Klaus Darilion wrote:
The backtrace is not useable. Try to rebuild Asterisk with the Don't
Optimize Option (make menuconfig and the the build options)
did that. no effect.
i've got exactly the same result.
Edwin Lam wrote:
Philip A. Prindeville wrote:
On 03/08/2010 04:31 PM, Edwin Lam
Hello,
Does asterisk func odbc support multi query? I'm executing stored
procedure which returns two tables. With tsql command I can see both tables.
But asterisk only shows the first.
My database is MSSQL.
Maybe there is workaround...
Thanks
--
Best Regards,
Giedrius
--
Thanks,
I have uploaded the patch to the website and will let you know the
feedback we receive.
Greetings,
Joachim
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
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Hash: SHA512
Edwin Lam skrev:
Klaus Darilion wrote:
The backtrace is not useable. Try to rebuild Asterisk with the Don't
Optimize Option (make menuconfig and the the build options)
did that. no effect.
i've got exactly the same result.
Edwin Lam
hi folks,
I want to change the callerid= variable generated from php page.
Let me explain :
in /var/log/asterisk/cdr-csv/Master.csv we have the following line :
device 360
we want to change it we don't want the extension name but the displayed name
instead of 360 we want Poste 360 or something
hi, All
one thing confused me a long time.
when i change the extensions.conf file. why not take effects after
restart the asterisk? details as follow:
my dailplan is :
[95040]
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
exten = _95040X,n(start),Answer
exten =
check the Goto cmd syntax
2010/3/10 Zhang Shukun bit...@gmail.com
hi, All
one thing confused me a long time.
when i change the extensions.conf file. why not take effects after
restart the asterisk? details as follow:
my dailplan is :
[95040]
exten =
On Wed, Mar 10, 2010 at 10:38 AM, Zhang Shukun bit...@gmail.com wrote:
hi, All
one thing confused me a long time.
when i change the extensions.conf file. why not take effects after
restart the asterisk? details as follow:
Type dialplan reload in your CLI
--
Hello list.
An incoming call goes to the queue. Then is routed to a free
SIP-member1. When this SIP-member1 transfers the call to another
SIP-member2, and this SIPmember-2 rejects the call, then the
communication is lost.
How can I make the call go back to the SIP-member1 ? Or maybe back to
the
On Tue, Mar 09, 2010 at 06:20:53PM -0500, sean darcy wrote:
Receiving a fax pstn - pstn with 1.6.2.6-rc2:
-- Executing [...@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in
new stack
-- Executing [...@incoming-pstn-line:2] Wait(DAHDI/4-1, 3) in new
stack
-- Executing
Hi all,
is it possible to send a parameter to the asterisk voicemail system?
I would like to create an IVR that asks for a client code and after
that transfer to a particular voicemail. Then the voicemail should
send an email with the client code in the subject.
The only problem is about the
Georghy a écrit :
hi folks,
I want to change the callerid= variable generated from php page.
Let me explain :
in /var/log/asterisk/cdr-csv/Master.csv we have the following line :
device 360
we want to change it we don't want the extension name but the displayed name
instead of 360 we want
On 10 Mar 2010, at 05:41, Gopalakrishnaiyer Venugopal-Q16770 wrote:
So is this a bug in Asterisk 1.6? Has anyone verified/reported this
issue?
Read what people send you. Are you using FreePBX? If yes, then that
ticket is a FreePBX bug report. If you read the words in the report it
will
you can consider the internal db of asterisk. look at:
http://www.voip-info.org/wiki/view/Asterisk+database
2010/3/10 Carlo Dimaggio jaasmail...@gmail.com
Hi all,
is it possible to send a parameter to the asterisk voicemail system?
I would like to create an IVR that asks for a client code
Hello,
I encountered the dtmf problem between my asterisk box (1.4.23) and
suppliers gateway (unknown vendor). I have dtmf mode set to rfc2833 and it
alway worked till supplier has changed something. Now I receive from him
dtmf payload 100. With the second supplier which sends dtmf with payload
This one is pretty clear - asterisk is expecting to go to a valid context
and tag. You are dialing 9504012345 and the goto is trying to go to
9504012345,welcome or 9504012345,3 when it should actually go to
${CALLINNUM}:0:5 to go to 95040,welcome for any call starting with 95040.
_
I understand the “ex-girlfriend” situation, in fact I want to do that, the
problem is when I don´t put the last line and call from 92 or 91 this don´t
work.
