On Sat, 2009-06-27 at 01:50 -0400, John A. Sullivan III wrote:
> Hello, all. I'm having a deeply frustrating time getting multiple
> parking lots to work and am wondering what I am doing wrong. I am using
> Asterisk 1.6.1.1. I defined two separate parking lots in features.conf
> as follows:
>
>
Hello, all. I'm having a deeply frustrating time getting multiple
parking lots to work and am wondering what I am doing wrong. I am using
Asterisk 1.6.1.1. I defined two separate parking lots in features.conf
as follows:
[parkinglot_a100] ; SSI
context => a100-parking
parkpos => 900-920
findslot
Hello, all. I'm having a nasty problem with call parking in Asterisk
1.6.1.1 that smells like a bug. When the call returns, it seems to be
returning to a "|" delimited extension and failing. Here is the output
from the console:
[Jun 26 22:20:42] NOTICE[7168]: chan_sip.c:18160 handle_request_inv
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Tzafrir Cohen a écrit :
| On Fri, Jun 26, 2009 at 07:50:08PM -0500, Moises Silva wrote:
|> On Fri, Jun 26, 2009 at 6:48 PM, Maris wrote:
|>
|>> I'm dealing with an idea to exchange data in a socket connection style
|>> or a sort of ftp transfer
>On Fri, Jun 26, 2009 at 11:10 AM, James Lamanna wrote:
> Hi,
> I remember seeing a T38 Gateway application for Asterisk 1.6 floating
> around, but I can't seem to find it again.
> Does anyone have any pointers to it? I really want to be able to send
> an incoming T38 connection directly to the PST
>On Fri, Jun 26, 2009 at 1:31 PM, James Lamanna wrote:
>> The use case is that a customer has a fax machine attached to an ATA.
>> The ATA sends T38 to Asterisk over SIP, then I need to forward that out
>> the PSTN.
> Got it. I'm saying why not skip the ATA and asterisk, and plug the fax
> into th
On Fri, Jun 26, 2009 at 8:39 PM, Moises Silva wrote:
[cut]
>
> I would think IAX ack just the signaling frames, not every single audio
> frame, does it?
That's correct. But I don't see why that can't be changed. Of course
the Audio doesn't have to be reliable
you can loose a few 20 ms frames and yo
I'm not sure where to check next, so I'm reaching out to those that
know this stuff better than I.
I've got Asterisk up and running, but I've still got an occasional
audio issue. Once in a while (maybe 1 out of every 20-30 calls), the
audio becomes heavily distorted, but only on the local si
On Fri, Jun 26, 2009 at 07:50:08PM -0500, Moises Silva wrote:
> On Fri, Jun 26, 2009 at 6:48 PM, Maris wrote:
>
> > I'm dealing with an idea to exchange data in a socket connection style
> > or a sort of ftp transfer with IAX2 as the transport medium.
> >
> > An IAX client on e.g. a notebook cou
On Fri, Jun 26, 2009 at 8:26 PM, Martin wrote:
> I'm sure he meant UDP not RTP.
>
> In order to guarantee the delivery you can simply do what IAX already
> does ... ACK the
> frames. This is what TCP does and ISDN PRI protocol layer 2 on the T1/E1.
>
> But why does he want to do it ? Share secret
I'm sure he meant UDP not RTP.
In order to guarantee the delivery you can simply do what IAX already
does ... ACK the
frames. This is what TCP does and ISDN PRI protocol layer 2 on the T1/E1.
But why does he want to do it ? Share secret / illegal files LOL ?
Martin
On Fri, Jun 26, 2009 at 7:50
On Fri, Jun 26, 2009 at 6:48 PM, Maris wrote:
> I'm dealing with an idea to exchange data in a socket connection style
> or a sort of ftp transfer with IAX2 as the transport medium.
>
> An IAX client on e.g. a notebook could establish a connection to any
> remote machine (also client) via any A
I'm dealing with an idea to exchange data in a socket connection style
or a sort of ftp transfer with IAX2 as the transport medium.
An IAX client on e.g. a notebook could establish a connection to any
remote machine (also client) via any Asterisk Server where both
clients are registered. Due to
Leah Newmark schrieb:
> I'm running asterisk 1.4.22 on a debian server.
