Olivier schrieb:
> Here http://www.voip-info.org/wiki/view/Asterisk+database , you can read:
> "Also, since it's a normal Berkely db1 (version185) file its contents can be
> viewed/dumped with the standard *db1_dump185* tool. Thus db1_dump185 -p
> /var/lib/asterisk/astdb will show the complete data
Some of us have specific ideas about this (hi dave), whatever yours
are they'd be appreciated in the conference.
Also, ZipDX is again letting us use their g722 bridge connected to our
Talkshoe bridge.
Was hoping for a Skype bridge from Steve Sokol's beta, but no news from him ATM.
IRC: #voip-use
Ok, I found the problem. I suggested that I disabled completely my
shorewall-firewall, because there were no rules loaded. But I were
mistaken... shorewall loads some kernel-modules, especially ip_nat_sip
and ip_conntrack_sip, and these modules interfere with asterisk!
http://www.mail-archive.
Hi all,
Does anyone now if it is possible to make Re-INVITE work (canreinvite=yes) at
the same time that arguments T,t (transfer) are used in the Dial() command.
I know that in order to receive the # signal during service invocation the
Asterisk server needs to be in the media path, bu
Hi,
Whenever Asterisk restarts, some values such as DEVSTATE ones, should be
reset.
How would you proceed to have those settings done anytime Asterisk restarts
?
I was thinking of using #exec statement in appropriate .conf file.
Ant better idea ?
Regards
_
IMO this would be elegant, but others probably view it as a hack. (That's
ok; 30 years ago I was a hacker, now I'm just a hack).
If you are starting asterisk as a service, simply create a call file to
reset the DEVSTATE values and copy it when starting asterisk.
I've got a call file that I us
On Fri, Jan 30, 2009 at 03:08:52PM +0100, Olivier wrote:
> Hi,
>
> Whenever Asterisk restarts, some values such as DEVSTATE ones, should be
> reset.
> How would you proceed to have those settings done anytime Asterisk restarts
> ?
Sounds racy. #exec is called whenever configuration is parsed (on
Hi,
I am using asterisk version 1.4.22.1 on a centos 5.2 machine.
Is there any way to run a script somebody puts the call on/off hold ? The
script must be run both on hold and off hold.
Best ragards.
Idris
___
-- Bandwidth and Colocation Provided by h
Matt Florell wrote:
> Yep, my bad I found them once I searched with the dash '-' after the
> 1.4.23. They were lost in the flood of users list mail in my inbox.
>
> I wonder if these could also be posted on the asterisk-announce list
> more consistently? I see a few releases on the announce list,
Idris AVCI schrieb:
> I am using asterisk version 1.4.22.1 on a centos 5.2 machine.
> Is there any way to run a script somebody puts the call on/off hold ? The
> script must be run both on hold and off hold.
You'd see an event on the manager interface I guess.
http://www.voip-info.org/wiki/view/
The dialplan AFAIK doesn't cover HOLD handling. If you can spare the
overhead, you can make a daemon to watch hints and run a script whenever the
hint for a line goes to hold and changes from hold to inuse. Just run
"asterisk -rx "core show hints"" and "asterisk -rx "core show channels"" and
inte
Funny how a topic will come up that you have never dealt with before, and
suddenly it comes up from multiple directions at the same time. I was
recently involved in a meeting where TAPI (which I understand only
vaguely) was proposed as way to link a custom application to Asterisk for
outbound
Click-to-dial in Outlook is a feature of trixbox PRO. HUD does have a TAPI
interface. You can even click-to-dial any phone number that shows up in a
browser window.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: aste
hI,
Trying to understand how to setup two PRIs in sip.conf. Using Asterisk
1.4.23.
I have a provider giving me two PRI (different rate centers) through SIP.
Both PRI comes in from the same IP on the provider side, but go to two
different IPs (both on the same box) on my side.
