On Thursday 03 June 2010 21:28:39 Richard Kenner wrote:
> I did a usual "svn update", "./configure" and "make" and got
>
>[CC] chan_oss.c -> chan_oss.o
> gcc: @SDL_INCLUDE@: No such file or directory
>
> I don't see any changes to chan_oss recently, so don't understand this.
> What could be goi
I did a usual "svn update", "./configure" and "make" and got
[CC] chan_oss.c -> chan_oss.o
gcc: @SDL_INCLUDE@: No such file or directory
I don't see any changes to chan_oss recently, so don't understand this.
What could be going on?
--
On 06/04/2010 02:27 AM, Kyle Kienapfel wrote:
> http://en.wikipedia.org/wiki/G.729
> Looks like theres A and B and no "A/B" so theres nothing to worry about
>
What's the point of quoting a page, if you are not actually going to
read it?
> On Thu, Jun 3, 2010 at 9:09 AM, Alejandro Cabrera Obed
Hi,
I work for a small VoIP provider in Southern California. We are looking
for someone very knowledgeable in Asterisk to help work on the following:
- Maintenance of current Asterisk servers, updating Asterisk, monitoring
load, and other sysadmin tasks
- Devise and implement high-availability st
On Thu, Jun 3, 2010 at 5:02 PM, Warren Selby wrote:
> The resolution [1] to this issue was to uninstall and reinstall [2] the
> kernel headers on the machine...just in case anyone else runs into this
> issue and would like to know how it was solved.
> [1] https://issues.asterisk.org/view.php?id=17
The resolution [1] to this issue was to uninstall and reinstall [2]
the kernel headers on the machine...just in case anyone else runs into
this issue and would like to know how it was solved.
[1] https://issues.asterisk.org/view.php?id=17411
[2] run these commands to reinstall kernel headers
I filed the following bug on the 28th of May.
0017371: [patch] [regression] DAHDI analog FXS port segfaults after dialling
2nd DTMF digit
Please see https://issues.asterisk.org/view.php?id=17371
You problem sounds the same, if it is the same please report this on the
bug.
Alec Davis
-Origin
Just curious,
Any chance of using amr for asterisk?
http://en.wikipedia.org/wiki/Adaptive_Multi-Rate_audio_codec
The codecs (both wb and nb) seems to be available at packman:
http://ftp5.gwdg.de/pub/linux/misc/packman/suse/11.2/src/amrnb-7.0.0.2-0.pm.5.1.src.rpm
http://ftp5.gwdg.de/pub/linux/mis
If you do AGI and DEADAGI in the h extension, all calls will be handled but
you'll get a "goober" warning from whichever one is not applicable. I'm
guessing that ideally you would try and to the AGI in the live context where
applicable. See what you can do with this example -
http://www.voip-inf
Hi. For several months now asterisk will mysteriously stop inserting
records into cdr database. I am using mysql and the asterisk addons
1.6.2 to accomplish this. Sometimes there is a strange error about
column names, but often there is no error, it just stops. I just have
to restart asterisk t
DeadAGI is executed if call is successful. I wanna ask how to execute agi
script if the call is not only successful but also reject, busy, etc...
2010/5/5 Danny Nicholas
> Regular AGI with SIGHUP detection?
>
>
>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI
>
>
> ---
Hi. I have been using asterisk-1.6.2 and if I update the version --
using svn -- to around May 19 or after, when I dial a digit on my fxs
port which is on an X400p card, asterisk seg faults. If I go back
before about this date, this problem does not occur. The dahdi version
is svn 7445.
Any id
Hi,
I'm trying to get the match_auth_username=yes sip configuration working.
It's mentioned as an experimental new feature of 1.6.2.x. (I'm using 1.6.2.8)
The sip.conf example states:
; if available, match user entry using the
; 'username' field from the authentication line
; instead of the From:
On Thu, 3 Jun 2010, Steve Edwards wrote:
>> -Also, what about slow queries? If a query takes a few seconds to
>> complete, does the call wait for the query to complete or are there
>> timeouts for the query that could result in dropped calls?
