Thanks Alot @ Jeff LaCoursiere,@Arjan Kroon,@Robin,@Joseph
@ Jeff LaCoursiere
Well you already suggested that you would send all files to server A, so A
is your server
Sorry For the wording actually i need to send to a central server. then a
central server to all others. Because all servers have
With dropbox i mean a service (http://getdropbox.com). I've been thinking
about using dropbox for stuff at my asterisk servers, but haven't done so
yet. It was just an idea that came to mind when reading your question. You
could check out the site though, maybe it is the right solution for you.
I don't know if you server is running under Unix.
If so, here is a wiki link about mounting
http://en.wikipedia.org/wiki/Mount_%28Unix%29
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens ABBAS SHAKEEL
Verzonden: 21-10-2009 08:59
Aan:
while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error;
make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml'
gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a
mxml/libmxml.a -lncurses
make[2]: Leaving directory
After a lot of debugging i have reproduced the error and the behaviour
look me very strage:
i've tried to change dtmfthereresold, vpmdtmfsupport and other kernel
module settings without noting any significative change.
But what i've notice (recording all the IVR calls and then listening
the
hi jeff,
we use much of this phones, but i don't have seen such a symbol. The
only thing i know is when you have an unregistered account (failed or
not reachable) that the phone symbol has a red cross over it, which
means its not online.
Maybe on the phone a user pass has been set?
best regards
Hi list.
Does anyone know how to keep calls between 2 local SIP-phones on the
local private network when the 2 local IP-phones are registered to an
online public Asterisk-server ??
What network-element / router do I need to install to prevent the
RTP-traffic from flowing via the internet ?
Your best option without a local asterisk server is to set up the remote
server to do reinvites when calls are going local-local
The calls will end up routed through your internet router, but not beyond
that.
Downside: might have to make each ip phone available via port forwards
If you're
I'm having loads of problems with recordings, as in crappy audio quality and
lost pieces of the recordings. I've been searching for a solution and the
solutions i find on the interwebs include a ramdisk, for local recording, or
another machine, handling the recording. I guess the ramdisk would be
Hi
Just download tar.gz of your kernel version and extract into
/usr/src/kernels/ directory
!
--
Regards,
Chandrakant Solanki
On Wed, Oct 21, 2009 at 1:34 PM, PATRICK KANGETHE patricemb...@yahoo.comwrote:
while compiling zaptel drivers for my yeaster TDM800 hardware, I get this
- Kevin P. Fleming kpflem...@digium.com ha scritto:
| Da: Kevin P. Fleming kpflem...@digium.com
| A: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
| Inviato: Lunedì, 19 ottobre 2009 14:03:53
| Oggetto: Re: [asterisk-users] Calls hang up after 20
Your best option without a local asterisk server is to set up the
remote server to do reinvites when calls are going local-local
The calls will end up routed through your internet router, but not
beyond that.
So by placing canreinvite=yes in sip.conf, the RTP-traffic would flow
between the
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
Thanks solanki it worked fine.
From: Chandrakant Solanki solanki.chandrak...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, October 21, 2009 1:45:42 PM
Subject: Re: [asterisk-users] error -
There are 2 issues i think, one is the seek time on harddisks and the
lack of a big buffer in Asterisk (saving 10 streams at the same time
will cause a lt of random writes).
The other one is the interrupts being taken up by the harddisk.
So an SSD might help, saving to an network drive
Thanks for your response.
The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)...
But memory is rather cheap nowadays. If i'd buf up the server with 8 extra
gigs for use as a ramdrive, do you think that might be enough to record
between 30-60 simultanious streams? Or should it
Hi there,
I'm Using TxFAX to send faxes via Zaptel PRI. I have 2 PSTN PRI Providers,
with the first provider, all faxes are trasmited fine. With the second
provider, faxes can't be sent, we suspect about the setting of this PRI
provider, perhaps is doing some compression somewhere. Any suggestion
Kevin P. Fleming wrote:
It's not present in the current 1.4 doc/imapstorage.txt file, or any
later version. I don't even know why the storage format would matter,
since that would be very specific to the IMAP server that is managing
that folder.
