Hi All,
I am new to Asterisk. Now I got one question on the identifier parameter of the
Dial() command. I saw as below:
exten = 20,1,Dia(Zap/3/5551234).
Would you please let me know the meaning of 5551234?
Thanks,
Songtao___
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I would strongly suggest you browse:
http://www.asteriskdocs.org/
Kind regards,
PaulH
Songtao Yu wrote:
Hi All,
I am new to Asterisk. Now I got one question on the identifier
parameter of the Dial() command. I saw as below:
exten = 20,1,Dia(Zap/3/5551234).
Would you please let me know
Hello
Dial(Zap/3/5551234)
here 3 is the channel. 5551234 is PSTN number
how ever you will have a better understanding after reading this
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
2009/9/7 Songtao Yu yustao_2...@hotmail.com
Hi All,
I am new to Asterisk. Now I got one question on
On Mon, 7 Sep 2009 02:43:54 +0100
Ex Vito ex.vitor...@gmail.com wrote:
The system specs mention PCIe expansion slots, so your only
option is the TE420B.
--
exvito
Hi Ex Vito,
shouldn't the card be low profile ?
Thanks and have a nice day.
Hi. Has anyone else seen the problem I am having where one touch
recording causes a seg fault? I filed a bug, however I would like to
know if anyone else has seen this and if there is a work around unless
no one uses this very nice feature.
Thanks much for any ideas on this subject.
--
Your
Hello list,
is it possible with the monitor-command to record conversations and
place the soundfile in a pre-defined directory per user ?!
So when extension 200 presses '*#' to record the conversation, the
resulting sound file is written to his home directory on the
Samba-server.
This way each
You are right, I think the provider has some problems, and this should
be fixed.
But is also good to know that I can do this workaround for the worse
case scenario.
thank you !
Olle E. Johansson wrote:
4 sep 2009 kl. 13.40 skrev Marius Ciorecan:
Hello, all. I have an asterisk 2.3.2 and
Hi Martin,
Thanks for the answer!
2009/9/7 Martin asteriskl...@callthem.info
that's probably for ADSI phones ... chan_local confuses the VoiceMailMain
app
and you hear it ...
I'm experiencing this with different SIP phones and softphones.
Why do you need to call it via chan_local ?
This sounds more like the alarm system putting pulses/tones on the line
(maybe the alarm has a dialler/anti-cut-line-detection?
So, as the alarm is adding stuff AFTER the asterisk box - I doubt you
will see anything on the PC itself.
-Original Message-
From:
Hello,
I am writing an AGI script to achieve the following
- Users can Disable/Enable followme from their extension. They can also
change the followme details from their extensions.
I have looked at the follow me table for freepbx. I can't see the field for
the values enabling/disable
The only way around the 'auto-logout' problem I found was to call a script when
agents login. This script checks to see if they are already logged in or not -
then, if they are, it does whatever I want (I manually log them off the other
phone first - you could play a message instead).
HTH
...and did you switch the termination dip switches over (on the NT ports of the
B410P card)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy
Sent: 17 August 2009 07:56
To:
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:
On Sun, Sep 6, 2009 at 10:47 PM, Research resea...@businesstz.com
wrote:
Hello
Hello
I am trying to configure TE420P but i am confused what to give chan_dahdi :(
Below is configuration i am using for TDM400P
Please help what changes to make in it... Please provide a link as well
[trunkgroups]
[channels]
;default for channels
switchtype=national
rxwink=300
Hi All,
I am new to Asterisk and want to perform following on my test project.
I have 2 Asterisk servers SA and SB and an IAX trunk from SA to SB.
Now I set some variables on SB in the same context where an IAX call lands.
My question is , is it possible to access these variables in dialplan of
On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Sun, Sep 06, 2009 at 11:06:39PM -0400, Steve Totaro wrote:
On
Hi,
Sometimes during a confcall I generate calls into meetme to playback
some announcements... If during that time someone joins the conference,
the number of participants is announced counting my announcement call...
Will it be possible to add an option to the meetme application to mark a
I posted about this before without ever finding a solution, so I'm
trying again. I am upgrading from Fedora 8, asterisk-1.4.21, and zaptel
1.4.11 to Fedora 11, asterisk-1.6.1 and dahdi-2.1 . I have a couple of
VOIP phones via SIP and one Wildcard TDM400P REV I with 3 FXS and one
FXO port. I am
Hello
What is your Asterisk problem?, may be I can help you...
I had configure a T1 Card TE121 connected with and AVAYA PBX
Best regards
-Mensaje original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de
mancyb...@gmail.com
http://www.voip-info.org/wiki/view/Digium+TE420P
http://www.voip-info.org/wiki/view/Digium+TE420PThe best resource is also
quite
On Mon, Sep 7, 2009 at 3:14 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Hello
I am trying to configure TE420P but i am confused what to give chan_dahdi
:(
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On
hi,
can anyone knows a way to make automatic calls from a list of numbers stored
in a file, one by one, as the calls hangs up.
