Hi List,
http://asterisk-espeak.sourceforge.net/
this link tell how to install espeak module but nothing happened when we run
the command always dependency error.
Is there any alternate way to install it with Hindi and English voice
version.
On Tue, Aug 23, 2011 at 4:18 PM, Olivier
hi:
please check the redfone solution.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: aster...@a-domani.nl
To: asterisk-users@lists.digium.com
Date: Thu, 1 Sep 2011 23:48:46 +0200
Subject: Re:
Hi
I'm using asterisk 1.6.2.18.1
I'm having a problem where only the first four digits are collected in the
cdr when the call is dialed overlap but if the call is dialed en-block the
whole dialed digits are recorded
chan_dahdi.conf
[trunkgroups]
[channels]
language=uk
switchtype=euroisdn
Hello List,
I have seen that when ever asterisk gets a SIP INFO request from a SIP channel
it generates the requested DTMF tone and writes to the destination channel also
it forwards the SIP INFO message. As I am very new to this domain, it is really
confusing me. Why not asterisk writes only
Thanks for the reply!
i'll try to install asterisk on the same machine
--
From: Lee Howard fax...@howardsilvan.com
Sent: Thursday, September 01, 2011 6:29 PM
To: ales...@asistar.it
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Thx @James
(1) We do not use any analog / digital phone lines. SIP based DIDs and
Softphones.
Do I still need timing source ?
(2) What does timing source do, how does ithelp ?
Any insights will help.
Thx Rgds,
Sanjay
2011/9/2 James zhu zhulizh...@live.com
hi:
please check the
Hello,
Out of curiosity, has Asterisk been successfully compiled and ran
Asterisk on an Android smartphone?
I could use a small IVR on my smartphone to handle incoming calls.
Thank you.
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-- Bandwidth and Colocation
Hi list,
I want ot do basic work (add-edit-delete) into asterisk configuration files,
like sip.conf, manager.conf,musiconhold.conf etc.
Please guide me how to configure all these files from from AMI connection. I
am able to login into AMI from Login action but I want to do more task in to
it.
On Friday 02 September 2011, Gilles wrote:
Out of curiosity, has Asterisk been successfully compiled and ran
Asterisk on an Android smartphone?
I could use a small IVR on my smartphone to handle incoming calls.
Thank you.
TTBOMK it's been done; but without the necessary Zaptel / DAHDI
If I install asterisk i have the same problem.
can anyone help me?
thanks
--
From: Lee Howard fax...@howardsilvan.com
Sent: Thursday, September 01, 2011 6:29 PM
To: ales...@asistar.it
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
P.H.B. is insisting on having the ability to create a transparant SIP
tunnel between old style ISDN telephony PBX with overlap dialing:
PBX - ISDN - IAD - SIP - * - DAHDI - PRI
The idea is that dialed numbers a the PBX are transmitted to the PRI as
they are typed, whenever the PRI gets the
On Fri, 2 Sep 2011 13:23:18 +0100, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
TTBOMK it's been done; but without the necessary Zaptel / DAHDI drivers to
interface with the phone line, it's rather less useful than it sounds.
I'm looking for a way to an IVR in my smartphone to handle incoming
The error happens so quickly that I would suspect that it has to do with
fax detection within Asterisk re-routing the call to a different place.
Watch the CLI when a fax call comes in and see what happens there.
Alessio wrote:
If I install asterisk i have the same problem.
can anyone help
Do you want to run the entire PBX on the Android client or are you just
looking for a IAX programm to be installed for receiving calls?!
I think this is what you ment.
Here is the url:
https://market.android.com/details?id=com.bw.iax.ui
Am 02.09.2011 16:32, schrieb Gilles:
On Fri, 2 Sep 2011
1: from the phone i called the fax-server
2: from external fax i tried to send a fax to fax-server
the results:
1: from the phone ( I hear sound of fax )
__
Hello All,
We would like to change our dialplan a bit so that after a user dials a number
(any number, including domestic, international, internal) Asterisk firsts
prompts the user for a PIN before actually allowing the call to go through.
I know I could setup an IVR that would accomplish
Have you looked at pin sets in freepbx / trixbox / elastix? I haven't tested
it myself - but I know the feature is present there
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps
Sent: Friday,
Hi!
I recently upgraded Asterisk from version 1.6.2 to 1.8.5
Now about every 10 minutes all SIP TRUNKS becomes UNRECHABLE for a few seconds
or minutes after become LAGGED and later become OK.
I have no idea of the cause of this problem.
With the version 1.6.2 all runs perfectly.