Hmmm
Martin a écrit :
Ring is the state when the device sent 100 Trying after INVITE
When it actually sends 180 Ringing or gets the progress or so message
from another channel
(when used with Dial) then the status changes to Ringing
Humm. OK. So basically, it's Intended to ring...
Thanks for the
Want to make sure I understand why a caller might not hear ringing when
outbound calling.
A SIP phone is behind a firewall and is registered to an asterisk
server on a public network. Sometimes (but not always) when placing an
outbound call there is no ringing before the remote party
On Tue, 20 Oct 2009, Jimmy Godbout wrote:
Can you send a picture of this ?
Thanks
-Original Message-
From: j...@jeff.net
Sent: Tue, 20 Oct 2009 23:34:13 + (UTC)
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Linksys 962
Working with a new client that has
On Wed, 21 Oct 2009, Stefan Schmidt wrote:
hi jeff,
we use much of this phones, but i don't have seen such a symbol. The
only thing i know is when you have an unregistered account (failed or
not reachable) that the phone symbol has a red cross over it, which
means its not online.
Maybe
Miguel Molina a écrit :
Guillaume Yziquel escribió:
So what is this permission issue? Where are the changes from 1.0 to
1.1 documented?
When I was testing asterisk 1.6.0.X with the AMI Originate action, I
fell into the same issue as you. I found that it was that the
permissions now
Hi list,
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
We have a PRI line,I need to know what are the system requirements and
hardware requirement for Asterisk *IVR*,*OBD*(Outbound
dialer),*IBD*(Inbound dialer).
Thanks and Regards,
Kiran Reddy
On Wed, 21 Oct 2009, kiran.re...@mpowerglobal.in wrote:
I am new to asterisk. I need help for installing and configure Asterisk
IVR,OBD,IBD Server.
4 posts in 3 hours?
1) Don't repost, you just annoy people that may have helped you.
2) Ask specific questions, not I know nothing, please tell
Or charge for full access! Leave a few teasers, and charge some amount to
see them all. I would pay - even close to attendance price... could only
help you get past break even ;)
I agree, I would be quite willing to pay for full access to all the videos from
the Conference.
Bob
I am going to try to get a picture taken of this odd icon, since I haven't
actually seen it myself yet. It may become obvious once I have... Its not
that the phone isn't registered - in fact it doesn't seem to stop them
from using the phone at all...
Just because they can use the
Date: Tue, 20 Oct 2009 21:02:29 -0500
From: asteriskl...@callthem.info
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] troubleshooting NAT
if you're using SIP then you look at SIP headers ... SDP part
from INVITE's and 200 OK to INVITE. You check what IP/port is used
On Wed, Oct 21, 2009 at 4:01 PM, Bob Pierce pier...@westmancom.com wrote:
Or charge for full access! Leave a few teasers, and charge some amount to
see them all. I would pay - even close to attendance price... could only
help you get past break even ;)
I agree, I would be quite willing to
Here is what i think the is helpful from wireshark
OPTIONS sip:216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK5aac5845;rport
From: Unknown sip:unkn...@mypublicip;tag=as7b5287b3
To: sip:216.82.224.202
Contact: sip:unkn...@mypublicip
Call-ID:
Hello,
I have :
answeronpolarityswitch=yes
on chan_dahdi.conf
but it's making all my lines answer on polarity reversal, this causes a
problem for PSTN lines, so how can I set these lines to answer immediately
(when it rings)?
thanks
Just looking for some ideas here...
Single office with 1.4.26.2 - Frontend 1.4.26.2 w/sangoma A108 Gateway
I have been getting a few complaints about caller cant hear me or I
cant hear the caller I've listened to the recordings and can verify
what they are complaining about, with this being
Hello,
We use RAM to record to on almost all systems we set up, although we
usually use tmpfs, instead of a fixed RAM drive, because it is more
flexible.
The number of recordings you can handle is dependant on how long the
calls are. What would your average, minimum, maximum recording lengths
Hi Matt,
ain't you the vicidial guy? I'm actually trying to get this stuff fixed on a
vicidial system.
