Hi Steve;
Currently with Dahdi, for the below configuration that we were using it in the
rc.local, how it will be?
I think there should be something new to be used instead of ztcfg, what it is?
And what about other lines? They need to be changed?
touch /var/lock/subsys/local
/sbin/modprobe
Hi,
After upgrading to 1.6.x and hdvoice (g722) polycome phones I am
wondering how to optimize asterisk sounds and music on hold to take
advantage of that codec. I often listen to a special music extension on
my headset:
/usr/bin/wget -q -O - http://music.example.com | /usr/bin/madplay -Q -z -o
I have intalled Asterisk 1.4.20.1, it saves the voicemails wavs into a NFS
mounted directory and for an unknow reason all messages over 10 seconds was
recorded incorrectly, but if i save to a local directory works fine.
somebody can help me?
Thanks.
Ernesto
asterisk xload schrieb:
I have intalled Asterisk 1.4.20.1, it saves the voicemails wavs into a NFS
mounted directory and for an unknow reason all messages over 10 seconds was
recorded incorrectly, but if i save to a local directory works fine.
What exactly do you mean by incorrectly?
asterisk xload schrieb:
I have intalled Asterisk 1.4.20.1, it saves the voicemails wavs into a NFS
mounted directory and for an unknow reason all messages over 10 seconds was
recorded incorrectly, but if i save to a local directory works fine.
somebody can help me?
Thanks.
Ernesto
The first 10 seconds are recorded correctly but the rest of the message
seems to be recorded faster.
2009/6/10 Philipp Kempgen philipp.kemp...@amooma.de
asterisk xload schrieb:
I have intalled Asterisk 1.4.20.1, it saves the voicemails wavs into a
NFS
mounted directory and for an unknow
I need attended transfers, but I do not have time to talk to another
extension and see if they accept the transfer, my features.conf is:
[general]
parkext = 700 ; What ext. to dial to park
parkpos = 701-720 ; What extensions to park calls on
context = parkedcalls
thank you very much, i will test tonight and informs you tomorrow morning if
it works correctly.
Ernesto.
2009/6/10 Stefan Schmidt s...@sil.at
asterisk xload schrieb:
I have intalled Asterisk 1.4.20.1, it saves the voicemails wavs into
a NFS
mounted directory and for an unknow reason
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box for the motherboard is too small.
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8
Stefan Schmidt schrieb:
asterisk xload schrieb:
I have intalled Asterisk 1.4.20.1, it saves the voicemails wavs into a NFS
mounted directory and for an unknow reason all messages over 10 seconds was
recorded incorrectly, but if i save to a local directory works fine.
you should start
On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card - basically the box for the motherboard is too small.
You can probably find
Had a fairly horrible lightning storm night before last, and four of eight
ports in a 1.4.20 machine stopped answering.
In the CLI:
budsw*CLI zap show channels
Chan Extension Context Language MOH Interpret
pseudodefault en default
1
Hello,
I'm supporting an Asterisk setup in the Cayman Islands(I'm in Canada), which is
running a PRI to a local telecom provider. We are looking at improving the
setup, setting up high availability etc. My manager is interested in putting a
TDMOE device in place, so we can easily switch the
But I wonder why there is a problem with writing recordings to an
NFS mount directly. NFS should easily handle that.
hello philipp,
i dont know why this is a problem with nfs, but i had the same issue
with two servers behind one switch. So i know what helps.
I think that NFS had a problem
Hi All,
I'm looking for how to enable SIP Markers, or specifically, how to have
the TIME reset when a call route changes.
I'm debugging an issue, where a sip client we have switching to
one-way-audio, when an asterisk server fruther down the call path dials
out to the PSTN. Scenario is:
(resend as apparently I was blocked)
Hi All,
I'm looking for how to enable SIP Markers, or specifically, how to have
the TIME reset when a call route changes.
I'm debugging an issue, where a sip client we have switching to
one-way-audio, when an asterisk server fruther down the call
Hey Everyone,
i am using Asterisk 1.4.21.1 with a old Patton SN1400 and some SNOM
300,320,360 Devices.
