Hello Asterisk users,
We are having MANY but MANY problems configuring an analog fax machine to
work properly on Asterisk, the first thing we do was to plug in the Fax
analog machine to the FXS port of the Digium TDM410P, we set echocancel=no
on zapata.conf and also faxdetect=yes on general sectio
Hi,
I'm tying to delay the HungUp.
I tried this way:
exten => h,1,NoOp(Start)
exten => h,n,Wait(5)
exten => h,n,NoOp(End)
exten => h,n,Hangup()
but it doesn't work, Any idea?
Thanks in advance.
--
_
-- Bandwidth and Colocat
- "Steve Davies" wrote:
> I'll have to give that a go. Is there something similar available for
> all of the other Channel technologies, or at least for DAHDI and IAX?
This works for SIP and IAX from at least the 1.4 release, and in DAHDI since
1.6.0 release.
> Is TRANSFER_CONTEXT copied t
Hi all,
Thanks to Russ Meyerriecks for his previous reply in this thread,
which was very informative. I'm now hoping that someone will comment
on the following:
"ISDN uses LAPD for the D-channel and LAPB for data connections over
the B-channels. However, LAPB is irrelevant for Asterisk, bec
- Original Message -
From: "Kevin P. Fleming"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 15, 2010 4:45 PM
Subject: Re: [asterisk-users] Asterisk/Polycom Dialed Party Name
> Karl Fife wrote:
>
>> Question about how Called Party ID feature is im
I have defined a moh class in musiconhold.conf and set the class using
SetMusicOnHold in my dialplan:
; musiconhold.conf
[music1]
mode=files
directory=moh/music1
sort=alpha
; extensions.conf
exten => s,n,SetMusicOnHold(music1)
exten => s,n,Queue(${ARG1})
Inside /var/lib/asterisk/moh/music1 there
My problem really is find out how Asterisk::fastagi works.
> Date: Thu, 15 Apr 2010 13:05:03 -0800
> From: s...@inbox.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] FastAGiin Windows Server
>
> You could always ask someone to rewrite the perl code to something else.
Karl Fife wrote:
> Question about how Called Party ID feature is implemented in trunk/1.8:
> Will an instance be able to populate the Called Party ID with any dialplan
> variable, (think CNAM dip to an external DB) or will the Called Party ID be
> drawn from an asterisk-resident data structure s
- Original Message -
From: "Jared Smith"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 15, 2010 8:22 AM
Subject: Re: [asterisk-users] Asterisk/Polycom Dialed Party Name
>...Called Party ID will
> be supported in Asterisk 1.8.
Question about how C
You could always ask someone to rewrite the perl code to something else.
> -Original Message-
> From: listas_quij...@hotmail.com
> Sent: Thu, 15 Apr 2010 20:52:45 +
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] FastAGiin Windows Server
>
>
>
>
>
>
>
>> Da
Steve, Chris :
I too had this problem and the solution was not tweaking
the AMD parameters, but playing a short audio file (even
a really really short one) before executing the AMD function.
The key is executing the "Background" step before AMD()
Please see sample dialplan below :
ext
> Date: Wed, 14 Apr 2010 21:09:03 -0400
> From: dbackeb...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] FastAGiin Windows Server
>
> On Wed, Apr 14, 2010 at 8:55 PM, Edwin Quijada
> wrote:
> >
> > My problem is that I need to execute windows app using IVR
The Asterisk Development Team has announced releases of Asterisk-Addons version
1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1. These releases are available for
immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases resolve several issues reported by the community:
* Fi
Hi,
I am migrating some billing code from 1.2 to 1.6 cdr output. Mostly
this is not too bad, but I have a scenario where some data appears to
be "lost"
Call from SIP/100 to SIP/200, but the SIP/200 device is programmed to
send a redirect to extension 1234. chan_sip creates a
Local/1...@context ca
"Please note: A Zaptel timer must be present for conferencing to work!", but
if the user does not have "ZAP/DAHDI hardware", he can use ZAP/DAHDI DUMMY
2010/4/15 Luki
> > I use the option 'r' on 1.4, to record the meetme application.
> > Asterisk leaves these records at /var/lib/asterisk/sounds
Hi!
> If anyone has a better solution, please tell us :).
Simply add an articifical user to the conference that does the recording,
and let it exit the room automatically when everyone else (or the last
person) has left.
Philipp
--
___
> I use the option 'r' on 1.4, to record the meetme application.
> Asterisk leaves these records at /var/lib/asterisk/sounds/meetmeXX.
That option only works for conferences using ZAP/DAHDI hardware.
You can, however, start to Monitor() the channel prior to entering the
conference, but you s
Thanks!
That is some good info, I will check with them.
On Thu, Apr 15, 2010 at 1:49 AM, Vahan Yerkanian wrote:
> On 4/15/10 1:26 AM, Tonty T wrote:
>
> That's is all the overhead I am trying to avoid. What I need is a DID with
> unlimited channel, but they do not offer DIDs in that country.
On Thursday 15 April 2010 09:11:09 Richard Kenner wrote:
> Is there an explanation other than the one in the application documentation
> of exactly what this is for and when you'd want to use it and when you
> wouldn't? I find the explanation in the documentation a little confusing.