Anyway, the minimum length is 10-20 seconds, maximum can get as long as
15-20 minutes, and on average it's about 2-5 minutes, depending on the
campaign.
The server is now doing everything btw,
Have a quick look at this guide on NAT and SIP -
http://www.aocomputing.net/?p=3. This is the link given if you were to ask
this same question in the IRC channel...
--wcs
On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose sixfourimp...@hotmail.com wrote:
Here is what i think the is helpful from
Hello,
Yep, I'm the ViciDial Guy :)
In our most recent release we do have some instructions in the
SCRATCH_INSTALL.txt doc on setting up a tmpfs partition for recording.
8GB should be fine for the 60 concurrent recordings under the times
you gave, although with MySQL and Apache/PHP you may run
I'm assuming this is an issue with DAHDI. I am running asterisk 1.4.26
on Fedora 11 with dahdi-linux kernel modules 2.2.0.2-65 (both from
ATrpms). I have a Wildcard TDM400P REV I (4 modules) with one POTS
line and three local extensions (never can remember which is FXS and
which is FXO )-: and a
I'm on it, going to get me some new hardware tomorrow and hope to have it up
and running early next week.
tnx!
On Wed, Oct 21, 2009 at 17:42, Matt Florell astma...@gmail.com wrote:
Hello,
Yep, I'm the ViciDial Guy :)
In our most recent release we do have some instructions in the
i changed my sip_nat.conf file following the steps in that link. Still didn't
work same debug info
Date: Wed, 21 Oct 2009 10:33:18 -0500
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] troubleshooting NAT
Have a quick look at this guide on NAT and
Hi,
Siemens Gigaset line of products include an integrated web browser with
which firmware download is possible.
The trouble is you need to provide Internet access.
We use a couple of these boxes in LANs not connected to Internet for
security reasons.
So I would prefer to download firmware
Olivier wrote:
Hi,
Siemens Gigaset line of products include an integrated web browser with
which firmware download is possible.
The trouble is you need to provide Internet access.
We use a couple of these boxes in LANs not connected to Internet for
security reasons.
So I would prefer
Have you considered rsync? We use it to synchronize voicemail between
offices connected through a VPN. Of course you need to run rsync somehow,
which is easy with an external command every time someone checks their voice
mail, but no reason it couldn't be done with a cron job.
Sincerely,
Randy R wrote:
I missed the first part of this, but has anyone said: not all the
presentations were recorded.
Hi Randy.
Yes, that was mentioned. Actually, three of the four tracks were
videotaped IIRC.
Barry
___
-- Bandwidth and Colocation
Your best option without a local asterisk server is to set up the remote
server to do reinvites when calls are going local-local
The calls will end up routed through your internet router, but not beyond
that.
So by placing canreinvite=yes in sip.conf, the RTP-traffic would flow
between
Thanks for the information, I will look into both cisco and adtran see which
would be helpful
On Thu, Oct 15, 2009 at 4:09 PM, Alex Balashov abalas...@evaristesys.comwrote:
David Backeberg wrote:
On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com
wrote:
There's no
On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline blkl...@attglobal.net wrote:
Randy R wrote:
I missed the first part of this, but has anyone said: not all the
presentations were recorded.
Hi Randy.
Yes, that was mentioned. Actually, three of the four tracks were
videotaped IIRC.
Barry
Is THAT a summary :)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R
Sent: Wednesday, October 21, 2009 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Hi,
I tried getting our server setup for 400-500 simultaneous calls, calls were
going through properly but at around 200-250 calls, mysql (connect ...)
statement was taking at least 5-10 sec to connect to the database. I
optimized all possible parameters in my.cnf:
max_connection=1000
On Wed, Oct 21, 2009 at 7:36 AM, Robin ro...@zoap.org wrote:
Thanks for your response.
The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)...
But memory is rather cheap nowadays. If i'd buf up the server with 8 extra
gigs for use as a ramdrive, do you think that might be
Hi,
How can I integrate Asterisk to Nuance TTS engine instead of Cepstral?
