NVM,
This is happening for a specific destination numbre only.
I am considerting that destination number have a wired problem.
Thanks,
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Mon, 31 Jul 2006 08:30:
Hi,
From this morning, I am having a wired issue.
If I dial a number from my asterisk exteion handset, it end up at different
number than I dial and its happening everytime I dialed. I havn't change any
thing in my asterisk box.
And also, I tried my land phone handset which is not forked to a h
Why don't you do something like this:
exten => 12345678,1,Dial(10)
exten => 45874521,1,Dial(11)
exten => 32544884,1,Dial(12)
replace Dial(10) and so on with apppriate extension.
Thanks,
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> F
Hello,
To detect an answering machine I have found following two commands,
BackgroundDetect (comes with asterisk)
MachineDetect (asterisk add-ons)
First question, does BackgroundDetect works well with g729?
I havn't try MachineDetect yet, what is the benefit of MachineDetect over
BackgroundDet
When I read variables in AGI scripts, I see only the follwing 13 variables
agi_request
agi_channel
agi_language
agi_type
agi_uniqueid
agi_callerid
agi_dnid
agi_rdnis
agi_context
agi_extension
agi_priority
agi_enhanced
agi_accountcode
beside these, I found following variables documented on severa
@lists.digium.com
> Subject: Re: [Asterisk-Users] wav to g729
>
> Innocent Evil wrote:
>> I prefer something 'sox' like program.
>>
>>
>>
>> --
>> You don't have any choice, you already made it before you came here.
terisk-Users] wav to g729
>
> Innocent Evil wrote:
>> I prefer something 'sox' like program.
>>
>>
>>
>> --
>> You don't have any choice, you already made it before you came here.
to g729
>
>
> Try the new conversion module from redice li ..it is greate!
>
> Miklos
>
>
> IPFONE TELEFONIA IP
> Rua Caio Graco 735 São Paulo SP
> IPBX - +55 11 3488-3800
> http://www.ipfone.com.br
> [EMAIL PROTECTED]
>
> Balbus balbum intellegit
: [Asterisk-Users] wav to g729
>
> Innocent Evil wrote:
>> hello,
>>
>> how can I convert my existing wav file to g729.
>> Currently, i have all of them converted to gsm.
>> Isn't it right, If I had all my sound files in g729 format, my server
>> would use
hello,
how can I convert my existing wav file to g729.
Currently, i have all of them converted to gsm.
Isn't it right, If I had all my sound files in g729 format, my server would use
less resource and less channels.
I have couple of g729 liscences from digium.
Thanks,
--
You don't have any c
> Sent: Wed, 21 Dec 2005 17:16:11 -0500
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] caller_id and law
>
> trixter aka Bret McDanel wrote:
>
> >On Wed, 2005-12-21 at 13:39 -0800, Innocent Evil wrote:
>>
>>
> >>I would l
ROTECTED]
> Sent: Wed, 21 Dec 2005 13:47:18 -0800
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] caller_id and law
>
> On Wed, 2005-12-21 at 13:39 -0800, Innocent Evil wrote:
>> I would like to send auto invitation to members of one of my community
>>
I would like to send auto invitation to members of one of my community
organization.
I will use a trird party voip provider to make those call out.
Question, what caller_id I can pass to that auto invitation message.
What does law says?
--
You don't have any choice, you already made it before yo
I was also following this thread.
Would anybody please tell, what would be configuration file if mysql is a different machine than asterisk box?
Thanks,
--You don't have any choice, you already made it before you came here.
-Original Message-From: [EMAIL PROTECTED]Sent: Mon, 12 Dec 2
I have an AGI script but goal was little different.
But my script can do this by tweaking little bit.
I have a two tables,
caller_numbers and statuses
every new number is logged in caller_numbers.
If that caller call next time, based on the status call can be route to
1. Voicemail
2. Hangup
3. B
You can accomplish password per extention by using an AGI script.
model would be,
keep extension and password in a table
Execute a simple script to authenticate before dial-out
You can also accomplish dial-out time from an AGI script.
