We have just installed one machine with FC3 (with last updates) + asterisk
1.2.1 + spandsp-0.0.2pre21. From our tests it shows OK.
Ioan Indreias
Modulo Consulting
www.tenora.ro
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Tuesday
Hello,
We had the same problem in the past and the last idea I had was to remove
first the modules and load them (using /etc/rc.local) in the right order.
Like:
rmmod wcte11xp
rmmod wctdm
modprobe wcte11xp
modprobe wctdm
modprobe zaptel
Maybe not the best way to do the job but it works for us.
Maybe you could use something like:
exten = boss_ext,1,GotoIf($[${CALLERID(number)}=secretary_ext]?boss:secretary)
exten = boss_ext,n(boss),Dial(SIP/boss_ext)
exten = boss_ext,n(secretary),Dial(SIP/secretary_ext)
## nini @ www.modulo.ro ##
Jonathan k. Creasy wrote:
Why don't you just give
Hello,
Maybe using app_backticks will solve your problem.
I use it to call a script and saved the result into an Asterisk variable.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
Pavel Jezek wrote:
any
Hello,
Check app_backticks - it is an external application which should be
compiled on your system.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
[EMAIL PROTECTED] wrote:
Hi all,
is where a
Hello Larry,
A quick suggestion - probably is not the best one, but maybe it will help:
exten = s,n,GotoIf($[${CALLERID} = ]?anonymous:${CALLERID}|1)
exten = s,n(anonymous),Set(CALLERID(num)=NOCID)
exten = s,n(continue),..
exten = _4XX,1,Set(CALLERID(num)=Internal
exten =
Hello Larry,
Probably your variable (MYIP) is not accessible to asterisk process
environment.
Test it with ${ENV(PATH)} and you will have a result there
exten = s,n,Set(test=${ENV(PATH)})
-- Executing Set(IAX2/test_iax,
test=/sbin:/usr/sbin:/bin:/usr/bin:/usr/X11R6/bin) in new stack
--
Hello,
Maybe it is too late but it may help you.
Check the configuration for the SIP client identified by 192.168.0.123
(or the IP mentioned by the error line)because it tries to subscribe
to get BLF indications for the X extension. Most probably it is for
an old phone BLF configuration.
Hello John,
I'm not sure - but when tou try to define a context for testq queue with:
context=testing
it is useless. From what I know you could not have such an option inside
a queue.
Did you find any documentation specifying a context for a queue?
Best regards,
## nini @
Hello,
I see that you are using T option (allow the /calling/ user to
transfer the call) when dialling to internal extensions and t (allow
the /called/ user to transfer the call) when receiving calls (in home
context). This it is why inbound transfer works fine and only one time.
So, I
Hello Benny,
Maybe you could use the following solution (assuming that the regional
prefixes are the first 3 digits of the national number): Define for each
phone a full national CallerID number and use the same context:
[from-sip]
exten = _Z.,1,Goto(outgoing,${CALLERIDNUM:0:3}$EXTEN,1)
Hello,
I'm not familiar with A2billing but for me it is strange that you dial
SIP/777 - 777 should be an extension.
Could you post your user context - or at least the one which direct
you to:
Dial(SIP/9614-3896, SIP/777|200|rt)
Best regards,
## nini @ www.modulo.ro ##
[EMAIL PROTECTED]
Hello Benito,
I suggest to specify which span to be used as the clock source (check
http://lists.digium.com/pipermail/svn-commits/2005-October/007955.html)
span=1,1,0,ccs,hdb3,crc4
bchan=25-39,41-55
dchan=40
span=2,0,0,ccs,hdb3,crc4
bchan=56-70,72-86
dchan=71
HTH
Best regards,
## nini @
Hello Benito,
From http://www.beronet.com/download/card_installation_guide.pdf we
could find that:
/After loading the driver and the executing ztcfg, status LEDs for
each port should be flashing red, unless the port is connected to a
device. If the LED does not light up, the driver did not
Hello,
Use the cross-over schema for creating a self cross connector.