Has anybody done this? How is the architecture and can Java AGI be used to
communicate between them?
regards,
Vela Sivasankaran
___
-- Bandwidth and Colocation Provided by
B.Masoud @ SH wrote:
Hello,
I have :
answeronpolarityswitch=yes
on chan_dahdi.conf
but it's making all my lines answer on polarity reversal, this causes
a problem for PSTN lines, so how can I set these lines to answer
immediately (when it rings)?
thanks
According to asterisk-guru this has been done. If you're just looking for
TTS and not voice recognition, this shouldn't be too problematic.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vela
Sivasankaran
Sent: Wednesday,
Hi,
I think you should use the nvcmdline utility to synthesize your prompt
to a certain file to be specified. Afterwards, you could play that on
your asterisk, for example a wav file. But this could be some kind of
long lasting as the TTS synthesizes in realtime, i.e. the longer the
prompt is
On 10/21/09, David Backeberg dbackeb...@gmail.com wrote:
On Wed, Oct 21, 2009 at 7:36 AM, Robin ro...@zoap.org wrote:
Thanks for your response.
The hardware I have now is not sufficient to set up a ramdisk (just 4
gb)...
But memory is rather cheap nowadays. If i'd buf up the server
On Wed, 21 Oct 2009, das sandesh wrote:
I tried getting our server setup for 400-500 simultaneous calls, calls
were going through properly but at around 200-250 calls, mysql (connect
...) statement was taking at least 5-10 sec to connect to the database.
I optimized all possible parameters
On Wed, 21 Oct 2009, Christophorus Laube wrote:
Using the nvcmdline utility you should use bash AGI or something more
scripty.
I'd suggest something way less scripty like C and a proper API if
available.
You can execute xxx AGIs written in C in the time it takes a PHP or Perl
interpreter
Hello Team
I have connected * running centos 5.2, asterisk 1.6.1 dahdi 2.1 to the
telco but the link is very unstable (D-Channel restart after some few min)
Below please find part of 'pri intensive debug span 2' for your advice.
Looks like telco is sending disconnect request but cant establish
On Wed, Oct 21, 2009 at 2:30 PM, das sandesh sandesh...@gmail.com wrote:
I tried getting our server setup for 400-500 simultaneous calls, calls were
going through properly but at around 200-250 calls, mysql (connect ...)
statement was taking at least 5-10 sec to connect to the database. I
Sounds like it wasn't a very interesting track. ;)
N.
Danny Nicholas wrote:
Is THAT a summary :)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R
Sent: Wednesday, October 21, 2009 1:24 PM
To:
Hi Steve,
Thanks for your reply.
I am using only asterisk code (dial plan) in extensions.conf which also
includes connection to the database: like
exten =n,1, MYSQL(connect connid ipaddr uname pwd database) and
then the required select queries and the clear and Disconnect the
connection.
I think the key point is how many calls per second. That's what mysql is
concerned about. Other than that it is just asterisk. Did you monitor the
mysql, try log-slow-queries and set the time to 1 second.
-Jai
On Wed, Oct 21, 2009 at 12:57 PM, das sandesh sandesh...@gmail.com wrote:
Hi Steve,
Not my cup of tea, but I think I'd be trying an ODBC connection to reduce
some overhead here.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of das sandesh
Sent: Wednesday, October 21, 2009 2:58 PM
To: Asterisk Users Mailing
On Wed, 21 Oct 2009, Danny Nicholas wrote:
Not my cup of tea, but I think I'd be trying an ODBC connection to reduce
some overhead here.
[snip]
Does that reduce overhead or add it? Seems that direct mysql-client code
should be more efficient than adding ODBC in the middle...
j
Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a
multi-tenant environment with IMAP voice mail storage on Zimbra. One of
our clients is having a problem when transferring voice mails from one
mailbox to another (option 8 in the standard voice application menu)
using their Snom 320 and
I don't use ODBC or MYSQL, but the problem the OP mentions is that MYSQL
takes .X seconds longer each time he calls it until it takes 5-10 seconds to
connect on the 100th call. I know some guru out there is probably handling
1000 calls using a MYSQL database, so maybe yall can tell OP what is
On 22/10/09 7:30 AM, das sandesh wrote:
Hi,
I tried getting our server setup for 400-500 simultaneous calls, calls
were going through properly but at around 200-250 calls, mysql (connect
...) statement was taking at least 5-10 sec to connect to the database.