EX:
1º call - hang up - 2º call - hang up - 3º call ..
thanks,
pn
___
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Hello All
I have the dahdi channels working also I can have a call between the
equipments, but when I try to dial a second call I receive the error below:
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/4000-09a44a78' status is 'CHANUNAVAIL'
But the
Hello list
I had the following echo-test extension on my Asterisk 1.2 setup.
exten = 1003,1,Wait(1)
exten = 1003,n,Playtones(!1050/1000)
exten = 1003,n,Wait(1)
exten = 1003,n,StopPlaytones
exten = 1003,n,Echo
exten = 1003,n,Hangup
After migrating my testing server to Asterisk 1.4, and a minor
On Monday 07 September 2009 05:55:12 Asterisk User wrote:
I am new to Asterisk and want to perform following on my test project.
I have 2 Asterisk servers SA and SB and an IAX trunk from SA to SB.
Now I set some variables on SB in the same context where an IAX call lands.
My question is , is
On Monday 07 September 2009 09:36:53 Ivan Stepaniuk wrote:
I had the following echo-test extension on my Asterisk 1.2 setup.
exten = 1003,1,Wait(1)
exten = 1003,n,Playtones(!1050/1000)
exten = 1003,n,Wait(1)
exten = 1003,n,StopPlaytones
exten = 1003,n,Echo
exten = 1003,n,Hangup
After
Tilghman Lesher wrote:
On Monday 07 September 2009 09:36:53 Ivan Stepaniuk wrote:
I had the following echo-test extension on my Asterisk 1.2 setup.
exten = 1003,1,Wait(1)
exten = 1003,n,Playtones(!1050/1000)
exten = 1003,n,Wait(1)
exten = 1003,n,StopPlaytones
exten = 1003,n,Echo
exten =
On Mon, 7 Sep 2009 08:48:25 -0500
Juan Cardoza jcard...@tpmex.com wrote:
Hello
What is your Asterisk problem?, may be I can help you...
I had configure a T1 Card TE121 connected with and AVAYA PBX
Best regards
Hi Juan,
thanks for your help.
I'm going to choose a 4 ports PRI digium card
PCI Express x1 card will work and will fit in the x8 slot
PCI-X slots are usually 3.3V
Martin
On Mon, Sep 7, 2009 at 10:35 AM, mancyb...@gmail.commancyb...@gmail.com wrote:
On Mon, 7 Sep 2009 08:48:25 -0500
Juan Cardoza jcard...@tpmex.com wrote:
Hello
What is your Asterisk problem?, may
Hello,
with Asterisk 1.6.1.6 I try to hangup a call if called extension is not
existing. For this purpose I would use the internal i extension but
seems not to work.
[MyContext]
exten = s,1,NoOp(Call is treated as it should)
exten = s,n,NoOp(next step)
exten = s,n,NoOp(aso ...)
exten =
I am having a strange problem today. When I arrived into the office
today two Aastra phones were not working. They can receive calls but
not make them. The models are 480i and 480i CT. Other Aastra phones
like 55i and 57i work fine, it only seems to affect the older phones.
Today is a strange day. My asterisk server is suddenly saying that all
extensions are on hold. All my hints are like this:
-= Registered Asterisk Dial Plan Hints =-
4...@hints : SIP/4101
State:HoldWatchers 0
4...@hints
On Mon, Sep 7, 2009 at 10:09 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen tzafrir.co...@xorcom.com
wrote:
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
On
No # is not used for anything. Asterisk is acting up today so I guess
this is not directly related to the phone. All hints say all extensions
are on HOLD.
On Mon, 2009-09-07 at 13:40 -0400, Michelle Dupuis wrote:
Are you using the # symbol in the extension name or to access a feature
Are you using the # symbol in the extension name or to access a feature
(eg: outside line)??
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Sent: Monday, September 07, 2009 1:12 PM
To: Asterisk
Hi there,
I need some help with a 'custom' feature.
I have following feature defined in features.conf :
[applicationmap]
opnemencallee =
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
In my dialplan :
[from-HostAst]
exten = s,1,Set(__DYNAMIC_FEATURES=opnemencallee)
exten =
Administrator TOOTAI escribió:
Hello,
with Asterisk 1.6.1.6 I try to hangup a call if called extension is not
existing. For this purpose I would use the internal i extension but
seems not to work.
[MyContext]
exten = s,1,NoOp(Call is treated as it should)
exten = s,n,NoOp(next step)
Olle E. Johansson o...@edvina.net writes:
Imaging my surprise this Monday when I installed a plain old Asterisk
1.4 on a new HP server, a DL380 G6, and could run in circles around
the old IBM servers.
The G6 series is pure magic for everything I've let it touch
network-wise.
