Hi All,
I never use hint in asterisk and I don't know how to use Hints and why we
use Hints. If you give some details with example then it will be helpful for
me.
On Tue, Apr 19, 2011 at 10:52 PM, Danny Nicholas wrote:
>--
>
> *From:* asterisk-users-boun...@li
Try this
http://www.freepbx.org/support/documentation/administration-guide/creating-an-ivr
On Wed, Apr 20, 2011 at 8:11 AM, Kaushal Shriyan
wrote:
> Hi,
>
> Is there a step by step guide to Configure IVR(Inbound and Outbound) in
> AsteriskNow using FreePBX ?
>
> Thanks
>
> Kaushal
>
> --
> __
In article
<2658e54b540d284981ea57e6a549ea70abd1fdf...@inblrk77m1msx.in002.siemens.net>,
Deka, Rajib IN MAA SL wrote:
>
> The requirement is little complicated as it is H/W specific.
> Basically we are integrating a radio gateway (SIP) with asterisk. The gateway
> will be
> connected to a meetm
hey try with app_rpt in asterisk
regards
dhaval
On Wed, Apr 20, 2011 at 1:24 PM, Tony Mountifield wrote:
> In article <
> 2658e54b540d284981ea57e6a549ea70abd1fdf...@inblrk77m1msx.in002.siemens.net
> >,
> Deka, Rajib IN MAA SL wrote:
> >
> > The requirement is little complicated as it is H/W spe
On Tue, Apr 19, 2011 at 04:50:39PM +0200, Mosiuoa Tsietsi wrote:
> Hi all,
>
> I downloaded a copy of bristuff 0.4.0-RC11 on my Ubuntu 10.04.2 LTS server
> machine with 2.6.32-24-generic-pae kernel.
Please, do yourself a favour and use Asterisk 1.8 with latest libpri
instead.
Bristuff uses a for
Outbound call only you can make like dial out rules, IVR is only for Inbound
calls. In outbound I dont how you are asking this IVR facility. You mean
like voice broadacasting? like dial the number and playing a voice file?
On Wed, Apr 20, 2011 at 2:14 PM, Kaushal Shriyan
wrote:
> Hi Gopal,
>
> Is
On Wed, Apr 20, 2011 at 2:56 PM, Gopalakrishnan A.N wrote:
> Outbound call only you can make like dial out rules, IVR is only for
> Inbound calls. In outbound I dont how you are asking this IVR facility. You
> mean like voice broadacasting? like dial the number and playing a voice
> file?
>
>
Hi G
Thanks a lot Tony and Dhaval for your much appreciable suggestions.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
Il 19/04/2011 23:41, Kevin P. Fleming ha scritto:
> If you are the receiver of the call (and thus they are the sender of the
> call), it is *your* system's responsibility to initiate the switch to
> T.38, not theirs.
Are you sure? So what's faxdetect=t38 for?
Cheers,
Darkbasic
--
___
is your problem solved or not
On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL <
rajib.d...@siemens.com> wrote:
> Thanks a lot Tony and Dhaval for your much appreciable suggestions.
>
> Regards,
> Rajib
>
> Rajib Deka
> SIEMENS Ltd.
> Robert V Chandran Tower, First Floor, West Wing,
> #149,
Op 20-04-11 01:07, Matt Riddell schreef:
On 20/04/11 1:58 AM, Mark Deneen wrote:
2011/4/19 Niccolò Belli:
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
What abo
In article ,
DHAVAL INDRODIYA wrote:
>
> is your problem solved or not
It will take a lot more time than that to try out the suggestions!
> On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL <
> rajib.d...@siemens.com> wrote:
>
> > Thanks a lot Tony and Dhaval for your much appreciable sug
If you want to create voice broadcasting kind of thing, you have to write
your own dialplan or agi related stuffs.