Meaning you will connect your TX pair to your RX pair. This will be the
test of the physical layer of your card and the flashing red light of
the led will have to turn in green. Otherwise something is not
working/configured
Hi Matt,
Probably you already found it - but I think it could help others:
http://www.voip-info.org/wiki/view/Aastra+Failsafe+Reboot+Script
You have to give the password - but for us it was OK.
Best regards,
## nini @ www.modulo.ro ##
Matt wrote:
Is there a command in Asterisk that will
Hi Clara,
You could put some data into astdb and query for the outgoing line and
callerid based on internal callerid (extension).
something like
user/201/outline 89859715
user/201/outcallerid 89859715
and so on...
By the way: _89859715 without the dot (.) is same like 89859715 - maybe
you
Hi Joe,
Maybe there it is a problem related to the rights on the specific files.
Please check if you have read rights for everybody for the files under
/tftpboot. Also check that tftpboot have r+x rights for everybody.
Regards,
Ioan.
www.modulo.ro
-Original Message-
From: [EMAIL
here:
http://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515
Even if you not use AMP, it give you some guides on how to configure
iax.conf and extensions.conf files
--//--
Ioan Indreias
IT Consultant
Modulo Consulting - http://www.modulo.ro
based on the audio
level injected into PC's audio card (mic port).
Hope it helps.
Ioan Indreias
Modulo Consulting - http://www.modulo.ro
I'm not really trying to monitor anything on the asterisk box at all. I
guess this is more of an SIP phone question. Really all I need is to get
the audio
Hi,
I have tried and here it works fine (asterisk 1.2.1), with the following
configuration:
zapata.conf
context=testing
channel = 5
extensions.conf
[testing]
exten = s,1,Dial(ZAP/1/07XX)
from CLI:
-- Starting simple switch on 'Zap/5-1'
-- Executing Dial(Zap/5-1,
,iKkTt*j*)
3. For more hints you could check
voip-infohttp://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
page.
HTH
Ioan Indreias
www.modulo.ro
On Wed, Sep 16, 2009 at 4:52 PM, Juan Cardoza jcard...@tpmex.com wrote:
I comment all the lines in my extensions.conf file to work only with the
lines
Hello Harley,
Please find the directions I've used in order to get this work on our
Asterisk machine.
The flow
===
A conference user (A) decide to invite somebody else (B) into the
conference. Pressing 0 from his dialpad A will hear a dial tone and he
have to enter the destination number
Hello Cyprus,
What is the output of moh files show CLI command ?
Best regards,
Ioan (Nini) Indreias
www.modulo.ro
On Fri, Oct 2, 2009 at 11:46 AM, Cyprus VoIP voi...@gmail.com wrote:
Hi,
I deleted all the default files and put one that I know that works on
another Asterisk, but since then,
DAHDI/DGTDM24/966505103250
This (DGTDM24) is strange. Could you provide the setup of the DAHDI trunk?
You should have something like DAHDI/g0/96 or DAHDI/10/96
Here are more info on this subject:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226642.html
HTH,
Ioan
I cant find Zapata.cfg
You have a DAHDI installation thus you have to find chan_dahdi.conf.
it should be located under /etc/asterisk
Regarding the configuration for tones you have to check indications.conf file
Best regards,
Nini
___
-- Bandwidth
Hello Jonas,
I had the same problem and from my own research I found that you
could not made a distinction.
The problem is that the peer is identified based on the IP (or
IP+PORT) information found in INVITE. And you (and me) have same IP
(in my case same port as well) for several SIP accounts.
Hello Anahi,
1. Do not use the Answer
2. Use Goto(3005,1) instead of Dial
Thus I would write only the following line:
exten = 2001,1,Goto(3005)
I do not understand exactly what you want to do but hey...
HTH,
Ioan (Nini) Indreias
www.modulo.ro
2009/10/8 Anahi Ludueña a_ludu...@hotmail.com:
On Fri, Oct 9, 2009 at 10:37 AM, jonas kellens jonas.kell...@telenet.be wrote:
So the call comes into the right context... that's not the problem.
But my CDR is messed up. The accountcode that I have set for user1 is always
replaced for the accountcode I've set for user 2.