How can I
Following up my own thread, I am kicking myself for quickly posting
without doing a bit of research. Apparently (no surprise) this
integration of Outlook and Asterisk is very old news, and there are many
projects out there. Anyone dealt with Thirdlane?
http://www.thirdlane.com/products/third
> The policy that we have been following is that only final releases will be
> announced to the asterisk-announce list. Betas and release candidates are not.
> The rationale is that asterisk-announce is supposed to be a low-volume list
> and
> that most subscribers to it would not appreciate all t
Noah Miller wrote:
The policy that we have been following is that only final releases will be
announced to the asterisk-announce list. Betas and release candidates are not.
The rationale is that asterisk-announce is supposed to be a low-volume list and
that most subscribers to it would not apprec
30 jan 2009 kl. 16.59 skrev Mike:
> hI,
>
> Trying to understand how to setup two PRIs in sip.conf. Using
> Asterisk 1.4.23.
>
> I have a provider giving me two PRI (different rate centers) through
> SIP. Both PRI comes in from the same IP on the provider side, but
> go to two different IP
this pdf
http://www.dialogic.com/products/docs/appnotes/10833_HMP_OpenSER_SIP_an.pdf
was enough to get me started with opensips
David fire wrote:
hi
i need a link or something about asterisk load balancing i cant find
any, i only found a paragraf in an email
anything wiil be wolcome
thank
Hi,
I'm very new to Asterisk. I tried VoiceMail() application. I'm able to
modify extensions.conf and voicemail.conf to send a voicemail audio file to
my email. It works great so far.
Now, I'm looking into publishing those voicemail files on a web page.
According to http://www.voip-info.org/wiki-
Hi !
I am trying to connect Asterisk with Avaya Definity.
I use this tutorial to do this
http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
The comunication between avaya and asterisk is fine but without sound. I can
call from Asterisk to Avaya and extension ring o
On 1/30/09, Mark Michelson wrote:
> Matt Florell wrote:
> > Yep, my bad I found them once I searched with the dash '-' after the
> > 1.4.23. They were lost in the flood of users list mail in my inbox.
> >
> > I wonder if these could also be posted on the asterisk-announce list
> > more consis
We use the default web interface, it comes with a file called vmail.cgi in
the defaiult Asterisk tree.
Thanks
l.
2009/1/30 Soonthorn Ativanichayaphong
> Hi,
>
> I'm very new to Asterisk. I tried VoiceMail() application. I'm able to
> modify extensions.conf and voicemail.conf to send a voicema
On Fri, 2009-01-30 at 17:17 +, Edwin Quijada wrote:
> Hi !
> I am trying to connect Asterisk with Avaya Definity.
> I use this tutorial to do this
> http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
> The comunication between avaya and asterisk is fine but witho
I recently completed PhoneClient 1.2 which is a Windows executable that
interfaces with Asterisk, with capacity to receive numbers from the
clipboard via a hotkey. PhoneClient is also a call manager and meeting room
interface. If interested please contact me off the list.
C. Savinovich
-O
Thanks
2009/1/30 Robin Rodriguez
> this pdf
> http://www.dialogic.com/products/docs/appnotes/10833_HMP_OpenSER_SIP_an.pdfwas
> enough to get me started with opensips
>
>
>
> David fire wrote:
>
> hi
> i need a link or something about asterisk load balancing i cant find any, i
> only found a pa
Hello list.
I'm having some problems with the CDR Radius in my Asterisk 1.4. I'm
using two TC400B cards for transcoding. When I reach nearly 100
simmultaneous calls, the CDR radius packets are being duplicated and I'm
getting this message in the asterisk console :
cdr_radius.c:227 radius_lo
There is a Digium solution to this as well...
Digium recently acquired snap-a-number and has put own their own version of
it already called "Asterisk Desktop Assistant"
It does exactly what you are looking for plus some more... Information is
still a little scarce since it is pretty new, but ther
Hi
I have this escenario:
|SIP or H323 phone|>|Cisco2600|E1-pri>|Asterisk|-->IVR,
A2Billing, etc...