Feel free to ask if you need further help.
Thanks,
--You
D]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Tuesday, 6 December 2005 08:11
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] h323 vs oh323
>
>
> Try chan_oh323 and if it is not ok, try chan_h323
> B
Hello,
Would you please share your experience regarding h323 and oh323 in asterisk.
I am confused to choose one.
Thanks,
--
You don't have any choice, you already made it before you came
here.___
--Bandwidth and Colocation provided by Easynews.com -
or example, and you might
>
>
> exten => s,7,System(echo "Call from ${CALLERIDNUM}" >> /tmp/something)
>
>
> but I haven't tested it yet. Can someone confirm this?
>
>
> Innocent Evil wrote:
>> Yes,
>>
>> I have someth
Yes,
I have something like this in my extension.conf
#include verizon.conf
and in verizon.conf file have verizon related dialplan
Thanks
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 01 Dec 2005 15:5
What you wanna to do if there have more than 2 parties in the conversation ? !!
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 01 Dec 2005 19:27:45 +0100
> To: asterisk-users@lists.digium.com
> Subject:
Try,
Set(CALLERIDNAME="Innocent Evil")
Thanks,
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 01 Dec 2005 12:48:13 -0500
> To: asterisk-users@lists.digium.com
> Subj
Hi Scott,
Yes, its possible
pass 'm' option to Dial command for MusicOnHold
If destination is unreachable, you need to get the return value of Dial
and from that value you will know whether a call was connected or not. Based
on that value you can execute Dial again or not.
You can put everything
> On Mon, 2005-11-28 at 12:24 -0800, Innocent Evil wrote:
>> What is the purpose of cdr_manager.conf?
> cdr_manager.conf allows you to configure asterisk to send call detail
> records (cdr) via the Manager API.
>
>> How I can configure it?
> to enable CDR via Man
Hi,
I am trying to execute the following asterisk command from one of my AGI script.
By providing 'C' flag, I exected CDR would reset.
Problem is, CDR was reset but CDR didn't grab destination number (extension)
from the Dial command.
Well my AGI script was executed after answering a call on a c
Hello,
While I was trying to get right CDR record from AGI script, I came across
cdr_manager.conf
I am trying to learn about cdr_manager.conf
What is the purpose of cdr_manager.conf?
How I can configure it?
I did google, really didn't have very good luck.
Would anybody please write couple of s
Hi,
I have an AGI script that is called after receving a call on a channel.
And my script executel AGI cmd Dial to make another call.
Is there any reason not to have CDR record for the call that was initiated in
the AGI script?
Or I am just missing something basics .
Thanks,
--
You don't have
>
>> 1. Add an option in Dial, Say R. to pass rate (price per minute) for the
>> call
>> to do that I will have to modify app_dial.c
>> 2. Dial would use option R to set cdr
>> 3. I will also need to add one more function in cdr.c, say something
>> like ast_cdr_setrate(...)
>
> None of this i
Hello,
I am trying to enhance my cdr records.
What I am trying to are:
1. Add an option in Dial, Say R. to pass rate (price per minute) for the call
to do that I will have to modify app_dial.c
2. Dial would use option R to set cdr
3. I will also need to add one more function in cdr.c, say som
you can use 'w' option with 'Dial' on 1.2.x
I don't think w do anything like 'wait', If I am wrong, correct me someone
please
According to app_dial.c
"w- Allow the called party to enable recording of the call by sending\n"
" the DTMF sequence defined for one-touch recording in
timelimit;
>
> Append $timelimit onto the end of your dialcommand. You can look at the
> code for ASTCC in the asterisk cvs or look at astpp-callingcard.agi in
> the cvs code available @ www.aleph-com.net/astpp
>
> Darren Wiebe
> [EMAIL PROTECTED]
>
> Innocent Evil wrote:
>
nternational Dialing Code
>
> Those came from astbill. I will make the changes and reupload, I have
> gotten a few more changes as well.. Thanks :)
>
> On Mon, 2005-11-21 at 22:03 -0800, Innocent Evil wrote:
>> Lots of country have wrong prefix.