I put the ex-girlfriend exception because without this, calls from 91 and
92 don´t match their extensions.
On Mon, Mar 8, 2010 at 4:52
This may just be my opinion, but EG logic works best in an established call,
like this
[TRONCAL-SIP]
exten = 225,1,answer
exten=225/91,2,Answer
exten=225/91,3,Echo
exten=225/92,2,Answer
exten=225/92,3,Playback(conf-invalid)
exten=225,hangup
This way, 225 is answered and hungup regardless of
Andreas Brodmann wrote:
Lief,
I'd be glad to receive your feedback.
I don't think it's a limit of lines by itself. I haven't found any useful
debug information so far, but I think the dialplan parser stumbles
over something.
The problem is reproduceable on different hareware, it only
Il giorno 10/mar/10, alle ore 14:48, Emanuele Carbone ha scritto:
you can consider the internal db of asterisk. look at:
http://www.voip-info.org/wiki/view/Asterisk+database
What about the matching between the parameter stored in db and the
voicemail message?
Asterisk can be configured to
I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN).
When chan_ooh323 first loads it tries to establish a connection with the gk
but I it fails. I have the following extract from the ooh323 log. Can
anyone give some insight?
Thanks!
MD
23:02:59:045 Sent GRQ message
We are coordinating a connection to a SIP provider who told us they use two
port ranges for RTP, 7000-8000 and 1-2.
We've never encountered that before (and I believe rtp.conf only supports a
single range). We can obviously setup 7000-2 within RTP.conf, but I'm
wondering if there is
Klaus Darilion wrote:
That's weird. AFAIK Asterisk does not allow multiple ranges. Maybe they
are having 2 ranges for RTP and UDPTL (T.38). Asterisk allow
configuration of different ranges for UDPTL and RTP (although it
shouldn't be a problem to configure the same ports in rtp.conf and
On 10.03.2010 16:35, Michelle Dupuis wrote:
We are coordinating a connection to a SIP provider who told us they use
two port ranges for RTP, 7000-8000 and 1-2.
They use these ports. So there is nothing you have to do on Asterisk
side to handle this, as Asterisk's RTP ports are
I am running Asterisk trunk with ooh323 towards an Avaya CM 3.1 without
any problems. I need your ooh323.conf and all relevant CM config
(signal-group, trounk-group, ip-codec... ) before I can assist u. ;)
On Wed, 10 Mar 2010 10:34:30 -0500, Michelle Dupuis wrote: I'm trying to
connect an
I don't have access to the Avaya (client controlled) - but I will ask for
their config. Does the Avaya allow dumping the config to a file? (Or
screen shots only?)
Here's my ooh323.conf (IP's changed)
Thanks,
MD
[general]
port=1720
bindaddr=99.197.126.160
faststart=yes
h245tunneling=yes
Hello list,
Can I do something like this for BLF functionality :
[test-blf]
exten = _XX,hint,Macro(GetSIPaccount,${EXTEN})
exten = _XX,hint,SIP/${SIPACCOUNT}
GetSIPaccount is a macro that looks at a MySQL-database for the realtime
table sip_buddies where the SIPusername is taken from.
It
Yes, this work, thanks!
On Wed, Mar 10, 2010 at 11:33 AM, Danny Nicholas da...@debsinc.com wrote:
This may just be my opinion, but EG logic works best in an established
call, like this
[TRONCAL-SIP]
exten = 225,1,answer
exten=225/91,2,Answer
exten=225/91,3,Echo
exten=225/92,2,Answer
On Wednesday 10 March 2010 02:09:54 voipas wrote:
Does asterisk func odbc support multi query? I'm executing stored
procedure which returns two tables. With tsql command I can see both
tables. But asterisk only shows the first.
My database is MSSQL.
Yes, but only in 1.6.0 and above. You'll
On Fri, Mar 5, 2010 at 3:06 PM, Robert McGilvray rmcgi...@globeop.com wrote:
Does anyone use confbridge in a large installation and can provide feedback
on its stability, quality in comparison to MeetMe? I use a sangoma card in
my 1.4.2 box to provide timing and it has never been an issue. Can
hmmm... will be hard to help u without u having access... will do my best.