Following are the procedures
1. Need to make a script for autodial bunch of numbers from database or
from call file
2. Need to check the bridged call
3. If the call is br
On 04/20/2011 04:55 AM, Niccolò Belli wrote:
Il 19/04/2011 23:41, Kevin P. Fleming ha scritto:
If you are the receiver of the call (and thus they are the sender of the
call), it is *your* system's responsibility to initiate the switch to
T.38, not theirs.
Are you sure? So what's faxdetect=t38
Hi All,
Again i am posting this issue because i am curious why its happening, Do you
have this error with 1.8.x version please let me know.
If i dial non-exist number or peer i got following error on my CLI . even i
don't have this IP address in my system this wired number coming from where
Hello,
We have a sip trunk between our voip operator and our asterisk 1.6.2.9
We have no problem during voice communications.
But we can not send any t38 fax via this gateway.
We tried to trace the error made some tests..
There are 2 main tests we tried to do.
As i learned their voip path is like
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, April 20, 2011 2:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to know how many calls are into
On 04/20/2011 10:02 AM, Oguzhan Kayhan wrote:
Hello,
We have a sip trunk between our voip operator and our asterisk 1.6.2.9
We have no problem during voice communications.
But we can not send any t38 fax via this gateway.
We tried to trace the error made some tests..
There are 2 main tests we tr
hello all,
I have installed centos 5.5 ( linux text) and I have updated it with
# yum install bison bison-develèok
# yum install ncurses ncurses-devel==èok
# yum install zlib zlib-devel===èok
# yum install openssl openssl-deve===èok
# yum install gn
I would recommend you to use the binary packages. Method described here:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29
They are reliable, easy to maintain, and this is basically one of the
best way to get a clean Asterisk Installation.
N
Hello Guys,
In the case of a multiserver environment for outbound
automatic calls, can you share you experience and preference between call
files and ami originate ?
thanks
--
*Adolphe CHER-AIME
Network / VoIP Engineer
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3449-
Hi,
I have a problem with IAX accounts...
I set up a few months ago an Asterisk server, with mysql support to load iax
accounts.
Settings seems fine because apparently the system works as expected.
Yesterday I tried to add another iax account in the iax.conf directly. And I
have a problem with thi
thanks for your response
i have flow this link
$ vi /etc/yum.repos.d/centos-asterisk.repo
[asterisk-current]
name=CentOS-$releasever - Asterisk - Current
baseurl=http://packages.asterisk.org/centos/$releasever/current/$basearch/
enabled=1
gpgcheck=0
#gpgkey=http://packages.asteris
My trusty support group,
Any one come across this in there logs regarding the equals extension?
[Apr 16 13:29:42] NOTICE[23047] chan_sip.c: Call from '' to
extension '' rejected because extension not found.
I do not know how they would generate this.
--
_
On 11-04-20 01:16 PM, ERIC HERRON wrote:
My trusty support group,
Any one come across this in there logs regarding the equals extension?
[Apr 16 13:29:42] NOTICE[23047] chan_sip.c: Call from '' to
extension '' rejected because extension not found.
I do not know how they would gen
but when i finished i found a file in etc asterisk but is empty
You found "a file" in "etc asterisk"?, what file?
My English is not very good (sorry for that), but if you meant that you
found '/etc/asterisk' directory, but it was empty, then you are missing
the configuration files, do this
There is nothing in the dialplan show that remotely resembles this.
asterisk -rx "dialplan show" | grep ==
returns nothing.
Any other ideas?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of ERIC HERRON
> Sent: Wednesday, April 20, 2011 1:14 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] asteris
Hello,
I want that people from other servers like ekiga.net can make calls to
my users. When I do an "allowguest=no" then people from other domains
cannot call me. So I think I need "allowguest=yes".
Maybe something like this?
-
include => users
include => users
exten=_0.,1,Dial(SI
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Paul van der Vlis
> Sent: Wednesday, April 20, 2011 2:41 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] allowguest=yes, how?