[YOCAN-3starsnet]
Have you used a mobile phone when you test the fast speed DTMF sequence?
We have found that in GSM network DTMF digits are sent out-of-band from the
terminal (despite the tones generated by phones) and are injected in-band
into the audio channel _but_ with some delays between digits. At least this
Hello Xavier,
Unfortunately we are not aware of any Asterisk configuration which will
protect against of a brute force attack on SIP.
We use BFD - http://www.rfxn.com/projects/brute-force-detection/ .
We have found first details here: http://engineertim.com/?cat=15 and we are
currently
Hi Anees,
Have you tried to monitor the number of active channels?
Something like:
watch 'asterisk -rx show channels | grep active'
According with your setup the maximum number of active calls should be
7x120=840 - or near this number.
Maybe the calls are not closed properly and you
is taken into consideration = all calls are
sent to the context of that last extension.
You could check this if you configure a higher verbose/debug level
(like more than 10) and check into the Asterisk logs the information
displayed by chan_sip.c
HTH,
Ioan Indreias
www.modulo.ro
### extract
On Tue, Mar 2, 2010 at 1:51 AM, lesouvage i...@meetmecall.nl wrote:
You doesn't seem to have a proper context,extension,priority available
for internal calls while you have one for outbound calls. To get more
detailed help an even an answer you have to provide more info. The
cli output while
On Tue, Mar 2, 2010 at 4:17 AM, sean darcy seandar...@gmail.com wrote:
Do you have to Answer() to reach the fax extension?
That is assume you have:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected) ;; the fax line
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup()
(or other languages) I
encourage you to read it (the pictures and the results are very easy
to understand) and send your feedback or comments here.
Best regards,
--
Ioan Indreias
www.modulo.ro
Notes:
[1] -
http://www.modulo.ro/Modulo/ro/Articole
On Fri, Mar 19, 2010 at 3:13 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
Fail2ban is a must. I was a victim of such attacks, and have implemented
some other measures too, but fail2ban is a must have with the link posted by
Matt which describes how to set it up for asterisk. Make sure you put
I would say that from what I know DND function in FreePBX will not
automatically configure the phone DND function but it set a flag into
Asterisk DB:
-- Executing [...@from-internal:5] Set(SIP/117-01f6,
DB(DND/117)=YES) in new stack
You report that you do not hear nothing but in the log
Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
other ATA which have FXO ports).
HTH,
Ioan.
On Thu, Apr 1, 2010 at 12:29 AM, Kosa k...@piradio.org wrote:
I have two linksys spa2102 and a sap9000 but as far as I know I need
something else to connect the asterisk box to the
what about
cat /proc/zaptel/*
HTH,
Ioan.
On Fri, Apr 2, 2010 at 2:46 PM, Jaap Winius jwin...@umrk.to wrote:
# asterisk -rx 'pri show spans'
PRI span 1/0: Provisioned, Down, Active
--
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-- Bandwidth and Colocation Provided
requirement
(is based on intercom module present in FreePBX and adapted to have
only one way audio for 60 secconds). We have tested with Linksys
SPA9XX phones and works fine (hint: clear regional=call progres
tones=page tone in order to cancel the page tone if you need to be
super-silent).
HTH,
Ioan
We have used with success BBB (BigBlueButton - open source -
http://bigbluebutton.org) and I recommend to try their demo in order
to see if this solution gives all you need.
Voice conf is based on Asterisk.
HTH,
Ioan Indreias
www.modulo.ro
On Thu, Apr 15, 2010 at 2:04 AM, Stéphane Bauland baula
another idea you could test is to use a very short Timeout in your Dial command.
like Dial(ZAP/012345678,1) - will dial and exit after 1 sec with
DIALSTATUS set accordingly
HTH,
Ioan
--
_
-- Bandwidth and Colocation Provided
Hi Gilles,
Just to provide an alternative to sshguard: you could use BFD[1]
(based on bash scripts) and configure it to use iptables to block the
attacker host.
The default configuration is to check the logs at each 3 minutes
(using a crontab entry).
BFD rules for Asterisk could be found here
Hello,
We have chosen to upgrade our Trixbox installations (2.6.2.3, asterisk
1.4.20) and everything work smooth.