I have three
On Mon, Sep 7, 2009 at 4:19 PM, Benny Amorsen
benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
wrote:
Olle E. Johansson o...@edvina.net writes:
Imaging my surprise this Monday when I installed a plain old Asterisk
1.4 on a new HP server, a DL380 G6, and could run in circles around
the
Miguel Molina a écrit :
[...]
The 'i' extension only works in applications like Background(),
WaitExten() and everything that uses DTMF to route extensions within a
context.
Well, from reading voip.org it's not really clear than ...
[...] Because the call is not
accepted there's no need
On Monday 07 September 2009 13:40:16 jonas kellens wrote:
[applicationmap]
opnemencallee =
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
FeatureName =
DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]]
it looks like
It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT) have a
problem with the new SIP implementation in Asterisk 1.6.X that makes
them unable to dial. They can receive calls but when you attempt to
dial the phone remains silent. You can see in core show channels that
the first
On Mon, Sep 07, 2009 at 07:44:07AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 5:58 AM, Tzafrir Cohen
tzafrir.co...@xorcom.comwrote:
On Mon, Sep 07, 2009 at 01:15:12AM -0400, Steve Totaro wrote:
On Mon, Sep 7, 2009 at 1:03 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com
wrote:
Hi list,
I hope someone could help me. I've started using Asterisk 1.6.0.14 to get
queue logs in real time with odbc (our databases are all PostgreSQL) but
it's not working. However, cdr odbc is working well. When asterisk starts
next message appears:
WARNING[4217] config.c: Realtime mapping for
I imagine this setup will need those two communicating entities to be part
of the pabx. But support extension 100 of PABX A (legacy) calls 101 on the
same platform. I want asterisk connected to PABX A via E1/T1 to know about
that call and start recording (tap) without bridging or being part
How we will the channels will be handled. I want to test it for loop back .
On Mon, Sep 7, 2009 at 6:57 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
http://www.voip-info.org/wiki/view/Digium+TE420P
http://www.voip-info.org/wiki/view/Digium+TE420PThe best resource is
also quite
On Mon, 2009-09-07 at 07:20 -0600, Greg Woods wrote:
incoming calls
through the FXO line are dropped as soon as there is a button press.
The error logged is:
[Aug 23 18:15:39] WARNING[6532] chan_dahdi.c: Cannot handle frames in 2
format
[Aug 23 18:15:39] WARNING[6532] file.c: Failed to
On 8/09/09 2:21 AM, Juan Cardoza wrote:
Hello All
I have the dahdi channels working also I can have a call between the
equipments, but when I try to dial a second call I receive the error below:
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel
On 8/09/09 5:35 AM, Carlos Chavez wrote:
Today is a strange day. My asterisk server is suddenly saying that all
extensions are on hold. All my hints are like this:
-= Registered Asterisk Dial Plan Hints =-
4...@hints : SIP/4101
State:Hold
On Mon, Sep 7, 2009 at 12:29 AM, Steve
Totarostot...@first-notification.com wrote:
Did you push it past 300 on two year old hardware and software?
old hardware yes.
old software no.
The servers are more than 3 years old
Core 2 Duo Dell Dimension desktop as proof of concept?
are core 2 duo's
On Monday 07 September 2009 17:16:12 Daniel - Asterisk wrote:
WARNING[4217] config.c: Realtime mapping for 'queue_log' found to engine
'odbc', but the engine is not available
My configuration follows:
/etc/asterisk/ extconfig.conf
[settings]
queue_log = odbc,postgres,asterisk.queue_log
I
Sorry , I checked on my Asterisk pbx and there is no sip.cfg file on it .
Can you please let me know how can I make my Asterisk Call Parking as
functional ?
On Tue, Sep 1, 2009 at 6:23 AM, Lee, John (Sydney)
john@compuware.comwrote:
Please find attached my Asterisk sip.conf .
Can you
Hello I have the loop back connector and TE420P card but i dont know how to
configure that. Please let me know of any help.
I am facing the problem in configuration of channels.
i have make changes in chan_dahdi
[r...@te420 etc]# dahdi_hardware
pci::04:08.0 wct4xxp+ d161:0420
Thanks Tilghman for your quick reply.
I know that we should set variables through IAXVAR on source server to
access them on Destination server.
I just wanted to know the reverse case, where IAX channel variables set on
destination server are accessible on Source server or not.
Thanks again for
Dear All
Can you please do me favor and let me know why my converted sound files are
not being played and heared on my Asterisk ? Please find attached my sound
files . Actually , I had them recorded as *.wav files and I tried to convert
them to *.gsm as the followings :
#sox FR3.wav
confBridge in Asterisk 1.6.2.0-rc1 doesn't stable.
It causes segment fault very often and results in asterisk crash.
Ian
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Do you have an error message?
On Tue, Sep 8, 2009 at 1:29 PM, hadi motamedi motamed...@gmail.com wrote:
Dear All
Can you please do me favor and let me know why my converted sound files are
not being played and heared on my Asterisk ? Please find attached my sound
files . Actually , I had
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