>> Andorra,376 sh
Comeo'n AGI guys..
Please say something.
>
> Hi,
>
> Using AUTOHANGUP, I can force a call duration within a time limit.
> I would like to playback a message before 1 minute of autohangup.
> How can I accomplish it?
> Would anybody please give me right direction.
>
> Thanks,
>
>
>
>
> You do
erisk-Users] International Dialing Code
>>
>> On Sun, 2005-11-20 at 15:20 -0200, Hermann Wecke wrote:
>>> Innocent Evil wrote:
>>>> I am trying to download a list of international dialing codes.
>>>> Would anybody please post a link to get it
>>&
-0200, Hermann Wecke wrote:
>> Innocent Evil wrote:
>>> I am trying to download a list of international dialing codes.
>>> Would anybody please post a link to get it
>>
>> Google IS your friend. Did you try?
>
> http://www.0xdecafbad.com has one in the fi
Hi,
Using AUTOHANGUP, I can force a call duration within a time limit.
I would like to playback a message before 1 minute of autohangup.
How can I accomplish it?
Would anybody please give me right direction.
Thanks,
You don't have any choice, you already made it before you came
here.
I am trying to download a list of international dialing codes.
Would anybody please post a link to get it
Thanks in advance.
___
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http:
Hello,
My phone's VMWI (Visual Message Waiting Indicator) is able to detect SDT
message signal.
But how I would configure asterisk to send SDT message signal to a certain
extension?
Thanks,
___
--Bandwidth and Colocation sponsored by Easynews.com --
Why don't you write a couple of lines AGI scripts that will call asterisk
command WAIT(5)
Thankx
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Tue, 27 Sep 2005 13:42:31 +0200
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] wait before accepting the call
>
>
just an update for all curious folks
I just replaced my FXS card and everything is working great ..
I am running asterisk as non-root user.
Moreover, I couldn't figure it out how/why my FXS card got damaged.
Thanks,
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Wed, 14 Sep 20
como'n folks.. ...
>
> Well, as I told earlier.. my asterisk was running great with one fxo and
> one
> fxs module of a TDM400P
> All i tried last night to run asterisk with non-root
> I must did something wrong while I was trying to do that
>
> FXO module on channel # 1
> FXS module on channel #
le release)
> and I have never seen anyone successfully ran Asterisk
> on it.
>
> If you have a choice, switch to SuSE or other Linux
> distribution such as Debian.
>
> /Y.T.
>
>
> --- Innocent Evil <[EMAIL PROTECTED]> wrote:
>
> > My TDM400 on fc4 was wor
My TDM400 on fc4 was working great..
all of sudden ..i am having the same issue ..you guys are having
all i tried to run asterisk as non-root user.. and I was able to run it as
non-root
and was able to receive and send call using asterisk..
I am not sure.. what thing I did wrong and coz all the tr
All of sudden my FXS module is not working.
I have a TDM card with one FXS and one FXO, FXO module seems working fine.
I also noticed the LED is not on for my FXS module while it is on for my FXO
module.
Sep 13 12:11:44 WARNING[9870]: chan_zap.c:887 zt_open: Unable to specify
channel 1: No such de
Hi List,
My AGI seems work well in asterisk -vvvc mode,
other than that it doesn't work.
Its seems to me, when I run asterisk as daemon (service asterisk start ..
on fc4), it doesn't know about my library path.
How can pass libray path to my AGI script or asterisk?
Thanks__
com
> Subject: Re: [Asterisk-Users] DTMF not working
>
>
>
> Innocent Evil wrote:
>
> >I am having same problem .. DTMF is not working from a SIP phone while
> >sending to Asterisk cmd VoiceMailMain.
> >
> >
> >
> Have you set DTMF to out of band RFC2
Hello,
Would you please suggest me, where can I buy g723.1 liscence in cheap.