> I have php5 installed and it works correctly command line.
> When trying to run a php script via AGI, I get messages such as:
> GI Tx >> I>
> AGI Rx << #!/usr/bin/php5 -q
> AGI Tx >> 510 Invalid or unknown command
>
> The sc
On Fri, 26 Jun 2009, David Backeberg wrote:
On Fri, Jun 26, 2009 at 12:03 PM, drew einhorn wrote:
Have a bunch of old clunker boxes on the scrap heap that would
probably do the job as a server. But I'm not confident
they would be reliable enough.
Capable of supporting maximum of a dozen 2-li
Leif Madsen schrieb:
> Julian Lyndon-Smith wrote:
>> I want to be able to implement hotdesking where an agent will logon to
>> any phone. I got all of that working, without having to reboot phones,
>> but then hit a brick wall.
>>
>> Voicemail.
>>
>> I still want each phone to use the BLF for
Norm Heinen schrieb:
> On Thu, Jun 25, 2009 at 11:08 AM, Leah Newmark wrote:
>> Take a look at this:
>> /var/lib/asterisk/agi-bin/olehphone# head incoming.php displays:
>> #!/usr/bin/php
>> >
>> Running it shows this:
>> /var/lib/asterisk/agi-bin/olehphone# ./incoming.php
>> #!/usr/bin/php5 -q
Tha
On Fri, 26 Jun 2009, drew einhorn wrote:
> To support small biz, family.
>
> Have a bunch of old clunker boxes on the scrap heap that would
> probably do the job as a server. But I'm not confident
> they would be reliable enough.
>
> Capable of supporting maximum of a dozen 2-line ATAs.
>
> Don't
Can anyone help me with this question? Thanks!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zaheer Master
Sent: Thursday, June 25, 2009 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
S
On Fri, Jun 26, 2009 at 11:10 AM, James Lamanna wrote:
> Hi,
> I remember seeing a T38 Gateway application for Asterisk 1.6 floating
> around, but I can't seem to find it again.
> Does anyone have any pointers to it? I really want to be able to send
> an incoming T38 connection directly to the PSTN
On Fri, Jun 26, 2009 at 4:21 PM, Kevin P. Fleming wrote:
> Alejandro Cabrera Obed wrote:
>
> > Because sounds files in /var/lib/asterisk/sounds are a lot as I see.
>
> If you are using the Spanish sounds distributed by Digium, they are
> already available in G.729 format from downloads.asterisk.or
On Fri, 2009-06-26 at 16:17 -0300, Alejandro Cabrera Obed wrote:
> Do IP phones and GSM gateway include valid G.729 licenses or do I have
> to pay for them ???
You shouldn't have to worry about them -- the G.729 licensing for those
devices is typically included in the cost of the DSP chips inside
Just a short question: I will have Asterisk using G.729 codec and connected
to some voip devices such IP phones (GarndStream) and a GSM gateway
(Portech).
Do IP phones and GSM gateway include valid G.729 licenses or do I have to
pay for them ???
Thanks a lot
Alejandro
___
Alejandro Cabrera Obed wrote:
> Because sounds files in /var/lib/asterisk/sounds are a lot as I see.
If you are using the Spanish sounds distributed by Digium, they are
already available in G.729 format from downloads.asterisk.org.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technol
Dear all, I have an Asterisk SIP PBX using GSM codec at peers and in
voicemail sounds files (I have Spanish sounds).
But now I have a problem because I have to use G.729 mandatory at peers, and
I have GSM in voicemail sound files. I can't let Asterisk do trascoding
because I have no a DSP in the C
Hi -
We've received a request for some contacts of people in the
University/higher education systems who have recently or quietly
implemented Asterisk. I have some on my press contact list, and some
that I just know personally, but I'm sure there is quite a list that
are less obvious a
I have been trying to get format_mp3 to work in 1.6.1.1 (addons 1.6.1.0) to
no avail; Asterisk seems to find the file and try to start playing back the
.slin conversion of the mp3 file, but fails. These files work correctly
with addons 1.6.0.0 against asterisk 1.6.0.9. Any suggestions as to where
On Jun 26, 2009, at 10:44 AM, Tim Nelson wrote:
> - "David Backeberg" wrote:
>> On Fri, Jun 26, 2009 at 1:31 PM, James Lamanna
>> wrote:
>>> The use case is that a customer has a fax machine attached to an
>> ATA.