In the combination with asterisk and the patton, there are occuring some
strange behaviour, due to the calling and answering everything works
good, clear voice, great availability.
All the
I have quite an old version of Chameleon Mail, currently the prompts played
when leaving a message are –
-- Executing VoiceMail(SIP/209-3b0e, u5) in new stack
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'vm-isunavail' (language
Stefan Schmidt schrieb:
But I wonder why there is a problem with writing recordings to an
NFS mount directly. NFS should easily handle that.
i dont know why this is a problem with nfs, but i had the same issue
with two servers behind one switch. So i know what helps.
I think that NFS had a
You could just do voicemail(s5). That should just play the beep and record
the message.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Razza
Sent: Wednesday, June 10, 2009 9:01 AM
To: asterisk-users@lists.digium.com
Hello,
i've recently configured my asterisk for internal sip calls.
while testing, i noticed that 1 out of 10 calls works..
at first i thought my router dropping packets around the way as it were a
bottle neck..
so i've added a switch.
once i tested again same prob occurs...
im using xlite as
On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes
Pereiragomespere...@startel.pt wrote:
Hello to all
I'm trying to record the calls going to my queues, but asterisk creates
2 files, one with the inbound and another with the outbound sound.
I know Sox should mix the 2 files automatically in the end,
Hi,
we're using the limit option like this:
Dial L(6:3)
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] --
Limit Data for this call:
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41]
timelimit = 6
[Jun 10 16:14:41] VERBOSE[12196]
Hello,
I am looking for a dialer program, free or not, that allows me to perform
scheduled calls, generate reports and let me upload sound files. Is there
something that fits these features?.
If there is not any product like I mentioned before I am interested to build
this kind of software but I
Does asterisk support T38 passthrough now? What version onwards?
ANy ideas on how to configure it for a host?
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Hola Carlos,
Have you searched for ViciDialer? It's a good one.
Give it a shot, it might be what you are looking for.
Carlos Ruiz Diaz wrote:
Hello,
I am looking for a dialer program, free or not, that allows me to
perform scheduled calls, generate reports and let me upload sound files.
Thank you Jose.
Interesting suggestion!
Is there any other?
On Wed, Jun 10, 2009 at 11:21 AM, Jose P. Espinal j...@slackware-es.comwrote:
Hola Carlos,
Have you searched for ViciDialer? It's a good one.
Give it a shot, it might be what you are looking for.
Carlos Ruiz Diaz wrote:
Klaus Darilion wrote:
Steve Underwood schrieb:
There seems to be a common misconception about 488. It represents an
irrevocable failure of the call. Once a 488 is sent the call is
essentially dead. A number of systems are able to continue beyond a 488,
and allow further
Jay Ray wrote:
Does asterisk support T38 passthrough now? What version onwards?
I thought it came in at 1.6.0 .
ANy ideas on how to configure it for a host?
There are lots of guides to this on t'internet.
Jay Ray schrieb:
Does asterisk support T38 passthrough now? What version onwards?
Since 1.4
ANy ideas on how to configure it for a host?
see sip.conf und search for 38 or udptl.
you should also look at udptl.conf and configure these ports in the firewall
regards
klaus
David Backeberg escribió:
On Sun, Jun 7, 2009 at 12:51 PM, Joao Gomes
Pereiragomespere...@startel.pt wrote:
Hello to all
I'm trying to record the calls going to my queues, but asterisk creates
2 files, one with the inbound and another with the outbound sound.
I know Sox should mix the 2
Thomas Kenyon escribió:
Jay Ray wrote:
Does asterisk support T38 passthrough now? What version onwards?
I thought it came in at 1.6.0 .
T.38 passthrough was introduced in 1.4. On 1.6.X, asterisk supports T.38
fax sending and receiving too (I think it's called app_fax).
ANy
Thanks kindly works a treat :o)
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Klaus Darilion wrote:
Steve Underwood schrieb:
There seems to be a common misconception about 488. It represents an
irrevocable failure of the call. Once a 488 is sent the call is
essentially dead. A number of systems are able to continue beyond a 488,
and allow further
There is also GNUdial but i would prefer VICIdial anyday over it ( personal
opinion :) ) .
