for
the outbound leg but the AGI variables do not include the DNID equivalent.
Any ideas?
/Obelix
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and FXS options have me at a loss.
What is the X100p capable of, and how does it relate to the more expensive
versions?
/Obelix
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Is Asterisk 1.4 on schedule for release in July?
/Obelix
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If you need to use prerecord voicer files for G729 codec, how do you configure
them?
Do they have to be specially named, copied to their own folder or something?
Can asterisk automatically find them even if you use standard names?
/Obelix
Is the current G729 codec compatible with Asterisk trunk?
/Obelix
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to move round or relink.
Is there way I can find out what file names are compiled and where they are
installed by an asterisk installation?
Does anyone have something like this working?
/Obelix
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What and When is the next version of Asterisk?
/Obelix
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Which Actions and events to the read/write options in manager.conf give access
to, ie the options below.
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Are they documented somewhere?
/Obelix
will be able to do the rest myself.
You can check that info in www.asterisk.org or voip-info.org
If you have problems applying the patch let me know, may be I can make
you a patch for the 1.2.7.1 specially.
Regards
On 5/19/06, Obelix [EMAIL PROTECTED] wrote:
Quoting Moises Silva [EMAIL
specially.
I am rather ignorant about how the versioning works and how obtain the patch. I
would be much obliged.
Regards
/Obelix
Regards
On 5/19/06, Obelix [EMAIL PROTECTED] wrote:
Quoting Moises Silva [EMAIL PROTECTED]:
Hi,
I am ready to try out this patch, both PlayDTMF and SendDTMF
to it or use
the latest from SVN.
Can you give me a list of commands I should apply to SVN?
/Obelix
I have uploaded a patch for some manager events that allow to know
when DTMF has been received or sent. Please take a look at this:
http://bugs.digium.com/view.php?id=6082
and if you can
Does Asterisk have voice prompts for the following.
1. The number you dialled is not available. Please try again later.
2. The number you dialled is not recognised
/Obelix
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Is there a way to monitor a call for DTMF tones an trigger some actions based on
those DTMF tones?
I am interested in any arbitrary DTMF tones, not those related to the usual PBX
functions like call transfer, music on hold, call diversion etc
/Obelix
Is there a way to monitor the DTMF tones on a channel?
I have a prepaid application working in asterisk. When the user dials a call and
wants to cancel the call before it is answered, there is now way to do it
without hanging up and redialling the access number.
Is there way to monitor a
Is there a way to monitor the DTMF tones on a channel?
I have a prepaid application working in asterisk. When the user dials a call and
wants to cancel the call before it is answered, there is now way to do it
without hanging up and redialling the access number.
Is there way to monitor a
Which USB Phones, come with G729 support?
I am looking for one which has the G729 in the software installed on disk
itself, so that if the users can use onscreen dialling with headphones if they
want.
/Obelix
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Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application in which you want to make another
call, because the current number it is not being answered, and you don't want
to hangup before dialling another number.
/Obelix
Is there a way to terminate a ringing call before it is answered?
I am speaking of prepaid card application in which you want to make another
call, because you current number it is not being answered, and you don't want
to hangup before dialling again.
/Obelix
How can you check if transcoding is configured to work properly on a system?
Is there a way of knowing that transcoding is configured properly and is giving
some output to indicate so?
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Quoting Jean-Michel Hiver [EMAIL PROTECTED]:
I don't think there is any way around this problem.
This is more a question of the terms of the agreement between both parties as to
what happens if a particular number was matched by a prefix not listed in the
providers A-Z.
A provider must list all
Is there a key sequence to stop a call as its ringing, before the call is
answered?
The * key stops a call after it is answered, but I'd like a way to cancel the
call during the ringing phase.
/Obelix
This message was sent
, how which parameters
contain the right settings.
This appears to be the relevant line. What changes does it require to change the
setting?
{ AST_FEATURE_DISCONNECT, Disconnect Call, disconnect,
*, *, builtin_disconnect, AST_FEATURE_FLAG_NEEDSDTMF }, };
/Obelix
Hello,
The idea
will upgrade or tweak the sources...
Bogdan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf
How do you check whether a channel is active and the number of calls on it?
Is it simple and complicated?
/Obelix
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BLAH
zap show channel BLAH
On 1/3/06, Obelix [EMAIL PROTECTED] wrote:
How do you check whether a channel is active and the number of calls on it?
Is it simple and complicated?
/Obelix
This message was sent
I want to modify features.conf to set a different key to hang up call. Rather
than the usual * key. I gather it involves some application map settings etc.
Does anyone have a clue? I have read the docs but can hardly find any examples.
Regards
Obelix
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 4:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to set features.conf to change the hangup key.
I want to modify features.conf to set
:[EMAIL PROTECTED] On Behalf Of Obelix
Sent: Saturday, December 31, 2005 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How to set features.conf to change thehangup
key.