Here is my ooh323.conf anyway...
sip:/etc/asterisk# cat ooh323.conf
[general]
bindaddr=213.88.138.183
port=5088 --
_
-- Bandwidth and Colocation
On Wed, Mar 10, 2010 at 3:38 AM, Zhang Shukun bit...@gmail.com wrote:
[95040]
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
exten = _95040X,n(start),Answer
exten = _95040X,n(welcome),Background(${welcomefile},,123)
...
exten = i,1,Playback(invalid)
exten =
Are you using a Gatekeeper (CLAN)?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus
Benngård
Sent: Wednesday, March 10, 2010 1:53 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] 00h323 cant get gatekeeper to
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
Try setting CALLINNUM to ${EXTEN}.
exten = _95040X,1,Set(CALLINNUM=${EXTEN}).
CALLERID(num) instead of CALLINNUM (which is 100% wrong)
--
_
-- Bandwidth and
Does the application PGSQL has been removed from Asterisk? Couldn't find it on
Asterisk source and addons.
Atenciosamente,
Vinícius Fontes
Gerente de Segurança da Informação
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Information Security Manager
Canall
Hello, I am using asterisk 1.6.2.0 and while running the app_meetme I am
finding that all conferences created either statically or in real time are
being closed after one hour. This happened even if a set a limit on the
conference. Is there I default setting somewhere that I need to adjust?
On Wed, Mar 10, 2010 at 2:09 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
exten = _95040X,1,Set(CALLINNUM=${CALLERID(dnid)})
Try setting CALLINNUM to ${EXTEN}.
exten = _95040X,1,Set(CALLINNUM=${EXTEN}).
CALLERID(num) instead of CALLINNUM (which is
On Wednesday 10 March 2010 14:32:56 Vinícius Fontes wrote:
Does the application PGSQL has been removed from Asterisk? Couldn't find it
on Asterisk source and addons.
That application has never been a part of Asterisk in the first place.
--
Tilghman Lesher
Digium, Inc. | Senior Software
Did anyone else just get what looks like a phising attempt pretending to
be from digium?
It appears to be full of links to http://app.en25.com/e/er.aspx
I must admit, it looks genuine.
--
_
-- Bandwidth and Colocation
On Mar 10, 2010, at 5:35 PM, Thomas Kenyon wrote:
Did anyone else just get what looks like a phising attempt pretending to
be from digium?
It appears to be full of links to http://app.en25.com/e/er.aspx
I must admit, it looks genuine.
I suspect you'll find it _IS_ genuine. The en25.com
This is off topic but I have inherited a Nortel Option 11c and am curious
what features you are using that make you want to continue using it in
conjunction with Asterisk as opposed to moving completely over to Asterisk.
I am just learning the Nortel and find it powerful but haven't found the
I am a complete newbie, completed editing the extensions.conf file, having
problem reloading my diaplan via asterisk console, tried to reload it with
diaplan reload command, but it says command does not exist.
Please help
CALLERID(num) instead of CALLINNUM (which is 100% wrong)
How is it wrong if he's creating his own variable named CALLINNUM?
Maybe you are right, it just looked to me like a typo in for the old and
deprecated ${CALLERIDNUM}.
Philipp
--
On Wed, 2010-03-10 at 15:48 -0700, James Noble wrote:
This is off topic but I have inherited a Nortel Option 11c and am
curious what features you are using that make you want to continue
using it in conjunction with Asterisk as opposed to moving completely
over to Asterisk. I am just learning
Citel makes SIP gateways for Nortel handsets that allow you to forklift
the Nortel PBX and drop in Asterisk or any other SIP based platform,
while retaining the Nortel desktop experience for the users.
Cory J Andrews
Carlos Chavez wrote:
On Wed, 2010-03-10 at 15:48 -0700, James Noble
- Tilghman Lesher tles...@digium.com escreveu:
On Wednesday 10 March 2010 14:32:56 Vinícius Fontes wrote:
Does the application PGSQL has been removed from Asterisk? Couldn't
find it
on Asterisk source and addons.
That application has never been a part of Asterisk in the first
place.
Jonathan Addleman wrote:
However, I can't find any way to interact with an existing confbridge
conference. Surely there's some equivalent to meetme's 'meetme list'
command? Anything else I can use through the cli or manager API? I just
need to list conferences and members. Thanks!
Replying
On Wednesday 10 March 2010 16:58:58 ayodele abejide wrote:
I am a complete newbie, completed editing the extensions.conf file, having
problem reloading my diaplan via asterisk console, tried to reload it with
diaplan reload command, but it says command does not exist.