The problem is that I can not hear any audio when call from 'sip or H323
phone' and configure something like: exten =>
_01XXX,1,Playback(thank-you-for-calling|noanswer) ...
It
Why off-list?
If you have an open-source application to share with us, why not share it
here?
If it's a commercial application, maybe it would be better to contact your
"prospect" directly.
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Origi
Yes it is shareable. Thanks for the interest :)... just that I feel I am a
few days away from the announcement
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Friday, January 30, 2009 3:0
> Recently I had the same problem using H323 with Cisco and I solved it
> by changing "bindaddr = 0.0.0.0" to the IP address of the Asterisk
> server.
>
You are my HERO!
This was the error!!!
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de
Thank you Lenz. That's exactly what I'm looking for.
On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri wrote:
> We use the default web interface, it comes with a file called vmail.cgi in
> the defaiult Asterisk tree.
>
> Thanks
>
> l.
>
>
> 2009/1/30 Soonthorn Ativanichayaphong
>
>> Hi,
>>
>> I'm
Replying to my own message. How difficult would it be to add a "bindaddr"
(and possibly bindport) PER PEER in SIP.conf?
How much of a bounty would I have to pay to get this done you think?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.co
I just tried the vmail.cgi app. Although working, there is clearly a
permissions problem preventing playing the wav files.
I run Fedora 8, and the patch files (on the wiki) are apparently broken.
Does anyone have a solution for fedora?
Thanks
From: asterisk-users-boun...@lists.digiu
Strangely, I can DELETE the messages from the web interface...just playback
causes a permission error...
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical
Support
Sent: January 30, 2009 9:40 PM
To: Asterisk Users List
Subje
It might be browser security issues? Have you tried with different browsers?
On Fri, Jan 30, 2009 at 9:53 PM, OCG Technical Support wrote:
> Strangely, I can DELETE the messages from the web interface...just
> playback causes a permission error...
>
>
>
> *From:* asterisk-users-boun...@lists.di
I'm interested in taking a persons spoken recorded name (First, Last) and
converting the two spoken words to text. Is there any solutions out there that
would make this possible?
___
-- Bandwidth and Colocation Provided by http://www.api-digi
There are solutions ranging from free to many thousands of dollars, with
effectiveness ranging from nearly worthless to almost pretty good.
A lot depends on your application.
The most successful application would match an utterance from a known
speaker to a known list of a couple dozen nam
I wouldn't have a database to compare names to, each one would essentially be
unique and unknown. It's sounding like this idea may be not possible...What
high end options are available? I read about lumenvox, but I believe that
compares to a known list of names (such as a directory, or Yes No,
No - the server generates the error:
Software error:
Hrm, can't seem to open
/var/spool/asterisk/voicemail/internalextensions/230/Old/msg.wav
For help, please send mail to th
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Hi Alfred,
There is a research project by Carnegie Mellon University (CMU) on a very
versatile Speech Recognition Software. Its Sphinx
http://cmusphinx.sourceforge.net/html/cmusphinx.php . This application is in
raw state and the Version 2 of sphinx could be integrated with Asterisk.
Festival (Tex
Thanks. This has been useful, I've been toying with Sphinx4 but haven't gotten
very far with it yet. Took a while to get my environment set up to compile all
the java sources.. should have downloaded the binary distro...:-)
From: Kurian Thayil
To: Asteris
Have you tried FreePBX ? It allows Asterisk administration via web interface
and has module just for that.
G.
On Fri, Jan 30, 2009 at 7:56 PM, OCG Technical Support wrote:
> No – the server generates the error:
>
>
>
> *Software error:*
>
> Hrm, can't seem to open
> /var/spool/asterisk/voicemai
Hi all,
I have a basic understanding about Asterisk and its Dialplan setup.
I have a sample dialplan scenario to setup as below:
1. When users registered to asterisk dial any number Asterisk receive the
call and wait for DTMF entry. (without answering the call)
2. Based on that DTMF Asterisk rou
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