>
> (I prefer to call MySQL from an AGI instead of th
http://en.wikipedia.org/wiki/G.729
Looks like theres A and B and no "A/B" so theres nothing to worry about
On Thu, Jun 3, 2010 at 9:09 AM, Alejandro Cabrera Obed
wrote:
> Dear all, I've read that Asterisk supports only the G.729 A audio
> codec. I have several Grandstream IP phones with G.729 A/B
On Thu, 3 Jun 2010, Zeeshan Zakaria wrote:
> Can somebody please confirm that Wait or Playback commands can't be used
> in h extension. This is for asterisk 1.4. Is there a way to delay the
> hangup by a new seconds once the call is over? Using 'g' option in the
> Dial command is not an option
Un-top-posting...
On Thu, 3 Jun 2010, Zeeshan Zakaria wrote:
> Its your personal opinion. Actually as a non-native-English speaker to
> me "Noop" sounds much better than "Verbose" which itself is a confusing
> word, plus I guess command "verbose" is new in 1.6, and I've never used
> it, so I'l
Can somebody please confirm that Wait or Playback commands can't be used in
h extension. This is for asterisk 1.4. Is there a way to delay the hangup by
a new seconds once the call is over? Using 'g' option in the Dial command is
not an option in my case. Is there a list of commands which can and w
Its your personal opinion. Actually as a non-native-English speaker to me
"Noop" sounds much better than "Verbose" which itself is a confusing word,
plus I guess command "verbose" is new in 1.6, and I've never used it, so
I'll stick with "Noop" which can be used with any version of asterisk.
Zeesh
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wed, Jun 02, 2010 at 21:50:43 -0300, Daniel Bareiro wrote:
>>> Another thing I want to try is to connect Asterisk with Siemens PBX
>>> so that the extensions on Asterisk can communicate with the
>>> extensions on the Siemens PBX and vice versa. For
Hi all,
do you know any firmware release which fixes that issue for cisco ATA186?
ATA 186 3.x.x Cisco ATA 186 v3.* CANCEL requests can be sent
with a completely bogus URI, making it impossible to cancel a call. bug
no workarounds
This is from: http://www.communigate.com/SIP/S
> - "Doug Lytle" wrote:
>> I use the mysql add-on, I'd create a subroutine that gets called at
>> dial
On Thu, 3 Jun 2010, Tim Nelson wrote:
> Thank you for the example. I'm using something very similar to this now,
> looking up the destination ${EXTEN}, and using GotoIf if the number of
Un-top-posting...
> On 2010-06-03 9:35 AM, "Necati Demir" wrote:
> I want to ask how to get call duration.
On Thu, 3 Jun 2010, Zeeshan Zakaria wrote:
> exten => h,1,Noop( >>> Call duration was ${CDR(duration)} seconds)
It would be a "better practice" to use the application provided expl
Dear all, I've read that Asterisk supports only the G.729 A audio
codec. I have several Grandstream IP phones with G.729 A/B codec
implementation.
Does G.729 A/B mean both version A and version B, or A/B is a new
version different from A and B and it's not supported by Asterisk ???
Thanks a lot
On Wed, 2010-06-02 at 15:35 -0600, Greg Woods wrote:
> Is there a reasonably easy way to increase the volume on a DAHDI
> channel?
Thanks to everyone for the pointer to rxgain/txgain in chan_dahdi.conf .
That seems to have done the trick.
--Greg
--
___
Tim Nelson wrote:
>
> I do have a few additional questions however:
>
> -How reliable is the MYSQL application?
>
I've been running it for years now with no failures.
> -In the event of a failed database query (load too high), what happens to the
> call?
>
The channel will get stuck in
- "Doug Lytle" wrote:
> I use the mysql add-on, I'd create a subroutine that gets called at
> dial
> time. As an example, I set outbound caller-id with the below
> subroutine. You would use ${EXTEN} for the lookup and decide the
> route
> to take on the results.
>
> [set_callerid]
>
> e
> Seems to me a similar argument for and against hosting ones own web
> presence in house with mixed results . Others choose to use a
> datacenter service, seldom but sometimes with poor results.
I think that's a good analogy. It's very hard to argue that one of those
choices is "right" and th
On Thu, May 27, 2010 at 6:17 PM, Theo Band wrote:
> I used to build Asterisk from source including the zaptel-dummy module.