Looks like you're missing
The only thing they will really miss is the MWI light on their phones.
Asterisk supports VMWI Visual MWI in various forms;
From a cheap analog phone point of view:
have a look at 'mwisendtype' in chan_dahdi.conf
line reversal if the phone has a LineReversal LED.
neon high
On Thu, 2010-03-11 at 13:27 +1300, Alec Davis wrote:
The only thing they will really miss is the MWI light on their phones.
Asterisk supports VMWI Visual MWI in various forms;
From a cheap analog phone point of view:
have a look at 'mwisendtype' in chan_dahdi.conf
line
A letter, I spelt it right at the CLI prompt but says it does not recognise the
command
From: tles...@digium.com
To: asterisk-users@lists.digium.com
Date: Wed, 10 Mar 2010 18:24:51 -0600
Subject: Re: [asterisk-users] Diaplan reload command not working
On Wednesday 10 March 2010 16:58:58
What version of asterisk are you using? Dialplan reload wasn't added
until 1.4. If for some reason you have a 1.2 or older asterisk
install, you'll need to use extensions reload (I think, I don't have
a 1.2 box in front of me to confirm the exact command).
Thanks,
--Warren Selby
On Mar
At 4:45 PM on 08 Mar 2010, equis software wrote:
I have this
[TRONCAL-SIP]
exten=225/91,1,Answer
exten=225/91,2,Echo
exten=225/91,3,Hangup
exten=225/92,1,Answer
exten=225/92,2,Playback(conf-invalid)
exten=225/92,3,Hangup
[...]
Dont work
If I add this rule
exten=225,1,Answer
Zoa wrote:
On friday we finally released Attrafax under a GPL2 license.
It comes with its own set of modems and built in transparent gatewaying.
The solution should be quite stable as long as the line quality is ok.
(Some tools for measuring the line quality are included in the release,
as
I am using the 1.6.2.0 version
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Date: Wed, 10 Mar 2010 19:29:02 -0600
Subject: Re: [asterisk-users] Diaplan reload command not working
What version of asterisk are you using? Dialplan reload wasn't added until
1.4. If for some
Hi all,
I am worried because on my production asterisk servers, I am receiving
these errors every 2-3 minutes. my log files are full of them:
WARNING[xxx] app_dial.c: Unable to forward voice or dtmf
and also, less frequent:
WARNING[xxx] app_dial.c: Unable to write frame
How can I find out
We're having an issue that isn't easily googleable so I thought I might might
try here.
We have several customers who want all their extensions to ring on incoming
calls. Frankly I think it is craziness to ring 11 extensions all at once but
that is how they want it.
We're doing this by
On Wed, Mar 10, 2010 at 8:02 PM, ayodele abejide ayodeleabej...@hotmail.com
wrote:
I am using the 1.6.2.0 version
Could you copy and paste your CLI from when you type it in and including the
error message?
From the cli, you should be able to hit your tab key to see a list of all
available
On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote:
This normally works fine but occasionally when someone picks up the call
other phones don't seem to realize the call has been answered and will
continue to ring. On at least once occasion I saw a call that went to
Chris-
Sounds like the Toyota bug has migrated to Asterisk... it's mutated into
runaway ringing :-)
-Jeff
Sorry for my attempt at levity; just couldn't help it plus I'm sure Digium guys
will know how to resolve.
We're having an issue that isn't easily googleable so I thought I might might
On Mar 10, 2010, at 9:05 PM, Warren Selby wrote:
On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote:
This normally works fine but occasionally when someone picks up the call
other phones don't seem to realize the call has been answered and will
continue to ring. On at
I'm experiencing runaway ringing too, can we make this a class action
against someone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
Brower
Sent: Wednesday, March 10, 2010 10:20 PM
To: Chris Owen
Cc:
On Mar 10, 2010, at 10:27 PM, Chris Owen wrote:
This normally works fine but occasionally when someone picks up the
call other phones don't seem to realize the call has been answered
and will continue to ring. On at least once occasion I saw a call
that went to voicemail and all the
Hi all,
I've been trying to add a custom mysql field to my CDR's, but I must
be doing something wrong.
I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I add:
exten = h,1,Set(CDR(q931)=${HANGUPCAUSE})
This extension is executed, I can see it in the asterisk console.
I have added a
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