>>> The ATA sends T38 to Asterisk over SIP, then I need to forward that
>> o
Yes, MOH requires a static configuration, but there is virtually no limit to
the number of classes you can define. You can also have single or multiple
files in each class. You could set up a database assigning a class to
extensions or groups of extensions and set the class before dialing based o
- "David Backeberg" wrote:
> On Fri, Jun 26, 2009 at 1:31 PM, James Lamanna
> wrote:
> > The use case is that a customer has a fax machine attached to an
> ATA.
> > The ATA sends T38 to Asterisk over SIP, then I need to forward that
> out
> > the PSTN.
>
> Got it. I'm saying why not skip the
On Fri, Jun 26, 2009 at 1:31 PM, James Lamanna wrote:
> The use case is that a customer has a fax machine attached to an ATA.
> The ATA sends T38 to Asterisk over SIP, then I need to forward that out
> the PSTN.
Got it. I'm saying why not skip the ATA and asterisk, and plug the fax
into the PSTN?
> On Fri, Jun 26, 2009 at 11:10 AM, James Lamanna wrote:
>> Hi,
>> I remember seeing a T38 Gateway application for Asterisk 1.6 floating
>> around, but I can't seem to find it again.
>> Does anyone have any pointers to it? I really want to be able to send
>> an incoming T38 connection directly to t
I am using version 1.6.0.x and you can do ³core show application dial² at
CLI to see info about the dial command.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: John Regal
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Fri, 26 Jun 2009 12
Thanks so much for this method. I am going to give it a shot. I am not
familiar with that "ghM" part. I tried looking for information on it - Is
that some undocumented macro call feature or something?
Thanks again.
John
_
From: asterisk-users-boun...@lists.digium.com
[mailto:aster
Optaining the name of the file to play dynamicaly is not a problem, the
limition is that musiconhold require a static configuration in a config file. I
was looking for a way to specify/select the file to play from the Dialplan.
From: asterisk-users-boun...@lists
On Fri, Jun 26, 2009 at 12:03 PM, drew einhorn wrote:
> Have a bunch of old clunker boxes on the scrap heap that would
> probably do the job as a server. But I'm not confident
> they would be reliable enough.
>
> Capable of supporting maximum of a dozen 2-line ATAs.
Spend the money on a 1-port di
On Fri, Jun 26, 2009 at 11:10 AM, James Lamanna wrote:
> Hi,
> I remember seeing a T38 Gateway application for Asterisk 1.6 floating
> around, but I can't seem to find it again.
> Does anyone have any pointers to it? I really want to be able to send
> an incoming T38 connection directly to the PSTN
ok, but the classes in "musiconhold.conf" are static and require a reload to be
modified.
From: asterisk-users-boun...@lists.digium.com on behalf of Philipp Kempgen
Sent: Thu 04/06/2009 7:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Thu, Jun 25, 2009 at 9:55 PM, John Regal wrote:
> When using this method, it appears that the call file creates the first part
> of the call, then creates a second call with the Dial() app. Once the call
> executed by the Dial() app is answered, the two calls are joined together.
> What I am exp
To support small biz, family.
Have a bunch of old clunker boxes on the scrap heap that would
probably do the job as a server. But I'm not confident
they would be reliable enough.
Capable of supporting maximum of a dozen 2-line ATAs.
Don't want to spend more than a couple hundred dollars
on a ne
asterisk*CLI> sip show domains
Our local SIP domains: Context Set
by
jocan.local (default)
[Configured]
192.168.1. (default)
[Configured]
[Jun 26 17:49:03] NOTICE[5570]: chan_sip.c:15889
han
On Wed, Jun 24, 2009 at 11:11 AM, David Backeberg wrote:
> On Wed, Jun 17, 2009 at 7:10 PM, Marshall
> Henderson wrote:
>> architecture, etc. On a brand new dual or quad core xeon type
>> system(quite likely multiple physical CPUs, each with multiple cores),
>> And finally, are there any hard or so
I issue this command:
RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ
ueue^SIP/GXP280_18))
Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds.