The way Caller
Hi List,
Ive got a bunch of Avaya IP9640 that we want to convert to SIP and then hook
up to Asterisk so we can dump this overpriced Avaya system. Ive got ahold of
the SIP firmware, but I cannot find anything on how to convert the phone
itself to SIP, when I go into setup mode it wants a "command"
Sangoma uses wanpipe. Channel drivers set upon wanpipe (very
simplifiedly speaking). You can configure chan_ss7 to use with sangoma
as decribed at http://www.voip-info.org/wiki/view/Sangoma+A102+SS7+setup .
huu giang schrieb:
> Hi,
>
> I have a question about point code. When I install a sangom
On 15 April 2010 14:11, Jared Smith wrote:
> On Thu, 2010-04-15 at 13:59 +0100, Steve Davies wrote:
>> Let us then assume that the contexts are configured in the config files as:
>> IAX/1234: context=external
>> SIP/100: context=default
>> SIP/101: context=superuser
>> SIP/102: con
Is there an explanation other than the one in the application documentation of
exactly what this is for and when you'd want to use it and when you wouldn't?
I find the explanation in the documentation a little confusing.
--
_
--
Hi Hoanh Anh,
Thanks for your answer. You are very kind.
With Asterisk, I'm just a newbie, so I really need your help. I'll very happy
if I can have a meeting with you next month in Vietnam.
Thanks
--- On Thu, 4/15/10, Ngo-Vi Hoai-Anh wrote:
From: Ngo-Vi Hoai-Anh
Subject: Re: [asterisk-us
Hi,
I have a question about point code. When I install a sangoma card, there
weren't any step require me to configure point code.
I thin when telco provide us E1 lines, MSC will configure to forward calles
through the E1 lines to my Asterisk ?.
Please point me how can I configure point code f
On Thu, 2010-04-15 at 09:09 -0400, Marc Smith wrote:
> On the Avaya's, when you dialed another user's internal extension, on
> the phone you are dialing from, it would display the user's name that
> you're dialing.
It's not supported in your version of Asterisk, but Called Party ID will
be support
On Thu, 2010-04-15 at 13:59 +0100, Steve Davies wrote:
> Let us then assume that the contexts are configured in the config files as:
> IAX/1234: context=external
> SIP/100: context=default
> SIP/101: context=superuser
> SIP/102: context=local
>
It's early and my brain hasn't fully
Hi,
We are in the process of moving from an Avaya Definity to Asterisk for
our institution's phone system. I got one feature that the Avaya had,
which I have not been able to reproduce with Polycom phones and
Asterisk; since this feature seemed so small and useless to me when
testing, I kind of ig
Hi,
Can anyone suggest a way of doing the following in Asterisk 1.6.2 - I
do not think it can be done trivially using TRANSFER_CONTEXT.
What I want is for the TRANSFER_CONTEXT for all technologies to be the
same as the initial context defined in the configuration of the device
initiating the tran
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Hi
I'm using Asterisk 1.6.0.22, and have a bit non-standard setup. I have a
queue with three static agents, and the ringall strategy.
While call A is answered by agent 1, two more calls, B and C enter the
queue. Agents 2 and 3 does nothing and when
I use the option 'r' on 1.4, to record the meetme application. Asterisk
leaves these records at /var/lib/asterisk/sounds/meetmeXX.
take a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe if you
need more information.
Regards.
2010/4/14 Renato bianchini
> Hello,
>
> I wanna r
Hello,
I'm trying to setup shared lines with Asterisk 1.4.26 and Snom phones.
It seems that Asterisk works correctly (I get "State:
SLA_TRUNK_STATE_RINGING" from the CLI) but the lamps on the phone are
not blinking even if I setup one function key on my phones as shared
line with number: whe
Hi, when i use Set(CDR(amaflags)=1) in my dial plan , after a call. i
look at the cdr table in mysql database.
the value of amaflags is -1 not 1, do you know what's wrong?
Thanks!
--
Best regards,
Sucan
--
_
-- Bandwidth and C
Hi there,
ISDN is not a protocol. See
http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network#Internationale_Verbreitung
for more details. As far as I know Vietnam uses DSS1 (Euro-ISDN).
The benefit of SS7 is that you can use a single signaling channel
(D-Chan) for more E1s. I have 8
hi huu,
ss7 is preffered for me.
tested on huawei msc / siemens ewsd / alcatel s12 just to mention few and
its working fine.
On Thu, Apr 15, 2010 at 8:04 AM, huu giang wrote:
> Dear Goe.
>
> Do you mean, I just need request Telco provide me a E1 line, ask them to
> configure MSC support SS7/IS
Hi
Can someone help me in configuring say.conf file for Indian Languages?
I want to play numbers and dates in regional languages. I need if for
Bengali, Kannada, Telugu, Hindi, Marathi, Malayalam, Tamil, Gujrathi.
Thanks & Regards,
Amit Patkar
--
Dear Goe.
Do you mean, I just need request Telco provide me a E1 line, ask them to
configure MSC support SS7/ISUP, so my Asterisk can receive calls.
What is the benefits if I use ISDN instead of ISUP/SS7 and vice versa.
Thanks.
--- On Wed, 4/14/10, Goke M Aruna wrote:
From: Goke M Aruna
Su
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