I do not think so. Nuance Vocalizer follows RealSpeak (4.5) which based
on some older products. I never have used RealSpeak on the command line,
but I think the "standard" tool is what you should be looking for. Some
years ago Nuance bought Rhetorical with their rVoice TTS. This had a
command l
Hello,
I'm wondering if I can take benefits of long prompts to compute in the
background the next step to be performed by Asterisk.
Do you know what will be the behavior of asterisk if I send a STREAM
FILE command immediately followed by another command ? Will asterisk
stack commands or will it s
On 22/10/09 6:52 PM, das sandesh wrote:
> 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,
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
MySQ
2009/10/21 Christophorus Laube
> I think you should use the nvcmdline utility
>
Is this nvcmdline bundled with every Nuance TTS ?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or up
2009/10/21 Leif Madsen
> 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
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 =>
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} = 4]?33,
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 wrot
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
>
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
Jeff LaCoursiere wrote:
>> Steve Edwards 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 >> through my program line by line, examining and cha
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 st
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 wrote:
>
>> Steve Edwards 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)
> Steve Edwards 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 > through my program line by line, examining and changing variables at will.
>>
Bah. If
Steve Edwards 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 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
On 22/10/09 1:41 PM, cov...@ccs.covici.com wrote:
> Steve Edwards 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
Steve Edwards 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 command line
>
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
" 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 wrote:
> On 22/10/09 1
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 ma
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.
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
[mailto:
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.
--
T
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 repe
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 uname pwd database) and then the required
> select queries and the clear and Disconnect the
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 th
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:
f
On 22/10/09 8:56 AM, David Backeberg wrote:
> On Wed, Oct 21, 2009 at 2:30 PM, 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 c
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
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 par
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 hose
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
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
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 L
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 wrote:
> Hi Steve,
>
> Thanks for your
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 uname pwd database) and
then the required select queries and the clear and Disconnect the
connection.
When
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
>
On Wed, Oct 21, 2009 at 2:30 PM, 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 possibl
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 re
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
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 paramet
On 10/21/09, David Backeberg wrote:
> On Wed, Oct 21, 2009 at 7:36 AM, Robin 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
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
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,
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)?
>
>
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 ht
On Wed, Oct 21, 2009 at 7:36 AM, Robin 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 enough to r
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
wait_timeou
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] A
On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline 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
And I wa
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 wrote:
> David Backeberg wrote:
> > On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg
> wrote:
> >> There's no one-step solution I'm aware of. Cisco sells
>
> 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
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 Pro
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,
Brent
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
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 upgrad
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 S
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 wrote:
> Hello,
>
> Yep, I'm the ViciDial Guy :)
>
> In our most recent release we do have some instructions in the
> SCRATCH_INSTALL.txt do
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
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
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 wrote:
>
> Here is what i think the is helpful from wireshark
>
>
>
> OPTIONS s
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,
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
be?
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 bei
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
_
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" ;tag=as7b5287b3
To:
Contact:
Call-ID: 311d57516ef9649b7dfab93720393...@mypublicip
CSeq: 102 OPTIONS
User-Agent: Asteris
On Wed, Oct 21, 2009 at 4:01 PM, Bob Pierce 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 pay for full a
> 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
>
>
>
>
>
>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 ca
> 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
___
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 te
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
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
> permissi
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.
>
>
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
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 answer
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
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
http://markmail.org/message/up3rf
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
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
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 mig
Thanks solanki it worked fine.
From: Chandrakant Solanki
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wed, October 21, 2009 1:45:42 PM
Subject: Re: [asterisk-users] error - sources for the 2.6.18-92.1.22.el5xen
kernel
Hi
Just download
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
> 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
betw
- "Kevin P. Fleming" ha scritto:
| Da: "Kevin P. Fleming"
| A: "Asterisk Users Mailing List - Non-Commercial Discussion"
| Inviato: Lunedì, 19 ottobre 2009 14:03:53
| Oggetto: Re: [asterisk-users] Calls hang up after 20 seconds
|
| SIP wrote:
|
| > In an ideal world, when Asterisk sent a
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 wrote:
> while compiling zaptel drivers for my yeaster TDM800 hardware, I get this
> error;
>
> make[3]:
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 t
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 reall
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 ?
Confi
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
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 regis
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 `/usr/src/zaptel-
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: Aster
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.
On
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
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