I optimized all possible
On 22/10/09 9:16 AM, Jeff LaCoursiere wrote:
On Wed, 21 Oct 2009, Danny Nicholas wrote:
Not my cup of tea, but I think I'd be trying an ODBC connection to reduce
some overhead here.
[snip]
Does that reduce overhead or add it? Seems that direct mysql-client code
should be more
On 22/10/09 8:56 AM, David Backeberg wrote:
On Wed, Oct 21, 2009 at 2:30 PM, das sandeshsandesh...@gmail.com wrote:
I tried getting our server setup for 400-500 simultaneous calls, calls were
going through properly but at around 200-250 calls, mysql (connect ...)
statement was taking at least
It should be reproducible in some way, how was asterisk installed on the
server its having a problem? If its from source compare the
apps/app_voicemail.c from whats in production with whats getting compiled in
the lab.
when imap is used only one format is stored
you could specify just one format:
I'm sorry - by the lab I meant the end points - it is the same server.
I was not aware that IMAP only stored one format. If I change the
setting in voicemail.conf, do I still have to worry about the grievous
warning message about being sure to delete all messages not using that
format? I would
On Wed, 21 Oct 2009, das sandesh wrote:
I am using only asterisk code (dial plan) in extensions.conf which also
includes connection to the database: like exten =n,1,
MYSQL(connect connid ipaddr uname pwd database) and then the required
select queries and the clear and Disconnect
Hi Matt,
I already used the tuning-primer.sh script to enhance the values for the
parameters, but still it was being slow to connect when there are lot of
calls (calls around 150-200 calls). Also I reduced mysql queries in the code
as well as many other steps, but only problem coming is with
On Wed, 21 Oct 2009, Steve Edwards wrote:
I'd take a look at using AGIs written in C. They make nice little
building blocks. They execute very quickly and can cleanup your
dialplan.
And you can debug them (AGIs in any language) from the command line
completely outside of Asterisk.
--
It's not caller ID issue,
I can make asterisk answer the line by omitting the line
answeronpolarityswitch=no , but this will take effect on all 24 TDM
channels, I want some to have answer on polarity, and some without polarity.
Thanks.
From: asterisk-users-boun...@lists.digium.com
If you're using file storage and specify three formats, app_voicemail will
save to those formats.
The dire warning is because when renaming (for example listening to
new/msg and it gets moved to old messages) and deleting files,
app_voicemail only touches the formats in the configuration file.
On 22/10/09 10:57 AM, das sandesh wrote:
Hi Matt,
I already used the tuning-primer.sh script to enhance the values for the
parameters, but still it was being slow to connect when there are lot
of calls (calls around 150-200 calls). Also I reduced mysql queries in
the code as well as many
The thing is, concurrent calls won't make any difference, it's the calls
per second.
And really you're unlikely to use too many queries per sec.
Exactly and you can see the slow-log-queries if mysql is taking time.
-Jai
On Wed, Oct 21, 2009 at 3:51 PM, Matt Riddell li...@venturevoip.com
Barry L. Kline wrote:
Kevin P. Fleming wrote:
It's not present in the current 1.4 doc/imapstorage.txt file, or any
later version. I don't even know why the storage format would matter,
since that would be very specific to the IMAP server that is managing
that folder.
Hmmm
Steve Edwards asterisk@sedwards.com wrote:
On Wed, 21 Oct 2009, Steve Edwards wrote:
I'd take a look at using AGIs written in C. They make nice little
building blocks. They execute very quickly and can cleanup your
dialplan.
And you can debug them (AGIs in any language) from the
On 22/10/09 1:41 PM, cov...@ccs.covici.com wrote:
Steve Edwardsasterisk@sedwards.com wrote:
On Wed, 21 Oct 2009, Steve Edwards wrote:
I'd take a look at using AGIs written in C. They make nice little
building blocks. They execute very quickly and can cleanup your
dialplan.
