
exten = _3XX,1,Answer
exten = _3XX,2,Dial(Sip/${EXTEN},6000,t)
exten = _3XX,3,Hangup
Why do you Answer before you Dial here? I had a problem where calls were
misbehaving and someone asked me that same question. Without really
understanding why I removed the Answer and it then just worked.
I
At 12:30 AM 03/15/2006, you wrote:
wget http://server.name/path/to/file
wget ftp://server.name/path/to/file
In fact, what I normally do is copy a link from my browser to the
command line in the terminal window and download it with wget. Saves me
an extra file copy around the net.
I'll get
On 01:08, Wed 15 Mar 06, Faisal Inam wrote:
Hello all!
I want to assign one of the PSTN lines to a specific extension
only.
Expecting an earlier response.
Thanks a lot.
Faisal
assume the PSTN line has a number assigned by the telco (ex.
333)
Now in your
steve wrote:
looks like your nmap only scanned for tcp connections. Try the -u switch.
netstat shows that udp 5060 is accepting connections.
Your iptables ruleset gives me a headache to look at and is quite
redundant. Wouldn't it be better to just disallow all packets at the
beginning and
Hello,When I start Asterisk, I get the following in my log (/var/log/asterisk/full):Mar 15 17:16:55 VERBOSE[4242] logger.c: == Parsing '/etc/asterisk/zapata.conf': Mar 15 17:16:55 VERBOSE[4242] logger.c: == Parsing '/etc/asterisk/zapata.conf': FoundMar 15 17:16:55 DEBUG[4242] chan_zap.c:
Is it onerous to backport or is it a case of fiddling around with the
makefile? Care to post a backported tar?
-Original Message-
From: Steve Davies [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 2:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Echo
It has to do with transcoding. If Asterisk cannot 'speak' the codec it
cannot answer the call and is 'unable to forward the frame.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
James Harper
Sent: Wednesday, March 15, 2006 5:08 PM
To: Asterisk
Thanks for the input everyone.
I though the second digit in the span = was the timing attribute, so only
getting master timing on span 1.
We have had timing issues, can we confirm this?
Thanks again
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Have you tried with apic turned off? And, on another note, our system
had bad sound (you might describe it as choppy) with acpi enabled.
Do you have access to a milliwatt test line?
Moj
sdgesa gaeharth wrote:
thanks for the info.
it is not sharing an irq:
0: 59840409 59803082
(because it's the DISA application that gives you the dialtone you wanted.)
Alexander Lopez wrote:
You are on the correct path with immediate, and using the s extension.
Place the phone in a context that does the following:
Wait,1
Palyback(hello)
DISA(contezxt for outgoing calls)
This whould
I looked on the voip-info wiki and found sparse and conflicting information on how to do this with Asterisk...
My incoming lines are all on Zaptel. Is there a simple why to implement a '*363 (do not disturb) toggle via the dialplan?
It would be nice to be able to pick up an extension, dial *363,
Please Ignore if you cannot edit the code.
You will have to modify app_dial.c in apps directory.
Look for code that calls ast_dtmf_stream(chan, ..., timeout)
The last parameter is the inter digit timeout, it can be set to as low
as 1 (1 millisec) a value of 0 it will default to 100millisecs.
The
Mar 15 12:54:01
NOTICE[24269] channel.c: Dropping incompatible voice frame on Local/[removed
[EMAIL PROTECTED],2 of format ulaw since our native format has changed to
slin
Can anyone provide an English
translation of what this means?
The extension is a Polycom
IP 501
The only
Well, I got off site today with my notebook and an x-lite install. I
was able to connect into to the system and hear things, etc...
But since the phone connects ahead, this may be a different thing than
an incoming gizmo call eh?
If someone could even point me in the direction to look, I would
Douglas Garstang wrote:
Boy, am I stuck...
I'm officially ready to toss Asterisk out the window.
Yay!! Why don't you?