I might need a liscence for 10-50 channels.
Thanks,___
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Asterisk-Users mailing list
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htt
gt; >>
> >>
> >>Huddleston, Robert wrote:
> >>
> >>>U joke - duh!
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>>-Original Message-
> >>>>From: [EMAIL PROTECTED]
&
t; -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Innocent Evil
> > Sent: Wednesday, August 24, 2005 3:53 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-User
to
> do the trick. The part of the string that reads "!0/100" just shuts the
> tone generator off.
>
> Rob
>
> Innocent Evil wrote:
>
> >I am having same problem .. DTMF is not working from a SIP phone while
> >sending to Asterisk cmd VoiceMailMain.
> >
hyst much simpler and possibly cheaper too. I
> believe the current IDE is 12.4K
>
>
>
> Innocent Evil wrote:
> > I would like to write AGI script in Ruby
> > Would anybody please show me right direction..
> >
> >
> > Thanks__
I would like to write AGI script in Ruby
Would anybody please show me right direction..
Thanks___
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I am having same problem .. DTMF is not working from a SIP phone while
sending to Asterisk cmd VoiceMailMain.
Would you please explain this line
"!941+1336/100,!0/100", /* 0 */
what value is what and how it affect on DTMF tone generation.
Thanks,
> I had a similar problem that seems to be ca
ncorrectly.
>
> /b
>
> On Aug 19, 2005, at 1:46 PM, Innocent Evil wrote:
>
> > my sip phone have dtmf relay: rfc2833
> > asterisk sip.conf have dtmf relay: rfc2833 in associated context.
> >
> > I tried with Inband.. but g729 doesn't support it. I have
or SIP
> INFO. Your SIP phone should also allow you to set how DTMF is sent
> (although it may not support all of these formats.) Preferably, use
> RFC2833 or SIP INFO. Find a setting that is available on your phone and
> on *, and make sure they're set to match. Once you
Hi,
I am using Asterisk cmd VoiceMailMain to manage voice mail.
Problem is, voice mail box can't read password sent from SIP phone, but I
don't have any problem to read password from the handset attached to my
asterisk box.
Your help will be greatly appreciated.
Thanks,__
n.
Thanks,
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Fri, 19 Aug 2005 09:02:21 -0500
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Out of G.729 Decoder Licenses!
>
> Innocent Evil wrote:
> > Hello,
> >
> > I have SIP and Aster
Hello,
I have SIP and Asterisk. On Asterisk I have 2 liscenses of g729 (from digium
website)
SIP user (100) is calling another SIP user (101).
As 101 is not online, my SIP server is redirecting that call to Asterisk.
Asterisk forward it to 101's voice mail box.
SIP user 100's phone have g729 code
I noticed their mysql server is down or can't connect to mysql server.
I tried to download there cvs format price list.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 18 Aug 2005 16:04:30 -0400
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] VoipJet Proble
Please change the subject to 'Advertisement of a VoIP Provider'
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 18 Aug 2005 11:55:50 -0700 (PDT)
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Searching For a Voip Provider
>
> Hi:
>
> Please advice me of a vo
Hello,
I was just able to connect to my festival server.. but the voice generated
by festival sounds too wired ..really.
I installed only festival, i didn't install speech_tools and couple progams
as was documented in voip-info.org
How can I tune up festival to have better voice (not as good as
uot;nil" (don't use double quotes).
>
Isn't this change going to make my festival server accept connect from
anybody?
If it is, I dont want to do that.
I just want to add my asterisk sever to festival's client list.
Thanks,
>
>
> Innocent Evil wrote:
> >
Sorry, the file I am talking about is in right place !!
but not sure what to add in festival_server script.
Thanks,
> Hello,
>
> I have installed festival (the rpm package came with fc4).
> But getting this:
> client(1) : rejected from myserver not in access list
> whenever I try to access it
Hello,
I have installed festival (the rpm package came with fc4).
But getting this:
client(1) : rejected from myserver not in access list
whenever I try to access it from asterisk .