The problem we face now is that asterisk14-addons-mysql looks to have
not been compiled with uniqueID feature and we are asking your opinion
about what should be the best fix for
[Danny Nicholas]
IMO, one of the selling points of the add-on modules is that they can be
compiled/tweaked without too much input from the base installation. I don't
think you're going to get too far with the new/modified RPM request.
Well - it looks we are the only ones needing that RPM
On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham rizwanhas...@gmail.com wrote:
I am in desperate need of this feature. I want to play background music
during a call while the 2 parties are having some lovely conversation (or
maybe give them a sort of cursing background if they are cursing each
I have tested the following dialplan and it could be used as a
starting point. What you have to resolve is how to generate different
MeetMe conference room - in the example we have only one room = 1234
If you prefix the dialled extension with 1 = you will have a lovely
chat. With 2 - cursing
,MeetMeAdmin(${MM},K)
+
On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias indre...@gmail.com wrote:
I have tested the following dialplan and it could be used as a
starting point. What you have to resolve is how to generate different
MeetMe conference room
Glad to know it works for you.
I would like to hear your love/curse MOH - do you have some links to
your mp3 files? :)
BR,
Ioan (with capital i)
On Tue, May 10, 2011 at 6:59 PM, Rizwan Hisham rizwanhas...@gmail.com wrote:
Ooops,
here is the correct version, Missed the capital X option in
Hello,
I'm writing here hoping to have a hint from Asterisk/Digium packager
maintainer, Jason Parker (of course that any other's opinion is
welcomed).
We have installed an asterisk machine using Asterisk and Digium repos.
Unfortunately we have found that an Astribank could not be connected
due
Maybe you could use a very simple sollution like a meetme room - you have
only to be creative with the dialplan.
Ioan
www.modulo.ro
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
Hello,
Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM
version from Asterisk repo I found that asterisknow-version is needed
by package asterisk18-core-1.8.7.0-2
How could this be explained?
Best regards,
Ioan
#
[root@localhost ~]# yum update asterisk18* -x
On Mon, Oct 17, 2011 at 10:37 PM, Jason Parker jpar...@digium.com wrote:
On 10/17/2011 02:22 PM, Ioan Indreias wrote:
The asterisknow-version package contains the repository files (see
/etc/yum.repos.d/) for the repositories on packages.asterisk.org and
packages.digium.com. Installing
On Thu, Jan 12, 2012 at 7:50 PM, mahesh katta maheshka...@flexydial.com wrote:
I was search for free license but for this Digium require purchase any
Hardware then they can provide Free License.
But I have no Digium Device , I am using Grand stream FXO Gateway and
Asterisk.1.8.XX .
I was
This is a FreePBX question as the Asterisk dialplan is managed by it.
I suggest to use 'extensions_override_freepbx.conf' (details in
extensions.conf) and place there your modified [macro-dialout-trunk].
HTH,
Ioan
On Fri, Feb 10, 2012 at 1:13 PM, ing.Achim Alexandru
alexandru.achi...@gmail.com
FreePBX have also an ISO distribution - I would recommend to use that one.
HTH,
Ioan
On Wed, Feb 29, 2012 at 7:43 PM, Danny Nicholas da...@debsinc.com wrote:
Asterisk Now should serve your needs nicely.
--
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-- Bandwidth and
On Thu, Mar 22, 2012 at 6:29 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
I'm following
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-PreparingAsteriskToProduceCoreFilesOnCrash
But there is nowhere information on possible error-messages that you can
Or you could use a System call in the hangup dialplan and trigger a new
call as soon as an old one just finished. Maybe a silly idea but it shpuld
just work.
Ioan
--
_
-- Bandwidth and Colocation Provided by
On Sat, May 26, 2012 at 9:52 AM, Moises Silva moises.si...@gmail.comwrote:
There is nothing hybrid like that (GSM + Analog) in the NorthAmerica or
Europe to my knowledge. We at Sangoma (from Canada) have a 4-port GSM card
though which uses chan_dahdi (patching needed at the moment).