> Last year I decided to upgrade and use a yum repository. I hoped that
> this would be less hassle compared to manually chasing after the latest
> release, compiling etc. An
This is a typical scenario with so many companies who fail to recognize that
they need to take asterisk deployment seriously. I have seen so many of
these companies since 2004, since when I am in this industry. Many of them
don't want to spend money on right hardware, bandwidth and or the right
con
First thing which comes to mind is:
exten => h,1,Noop( >>> Call duration was ${CDR(duration)} seconds)
exten => h,n,Hangup()
There is also a variable ${CDR(billsec)} which shows only the duration the
call was actually connected between two channels, however this may not match
with the duration of
Do you really expect an unbiased response from this community?
Seems to me a similar argument for and against hosting ones own web
presence in house with mixed results . Others choose to use a
datacenter service, seldom but sometimes with poor results.
Placing ones business lifeline in the ha
On 3 Jun 2010, at 14:24, Necati Demir wrote:
> I want to ask how to get call duration.
Go on then
When you do ask the question you might want to include a few details. Are you
trying to get call duration during a call? If so then the cli will help 'core
show channels'. If it's after the cal
Silly. My guess is that someone that doesn't know anything about phones
decided to install it and failed. Lots of erroneous statements:
" Asterisk because it required a custom-built server" - Nope. You can
pretty much use any old server or really even a desktop machine for an
install this small
On Thu, Jun 03, 2010 at 08:24:11AM -0500, Danny Nicholas wrote:
> Txgain/rxgain in dahdi.conf control this - you will have to restart asterisk
> on each change to test the values to set to your liking
reload dahdi, you mean:
module reload chan_dahdi.so
Or simply 'reload'
--
Tz
Gilles wrote:
> Hello
>
> I just read this article and would like some feedback from
> experienced Asterisk users:
>
> ===
> "Failed open source VoIP deployment leads to hosted VoIP strategy" By
> Jessica Scarpati
>
> "When budgets are crimped, open source voice over IP (VoIP)
Txgain/rxgain in dahdi.conf control this - you will have to restart asterisk
on each change to test the values to set to your liking - my settings are
rxgain=8.0
txgain=4.0
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Hello,
I want to ask how to get call duration.
--
Necati DEMİR
http://demir.web.tr
http://friendfeed.com/ndemir
ndemir ~ demir.web.tr
---
--
_
-- Bandwidth and Colocation Provided by http://w
Hello
I just read this article and would like some feedback from
experienced Asterisk users:
===
"Failed open source VoIP deployment leads to hosted VoIP strategy" By
Jessica Scarpati
"When budgets are crimped, open source voice over IP (VoIP) solutions
look attractive -- a l
I am using DeadAGI script and using this context.
exten => 10,1,Dial(SIP/${EXTEN})
exten => 10,n,Wait(1)
exten => 10,n,Playback(${PLAYFILE})
exten => 10,n,Wait(1)
exten => 10,n,Hangup()
exten => h,1,DeadAGI(script.agi)
DeadAGI script executes only if the call is successful. How to run DeadAGI
sc
Just googled it:
http://www.iptel.org/service
Echo test call
Call echo (sip:e...@iptel.org) or the vanity number 3246 for an echo
test call. You can change the buffering while in the call by pressing
the star key.
Music test call
Call music (sip:mu...@iptel.org) to listen to a wonderful fado of
A
Hi, in trixbox I don't know what create an extension with letter but only
with number.
Thanks.
--
Salvatore.
- Original Message -
From: "Kingsley Tart"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, June 02, 2010 5:34 PM
Subject: Re: [asteris
Hi, I have tried with a some change in IVR configuration but the result
isn't changed, I have tried with "Enable Directory" and "Enable Direct Dial"
disabled, also I have tried with timeout=1 but nothing is changed !
My IVR configuration is:
trixbox1*CLI> dialplan show ivr-2
[ Context 'ivr-2' c
Hello Kyle!
Earlier I was listening to my voicemail using my ISDN-card and a simple
telephone. But this card is no longer supported. So I just go to:
/var/spool/asterisk/voicemail/default/1234/INBOX
Then of course I use JACK. I seems some doesn't work as expected. I have the
feeling, the pe
44 matches
Mail list logo