I found in documentation:
If you see a message such as:
client(1) : rejected from myserver.mydomain.com not in acc
>
> On Monday 15 August 2005 21:08, Innocent Evil wrote:
> > I am getting this whenever I start asterisk.
> > Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device:
> > Resource temporarily unavailable
>
> sounds like your soundbard is blocked
[EMAIL PROTECTED] ~]# netstat -naptu | grep asterisk
tcp0 0 0.0.0.0:20000.0.0.0:*
LISTEN 9231/asterisk
udp0 0 0.0.0.0:27270.0.0.0:*
9231/asterisk
udp0 0 0.0.0.0:45200.0.0.0:*
9231/asterisk
udp0
> Are you saying realtime mysql is not clever? That is exactly what it is
> supposed to do.
>
BTW, how do you integrate mysql with asterisk?
any link, documention, tutorials would be greatly helpful.
Thanks,___
Asterisk-Users mailing list
Asterisk-User
How about this:
1. Put all the routes of all the providers in a MySQL table
2. Write a script with a 'clever' algorithm to find out cheapest route of
each prefix.
3. Based on #2.. make a lcr_cheapest_route.conf
4. include lcr_cheapest_route.conf in extension.conf
But I don't know, how much res
Hello,
How do you guys implement LCR in Asterisk?
Thanks,___
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I am clear with this issue.
Thanks everybody for answering me.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Mon, 15 Aug 2005 10:16:34 -0800
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Switch between FXS ports
>
> Hello,
>
> I have two FXS port on my TDM c
I am getting this whenever I start asterisk.
Aug 15 15:03:58 WARNING[11754] chan_oss.c: Read error on sound device:
Resource temporarily unavailable
This is my sound card:
Multimedia audio controller: Fortemedia, Inc Xwave QS3000A
I am not sure... what I am doing wrong.
Please help.
Thanks,_
---
> ->From: [EMAIL PROTECTED]
> ->[mailto:[EMAIL PROTECTED] On Behalf Of
> ->Innocent Evil
> ->Sent: Monday, August 15, 2005 2:17 PM
> ->To: asterisk-users@lists.digium.com
> ->Subject: [Asterisk-Users] Switch between FXS ports
> ->
> ->Hello,
> ->
> -
om
> Subject: Re: [Asterisk-Users] Switch between FXS ports
>
> Innocent Evil wrote:
> > Hello,
> >
> > I have two FXS port on my TDM card.
> > channel 4 is attached with a telco line that I use frequently. And
> channel 3
> > have another telco line.
Hello,
I have two FXS port on my TDM card.
channel 4 is attached with a telco line that I use frequently. And channel 3
have another telco line. but I dont publish that number to my friends.
If I receive a call through channel 4, how can I handover that call to
channel 3 ..so that I can keep chann
Hi,
I am trying to make an outbound call from phone attached to FXS port.
My telephone (VoIP) line is connected to FXO port (Zap/4)
Default context for channel # 4 is 'directdial'
here is part of my extension.conf
[directdial]
ignorepat => 9
exten => 9,1,Dial,Zap/4/
exten => 9,2,Congestion
incl
how many 'register =>' I can have in
sip.conf___
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keep approx. 32kb per channel..
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Fri, 5 Aug 2005 11:16:32 -0700 (PDT)
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] how may channels
>
> how many channels using codec g729 can be done by an
> internet bandwidth to
Hello,
I am sure this has been answered so many times as it is one of the most
fundamental features of Asterisk.
Here is my scenario,
I have setup my asterisk server with a TDM400p which have one FXO and FXS
card.
My SIP server is up and its working fine only in SIP network ( I used ser)
For my
.
Thanks again.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Mon, 01 Aug 2005 18:19:03 -0500
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] g729 liscence question
>
> On Mon, 2005-08-01 at 10:53 -0800, Innocent Evil wrote:
> >
I have a TDM400P with one FXS and one FXO..
how many liscence(2) I will have to buy?
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