I have to admit it isn't necessarily all the fault of Asterisk either. It just
seems that every option I turn to suddenly ends in failure. I don't know if
it's me that's
Barry Flanagan wrote:
Hi,
We are trying to use attended transfer with Asterisk 1.2.5, but when we
do the transfer and dial the new number, it times out after 3 rings and
then the callee is put back to the original agent.
Where can I adjust the timeout which applies to the number we are
Steve Kennedy wrote:
OK, that explains it, just wondering why the sample extensions.conf
turns it off, while the O'Reilly Asterisk book and alomst everything you
see on the web uses it ???
The default is to have it on. If your extensions.conf doesn't override
it, then it is on.
The sample
Hi,
I need some advice from your. I need to develop the GUI which is similar to
[EMAIL PROTECTED], and the Web programming is PHP, and I wish to let the end
user to do
the programming via Web Interface and input data are directly put into the
PostgreSQL, what step should I do? For example
Richard OSS wrote:
rxgain= 0.0
txgain= 0.0
callgroup = 1
pickupgroup = 1
immediate = no
channel = 1-23
Where did you find any example that suggested using '=' for every
setting was appropriate? It's not. Use
Got it... Thanks!!!
--- Colin Anderson [EMAIL PROTECTED]
wrote:
The directory app uses entries in voicemail.conf. In
[EMAIL PROTECTED], I think it's in
/etc/asterisk/voicemail_additional.conf. To add a
user, add them in Amp and
they should be added automatically to the directory,
or you
Hi,
Anyone know how to
activate CallerID in FXS module (S100)? I've no problem to see the incoming
caller ID in * console, but somehow this caller ID is not seen in my analog
phone LCD (with caller ID enabled).
;[206]
signalling=fxo_ks usecallerid=yes
hidecallerid=no
You would use RealTime, but why re-invent the wheel when all of that is
already there in AMP (Asterisk Management Portal)?
chan (Alpha Trilogies Networks) wrote:
Hi,
I need some advice from your. I need to develop the GUI which is similar to
[EMAIL PROTECTED], and the Web programming is PHP,
Ooops...I thought I'd clean up my config files and check for typos. Turns out I was making my problems worse. grinAm surprised As terisk does not complain. Another thing happened to me also. I typed [channel] instead of [channels]. ztcfg looked ok. Asterisk did not complain when started. It
If you are using connecting the card to a smart jack (incoming line from Telco), then you need a straight T1/E1 Cable, which is identical to a straight Ethernet cable. If you are doing a back-to-back configuration, or connecting the card to another PBX or channel bank, then you need a cross-T1/E1
I have 4 telephone lines in the PBX server. One line will be usedby one extension only (i.e. for the boss) for incoming and outgoing.The remaining lines will be shared by all other employees.Some people will be having access to line 1 only. Some have access to line 1 line2 and some have access
Kenige Ho wrote:
the ooh323 is from Asterisk-addon-1.2.1. Is there a bug on this version
for the ooh323 and also how can i get the newer version of the
ooh323(0.8.1) to compile with? Many thanks to you all.
You will find 0.8.X in the asterisk-addons svn branch.
Regards,
Richard
Hi, I have an application,
script.exe, written under mono framework and for execute them in my linux box I
must write in console: mono script.exe The problem is that when I call
this application in dialplan with command: exten = 500,1,System(mono
script.exe) the application not run!
use group in your zapta.conf , ex.
group=1
channel =1
group=2
channel =2-4
hth
On 3/16/06, Faisal Inam [EMAIL PROTECTED] wrote:
I have 4 telephone lines in the PBX server. One
line will be usedby one extension only (i.e. for the boss) for
incoming and outgoing.The remaining lines will be
On Wednesday, March 15, 2006 9:49 PM Hagen Rode wrote:
I've got exactly the same problem with echo, where the mic feeds into
the speakers. I'm looking at purchasing the Tellabs 2572 64ms T1 echo
cancellation card to see if it will help.
Are we still talking about people attending MeetMe
Hi,
how can I send text directly to a specific device, something like:
exten = 103,1,SendTextToDev(SIP/7, hello) ??
TIA
Giorgio Incantalupo
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