On Thu, Jun 28, 2012 at 10:53 PM, Ernie Dunbar maill...@lightspeed.ca wrote:
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen), and
Voip3 has a Wildcard TE110P. Voip1 is the server that handles our PRI to
On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar
ellen.apolinar...@googlemail.com wrote:
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
Data: SIP/voipms/customer_number
HTH,
Ioan Indreias
Modulo Consulting // www.modulo.ro
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
You could dynamically change the queue penalties (QUEUE_MIN_PENALTY and
QUEUE_MAX_PENALTY) through queuerules.conf - check [1].
HTH,
Ioan
[1]
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id288932.html
--
I think you could use twice the Park action to park the channels -
https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Park
In the end you will have to bridge the parked channels.
HTH,
Ioan
On Wed, May 15, 2013 at 1:03 PM, Lenz Emilitri lenz.lo...@gmail.com wrote:
I never actually used
BTW - what was exactly the problem when trying to bridge the two channels
that you have sent to the wait application?
On Wed, May 15, 2013 at 4:29 PM, Ioan Indreias indre...@gmail.com wrote:
I think you could use twice the Park action to park the channels -
https://wiki.asterisk.org/wiki
Hello Shanavaz.,
Please find some quick thoughts:
* 2 main queues
* agents logged on one or on both main queues
* before sending a new call to one of the main queues check the number of
waiting callers (QUEUE_WAITING_COUNT function) and divert (for example for
30 sec) the call on a empty members
Hello Andy,
Have you tried using SetMusicOnHold command before Queue command?
BR,
Ioan
On Wed, Jul 10, 2013 at 7:55 PM, Andrew Thomas a...@datavox.co.uk wrote:
Hi All,
Sorry if this has been covered already, but I don't tend to follow this
list as close as I should these days.
Problem
Have you tried finish the dialed number with #?
I'm not sure if this is working on your setup but this is one workaround
that we give to our users when they initially complain about big delays on
SIP phones (where we have not changed yet the default dialplan).
HTH,
Ioan.
On Fri, Jul 12, 2013
Hi Rizwan ,
Have you tried to define astspooldir (usually /var/spool/asterisk) to a
shared filesystem? Or to create a symlink for outgoing directory (where the
call files have to be placed) to a directory placed on a shared filesystem
(eg on a NAS)?
Just brainstorming - not yet tried and maybe
Hello,
Same issue happens on one of our Call Center installation (using Asterisk
1.6) - random unresponsive Asterisk with self heal after 2-3 minutes.
Because we could not find the root cause (till now - many thanks Ishfaq) we
end up by nightly restart on Asterisk.
We are using CLI commands more
Have you tried to restart asterisk after setting the correct permissions?
HTH,
Ioan
On Thu, Nov 21, 2013 at 6:04 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Hi all,
I am syncing call files on my secondary asterisk server but without
permission to read for asterisk. So they should be
I would start to combine audio and video sources inside a conference room.
HTH,
Ioan
On Sun, Nov 24, 2013 at 11:44 PM, Eric Cooper e...@cmu.edu wrote:
I'd like to cobble together a videophone from an analog phone,
connected to an Asterisk FXS channel, and a co-located video camera,
Many thanks Tzafrir - it works like a charm.
Best regards,
Ioan
On Sun, Dec 1, 2013 at 1:46 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
Hi,
Long ago, On Wed, Feb 21, 2007 at 09:32:26AM +0200, Tzafrir Cohen wrote:
On Wed, Feb 21, 2007 at 07:56:18AM +0100, Olivier wrote:
[snip]
Hello Steve,
Have you tried to send the automated call to your dialplan instead of the
phone?
For example, instead of calling SIP/aastra_phone call
Local/aastra_phone@auto-answer-context and tweak auto-answer-context from
your dialplan as needed.
HTH,
Ioan
On Tue, Jan 28, 2014 at 6:56 PM,
You should check ${QUEUE_PRIO} channel variable. Before sending the call to
the queue set this variable for VIP callers, identified by CallerID.
We've made this for several of our customers by querying Elastix
AdressBook, MySQL database or custom CRMs (through HTTP requests).
HTH,
Ioan
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