I can't say
On Fri, Sep 02, 2011 at 11:14:57AM -0400, Brandon Phelps wrote:
We would like to change our dialplan a bit so that after a user dials a
number (any number, including domestic, international, internal) Asterisk
firsts prompts the user for a PIN before actually allowing the call to go
On Sat, Aug 27, 2011 at 01:11:00PM +0100, David Woodhouse wrote:
http://www.voipsupply.com/siemens-gigaset-a580-ip
I bought a Siemens Gigaset C475IP at the beginning of this year.
Strange, I haven't been able to buy this phone for a about 2 years. It
was replaced with the A580.
RFC2833
On 09/01/2011 04:39 PM, Hans Witvliet wrote:
From the asterisk-bible and the wiki's i learned that it is possible to
let asterisk do some of the presense-info by means of the jabber.conf
file and a seperate xmpp-server.
You are misunderstanding a bit; Asterisk can use an XMPP server and
Google for DISA.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps
Sent: Friday, September 02, 2011 10:15 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Prompt for PIN After
Why php? Isn't vi the only way?
On Fri, Sep 2, 2011 at 7:28 AM, virendra bhati virbh...@gmail.com wrote:
Hi list,
I want ot do basic work (add-edit-delete) into asterisk configuration files,
like sip.conf, manager.conf,musiconhold.conf etc.
Please guide me how to configure all these files
since www.ilbcfreeware.org is broken, asterisk installs that want ilbc
are failing.
I have no idea how to contact them since the site is offline. It's been
offline at least 12 hours - I can't imagine they *don't* know but at the
same time it's still offline..
pbx1 dig +norecurse
On Thu, 1 Sep 2011, RSCL Mumbai wrote:
I tried and failed with VirtualBox too. Timing seemed impossible to
maintain, even on beefy hardware (hexacore)
with plenty of RAM (16G), and nothing else going on (single instance). I
don't think VirtualBox is up to real-time
stuff.
We use LXC
On 2/09/2011 12:13 AM, Alessio wrote:
Hi!
from 2 days I'm trying to run hylafax server and iaxmodem with
Asterisk 1.8.5.
I have 2 computers in the lan, one is the Asterisk PBX and the other
is the server with hylafax and iaxmodem installed.
In Asterisk I set up an IAX trunk in this way:
I
What sort of things should I watch out for when upgrading from 1.4.39 to 1.8.5
Thanks,
--
Joseph
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
This showed up in my log today and I have no idea what it means. I'm
running 10.0.0-beta1 though I've had issues like this on and off for a while.
No idea if it's a problem with my machine, network or if it's a
problem with Asterisk. It does seem like when the errors show up we
that have
read the 1.6 README and the 1.8 README.
If you're using SIP you should expect changes with account
authentication, faxing, output regarding channel status and
performance.
I think that version of 1.4 is late enough you would already be on
DAHDI for hardware devices. If not, you need to convert
On 9/2/2011 4:15 PM, Jeremy Kister wrote:
since www.ilbcfreeware.org is broken, asterisk installs that want ilbc
are failing.
it appears this was done on purpose since Google bought them.
Asterisk is going to need fixing. I'll probably hook something up.
http://www.webrtc.org/ilbc-freeware
In asterisk 1.4 I had:
exten = s,n,Answer()
exten = s,n,SetMusicOnHold(default)
But in 1.6 1.8 I think don't need to use: SetMusicOnHold(default) (beside it is
deprecated) as it is default.
In 1.6 and UP I think it is: Set(CHANNEL(musicclass)=
Can somebody explain what do they mean by CHANNEL?
Hi guys,
does res_jabber support realtime? if not, are there any plans to?
Kelvin Chua
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
On 9/2/2011 8:33 PM, Jeremy Kister wrote:
Asterisk is going to need fixing. I'll probably hook something up.
https://issues.asterisk.org/jira/browse/ASTERISK-18412
a patch and brief instructions are now available at the above URL.
--
Jeremy Kister
http://jeremy.kister.net./
--
Hi,
I know that by using vi editor we can edit all the Linux files but I want to
use Php. So that from web page anyone can make some account into asterisk
server.
But thanks for your reply. And i have completed that task yesterday after
sending e-mail.
On Sat, Sep 3, 2011 at 12:53 AM, C F
Hi Virendra,
That's great
could you please share the sample for sip.conf and extensions.conf?
On Sat, Sep 3, 2011 at 10:09 AM, virendra bhati virbh...@gmail.com wrote:
Hi,
I know that by using vi editor we can edit all the Linux files but I want
to use Php. So that from web page anyone can
Hi all, i am beggining on asterisk and i would like to run my asterisk on
Ubuntu server 10.04 + asterisk 1.8.6.0 + dahdi
I have one tdm400p card with one fxo module for testing. I have connected
the pstn line to the fxo port.
I was looking for documentation and i found the official book, but it
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