>
> Hello
Hey guys!
I have just testing voicemail to text.. and i have an issue, Hope you will help
to fix them. I am following
http://www.circuitid.com/blog/how-to-setup-voicemail-transcription-with-asterisk-using-pocketsphinx
pocketsphinx application but issue is file format i am getting following err
On 11-04-20 02:13 PM, ERIC HERRON wrote:
There is nothing in the dialplan show that remotely resembles this.
asterisk -rx "dialplan show" | grep ==
returns nothing.
Any other ideas?
That is not the command I asked you to run, nor does this help you to
resolve your issue.
If you are seeing
On 11-04-20 12:20 PM, Adolphe Cher-Aime wrote:
In the case of a multiserver environment for outbound
automatic calls, can you share you experience and preference between call
files and ami originate ?
I prefer using the AMI as I have better call control. I also get to
monit
On Wed, 20 Apr 2011, satish patel wrote:
pocketsphinx application but issue is file format i am getting following
error when i try manually. so do i need to convert this file everytime
in 8khz format ???
Google says this will help:
http://nsh.nexiwave.com/2010/09/voicemail-transcrip
Hey Thanks for that reply after add following option it works but the text
output is totally different.. what its totally different is this dictionary
problem ?
-hmm /var/lib/asterisk/communicator -samprate 8000
In audio file its just: Hello satish this is test message
0: i starte
Thank you for your answer. I also prefer AMI for its flexibility.
However, i have an application developped in PHP used to make more than
10 calls a day by group of 120 concurrent calls. My problem with AMI is
that client keeps disconnected to AMI server. I use astmanproxy as proxy
serv
On Wed, Apr 20, 2011 at 4:35 PM, satish patel wrote:
>
> Hey Thanks for that reply after add following option it works but the text
> output is totally different.. what its totally different is this dictionary
> problem ?
>
> -hmm /var/lib/asterisk/communicator -samprate 8000
>
> In audio file it
Hi there,
I need a Python interface to asterisk manager for my own project. The
voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+manager+Examples)
lists 4 python projects for this purpose: Fats, py-Asterisk, pyst and
StarPy. Because my project is rather small and I don't want to involve
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Ye Liu
> Sent: Wednesday, April 20, 2011 4:32 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] py-Asterisk or pys
On 11-04-20 04:44 PM, Adolphe Cher-Aime wrote:
Thank you for your answer. I also prefer AMI for its flexibility.
However, i have an application developped in PHP used to make more than
10 calls a day by group of 120 concurrent calls. My problem with AMI is
that client keeps disconnecte
On 11-04-20 05:32 PM, Ye Liu wrote:
Hi there,
I need a Python interface to asterisk manager for my own project. The
voip-info.org (http://www.voip-info.org/wiki/view/Asterisk+manager+Examples)
lists 4 python projects for this purpose: Fats, py-Asterisk, pyst and
StarPy. Because my project is rat
Thanks Paul,
I will take a look at twisted i will let you know.
Regards
On Wed, Apr 20, 2011 at 5:38 PM, Paul Belanger wrote:
> On 11-04-20 04:44 PM, Adolphe Cher-Aime wrote:
>
>> Thank you for your answer. I also prefer AMI for its flexibility.
>> However, i have an app
Thanks a lot guys.
We planning to use different approach like maintaining a separate extension for
sending and receiving in dialog SIP:MESSAGE using SendText and RECEIVE TEXT.
Regards,
Rajib
DHAVAL INDRODIYA wrote:
>
> is your problem solved or not
It will take a lot more time than that to tr
hi,
Hint will work all VoIP hardware or specific hardware device ?
I am planing to using CISCO 79XX series so please suggest me..
And What about softphone ?
On Wed, Apr 20, 2011 at 8:57 PM, Danny Nicholas wrote:
>--
>
> *From:* asterisk-users-boun...@lists.digiu
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