On Wed, Jun 10, 2009 at 9:43 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com
wrote:
Thank you Jose.
Interesting suggestion!
Is there any other?
On Wed, Jun 10, 2009 at 11:21 AM, Jose P. Espinal
I can't find GNUDial web page :(
On Wed, Jun 10, 2009 at 1:44 PM, Jaswinder Singh vick...@gmail.com wrote:
There is also GNUdial but i would prefer VICIdial anyday over it ( personal
opinion :) ) .
On Wed, Jun 10, 2009 at 9:43 PM, Carlos Ruiz Diaz
carlos.ruizd...@gmail.com wrote:
Thank
As you can see below I am striping off the 8 before it ever goes to CCM
in the extensions.conf file.
exten = _8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1...@172.16.200.10:1720)
I have the H323 gateway in CCM configured to use the same Calling Search
Space as my phone extensions.
Jimmy Ezell
Carlos Ruiz Diaz escribió:
I can't find GNUDial web page :(
It looks like the www.gnudialer.org is down. However, the sources are
still in the same place:
http://dynx.net/ASTERISK/gnudialer/
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
It's not clear where the HA comes in. Can you explain?
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw
Sent: Wednesday, June 10, 2009 8:19 AM
To: Asterisk Users Mailing List -
I'm experiencing some problem using the transfer()
application,expecially when a call in received from a queue.
I'm using Asterisk 1.4.22.1
This is my scenario:
; this is the piece of code in extensions.conf that place the call in
the queue when is called
exten = ,1,Answer
exten =
On Wed, Jun 10, 2009 at 08:44:22AM -0400, David Backeberg wrote:
On Wed, Jun 10, 2009 at 7:17 AM, Alex Samada...@samad.com.au wrote:
Hi
recently bought a soekris net5501 and a tdm410 to place in there.
I am having some issues attaching 12V power to the card via the molex
card -
Xorcom makes Astribank devices that have two USB connections so one can go
to one system and one can go to a backup system.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Frank Bulk frnk...@iname.com
Organization: iName.com
Reply-To: frnk...@iname.com, Asterisk
Hi all,
I was wondering what the current development plans / patches etc are to
allow Asterisk to talk to Exchange 2007 Unified Messaging with respect
to adding SIP over TCP support?
I've been googleing and looking through various posts on the wiki and
all seem to suggest that it could be
Alex Samad wrote:
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8 just behind the molex connector on the tdm410.
can any one here confirm this, or have any info on the pwr2400b - ie how
it connects to the cards. The web site is a bit devoid of
Hi all,
Previously, I had the PrivacyManager working for me exactly as would
be expected, but after upgrading the OS to Debian lenny and Asterisk
to v1.4.21.2 that's no longer the case. Anonymous callers are still
confronted with the PrivacyManager, but now no matter what I set the
On Wed, Jun 10, 2009 at 05:49:22PM -0500, Kevin P. Fleming wrote:
Alex Samad wrote:
I have read up about a PWR2400b and it seems to use 2wire pin, I am
guessing to connect to P8 just behind the molex connector on the tdm410.
can any one here confirm this, or have any info on the
Hi All;
In dahdi: what we use instead of ztcfg -vv (that is existed /sbin/ztcfg -vv).
?
Regards
Bilal
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Try: dahdi_cfg. It works on the same principle.
~:# ls -w 5 /usr/sbin/dahdi_*
/usr/sbin/dahdi_cfg
/usr/sbin/dahdi_genconf
/usr/sbin/dahdi_hardware
/usr/sbin/dahdi_monitor
/usr/sbin/dahdi_registration
/usr/sbin/dahdi_scan
/usr/sbin/dahdi_speed
/usr/sbin/dahdi_test
bilal ghayyad wrote:
Hi All;
On Wed, Jun 10, 2009 at 6:00 PM, Waynewa...@planetwayne.com wrote:
Hi all,
I was wondering what the current development plans / patches etc are to
allow Asterisk to talk to Exchange 2007 Unified Messaging with respect
to adding SIP over TCP support?
I would ask the question the other way
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