And you
On Wed, 21 Oct 2009, Steve Edwards wrote:
I'd take a look at using AGIs written in C. They make nice little
building blocks. They execute very quickly and can cleanup your
dialplan.
And you can debug them (AGIs in any language) from the command line
completely outside of Asterisk.
On
Steve Edwards asterisk@sedwards.com wrote:
On Wed, 21 Oct 2009, Steve Edwards wrote:
I'd take a look at using AGIs written in C. They make nice little
building blocks. They execute very quickly and can cleanup your
dialplan.
And you can debug them (AGIs in any language) from
Steve Edwards asterisk@sedwards.com wrote:
Since I'm an old-school C programmer, I use emacs as my editor. I fire
up gdb (the GNU C (amongst other languages) debugger) in a window, give it
a command like b main; r dummy-input-for-block-ani and I can step
through my program line by line,
Hey now, I'm a newschool programmer and I use vim (and vi, when necessary).
Andrew
On Wed, Oct 21, 2009 at 8:02 PM, Jeff LaCoursiere j...@jeff.net wrote:
Steve Edwards asterisk@sedwards.com wrote:
Since I'm an old-school C programmer, I use emacs as my editor. I fire
up gdb (the GNU C
On 22/10/09 2:54 PM, cov...@ccs.covici.com wrote:
OK, but how do write the C program -- the Perl and php agis have defined
functions for the agi commands, how do you do this in c?
There is a library (haven't used it myself)
http://sourceforge.net/projects/cagi/
Basically you read from the
Jeff LaCoursiere wrote:
Steve Edwards asterisk@sedwards.com wrote:
Since I'm an old-school C programmer, I use emacs as my editor. I fire
up gdb (the GNU C (amongst other languages) debugger) in a window, give it
a command like b main; r dummy-input-for-block-ani and I can step
Folks,
Not sure what's going on, but suddenly Asterisk 1.6.1.6 is crashing,
usually when I exit the console or use asterisk -rx. The sip peers
entry always shows duplicate entries (once I had an extension over
half a dozen times) just before it crashes.
3182/3182 172.17.0.126
On Wednesday 21 October 2009 15:16:31 Jeff LaCoursiere wrote:
On Wed, 21 Oct 2009, Danny Nicholas wrote:
Not my cup of tea, but I think I'd be trying an ODBC connection to reduce
some overhead here.
[snip]
Does that reduce overhead or add it? Seems that direct mysql-client code
should
On Wed, 21 Oct 2009, cov...@ccs.covici.com wrote:
OK, but how do write the C program -- the Perl and php agis have defined
functions for the agi commands, how do you do this in c?
The same way. All languages need a library. Either you find a library that
talks AGI or you write one. I wrote
I am testing an ivr but I'm having problems. The call keeps looping and it
doesn't hangup the call after passing three times through the menu. Here's my
conf:
exten = s,n,NoOp(Here's Count)
exten = s,n,NoOp(${COUNT})
;123,n,Set(COUNT=$[${COUNT} - 1])
exten = s,n,GotoIf($[${COUNT} =
On Wed, 21 Oct 2009, Landy Landy wrote:
I am testing an ivr but I'm having problems. The call keeps looping and
it doesn't hangup the call after passing three times through the menu.
When it enters extension 33 it should hangup the call but, if the caller
stays on the line the exten =
2009/10/21 Leif Madsen leif.mad...@asteriskdocs.org
Olivier wrote:
Hi,
Siemens Gigaset line of products include an integrated web browser with
which firmware download is possible.
The trouble is you need to provide Internet access.
We use a couple of these boxes in LANs not
2009/10/21 Christophorus Laube christophorus.la...@semanticedge.de
I think you should use the nvcmdline utility
Is this nvcmdline bundled with every Nuance TTS ?
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asterisk-users
There were 2 problems that we faced, one was at around 50 calls, few calls
were just dead air, and when I saw the logs I could see that it was sent to
the sip provider and after that there was no log for that particular call
that was having dead air, but at around 200 to 250, we could see that
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