Asterisk rings again for 10 seconds. I would expect this to happen a total
of 4 times.
The problem is th
Hi,
I remember seeing a T38 Gateway application for Asterisk 1.6 floating
around, but I can't seem to find it again.
Does anyone have any pointers to it? I really want to be able to send
an incoming T38 connection directly to the PSTN.
Thanks.
-- James
___
Read my reply, it is a better way.
On Fri, Jun 26, 2009 at 9:46 AM, Arjan Kroon | Mobillion <
arjan.kr...@mobillion.nl> wrote:
>
> Hi,
>
> Your correct, this the best way.
> But we don't have any 'balancing' on the localhost.
> In some cases we have to connect directly to a central database. (we
There's an rxgain feature in voicemail.conf you can use to amplify or muffle
the recording at call time. Your best bet would be to make the default
format be wav49 instead of wav, then pass the file through sox to remove
white noise and level out the volume. Check out voip-info.org
-Origina
I use the following script to perform compression and normalization on
e-mailed voicemails. I put the script in as /usr/local/bin/sox and
pre-pend /usr/local/bin to the PATH before asterisk runs in the
startup script.
The values for the compressor are not scientific, I monkeyed with them
We generally get our voicemails emailed to us from asterisk, but some
people's messages are extraordinarily loud or quiet. I don't suppose
there is any feature to even out the volume level is there?
___
-- Bandwidth and Colocation Provided by http://
Hi Users!
I'm having problem with the Background command. It is not interrupted as
desired when the caller hit the keys. Or actually it is interrupted
randomly, sometimes and sometimes not.
We're using zap channels with an PRI ISDN.
Has anyone come across this problem before?
Regards,
Hanna
Hi,
I met Matt Florell at AMOOCON and tried to record an interview. I was
pleased with the results, but later found that the battery deleted the
audio file when it went dead. Today, we'll have Matt live to talk
about VICIDIAL and answer any questions you may have about it.
For more on this: http:
Hi,
Your correct, this the best way.
But we don't have any 'balancing' on the localhost.
In some cases we have to connect directly to a central database. (we
have only one central database)
If the machine where the central database is running on, is down, than
FastAgi will try to connect to this
Hi all, does anyone know of an application that will run in Windows (in my
case users PC's) and behave in a similar fasion to chan_mobile? I'd like the
app to register with asterisk, then talk to a (or a number of) mobiles over
bluetooth thus creating an FXO port? I'm not interested in SMS etc. jus
I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for
my problem
Hello,
During a call with canreinvite = no, at the beginning of the call I lose
2 seconds of audio.
is obvious when I call autoattendant.
schema:
SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1
Greetings,
I'm sorry I've been taking so long to reply, but I've been swamped and
didn't have the time to try to compile it.
First of all, thank you all for the help.
Kyle Kienapfel wrote:
>
> why is CROSS_ARCH=Linux? is this something the AVR32 distro is doing,
> or something you did? it shou
Hi,
Your correct, this the best way.
But we don't have any 'balancing' on the localhost.
In some cases we have to connect directly to a central database. (we
have only one central database)
If the machine where the central database is running on, is down, than
FastAgi will try to connect to this
Simpler solution.
What version of Asterisk. 1.2.x you had to enable (patch for) N+101, seems
later versions are more graceful.
http://www.voip-info.org/wiki/view/Asterisk+FastAGI
Error Handling+Asterisk 1.4 & 1.6.x As of Asterisk 1.4 a new channel
variable, AGISTATUS, is set to SUCCESS upon succ
On 10:42, Fri 26 Jun 09, Arjan Kroon | Mobillion wrote:
> Hi,
>
>
>
> How do you all handle the situation when a centrale fastagi server
> process(es) are down? AGI(..) prints "Unable to locate host" and the
> dailplan jumps to extension h.
>
> I'd like to handle the return value and keeping
Hi,
How do you all handle the situation when a centrale fastagi server
process(es) are down? AGI(..) prints "Unable to locate host" and the
dailplan jumps to extension h.
I'd like to handle the return value and keeping the caller in the
dailplan and not to the hangup extension.
Any tips abou
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