Quoting Bogdan Moldovan [EMAIL PROTECTED]:
Does this option work
I need some information on the syntax used in features.conf.
I want to use the applicationmap to assign different buttons to the Hangup()
command. Where should I look?
Obelix
I want to use '##' to terminate a call instead of the '*' used by the Dial
command's H option.
Is there a way
Quoting Matt Riddell [EMAIL PROTECTED]:
Hi Matt,
I have read up on features.conf but the documentation is rather sparse.
Can you show a more detailed example of the method involved?
Obelix wrote:
I want to use '##' to terminate a call instead of the '*' used by the Dial
command's H
I want to use '##' to terminate a call instead of the '*' used by the Dial
command's H option.
Is there a way to change the key or use another option to achieve the same
effect?
/Obelix
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Quoting Matt Riddell [EMAIL PROTECTED]:
That is a part of Asterisk I am not yet familiar with.
I will give it a try
Thanks
Obelix wrote:
I want to use '##' to terminate a call instead of the '*' used by the Dial
command's H option.
Is there a way to change the key or use another
Is there a source of Asterisk programming techniques in various languages - ie
Asterisk scripting in general, not the main Asterisk program itself?
Obelix
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itself.
I am looking for more scripting techniques
Obelix schrieb:
Is there a source of Asterisk programming techniques in various languages -
ie
Asterisk scripting in general, not the main Asterisk program itself?
What you are looking for is probably AGI (the Asterisk Gateway
Interface
-- warning_sound=timeleft
-- end_sound=UNDEF
Obelix
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Quoting Thor Atle Rustad [EMAIL PROTECTED]:
I think there are some settings which make the asterisk client appear to be
something else, in the same way some browsers spoof Microsoft Internet
Explorer. I am not quite sure what they are. Ask around some more.
I subscribed to a service based in
-- warning_sound=timeleft
-- end_sound=UNDEF
Obelix
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not
apply to all calls.
Can the extensions.conf option be applied on a peer, or user basis?
Obelix
Obelix wrote:
I want to play a sound on the connection of a call using the
LIMIT_CONNECT_FILE
option but can't find any examples.
Does anyone have any examples? Examples of the usage
Quoting Obelix [EMAIL PROTECTED]:
I tried this dial command
SIP/providername/002345678|42|HL(2658:61000:3:LIMIT_CONNECT_FILE=soundfile)
-- Limit Data:
-- timelimit=2658
-- play_warning=61000
-- play_to_caller=yes
-- play_to_callee=no
-- warning_freq=3
I want to play a sound on the connection of a call using the LIMIT_CONNECT_FILE
option but can't find any examples.
Does anyone have any examples? Examples of the usage of the other LIMIT_xx
options would also be appreciated.
Obelix
?
Obelix wrote:
Quoting Matt Riddell [EMAIL PROTECTED]:
They are not DTMF tones they are 1100Hz, 400Hz and 440Hz tones, used in
call
shop systems. They monitor call progress and trigger billing.
How long do you want them? Just pure sine?
--
Cheers,
Matt Riddell
Quoting Matt Riddell [EMAIL PROTECTED]:
They are not DTMF tones they are 1100Hz, 400Hz and 440Hz tones, used in call
shop systems. They monitor call progress and trigger billing.
Regards
Obelix
Obelix wrote:
Quoting Matt Riddell [EMAIL PROTECTED]:
Is there a way of converting the play
Quoting Matt Riddell [EMAIL PROTECTED]:
Is there a way of converting the play tone to a gsm file which can be played
using the A option?
Obelix wrote:
Is it possible to get Asterisk to issue a Playtones when an outgoing call
is
answered? The examples indicate what happens when
Is it possible to get Asterisk to issue a Playtones when an outgoing call is
answered? The examples indicate what happens when an incoming call is answered.
/Obelix
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Does asterisk have a module for generating tones, or a set of prerecorded GSM
tones, like 1100Hz tones et cetera?
/Obelix
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Quoting Erik [EMAIL PROTECTED]:
Where can I download it from? I searched the lists and the web for any reference
to it and there is no mention of it.
Regards
Obelix
app_milliamp is your friend
Obelix wrote:
Does asterisk have a module for generating tones, or a set of prerecorded
GSM
Quoting Obelix [EMAIL PROTECTED]:
I did some searching and I found them in the Asterisk 1.09 Sounds distribution.
I simple searched google for asterisk pounds.gsm. So much for forgetting about
the obvious.
Silly me :-)
Is there a .gsm file for announcing UK pounds and pence after the credit
I'd like to know whether is possible to play some white noise or low level
background noise to keep a connection up. One of my providers have an RTP
timeout which kicks in quite quickly, and I need to know how to avoid it.
Are there some known means of stopping this?
Regards
/Obelix
Is there a .gsm file for announcing UK pounds and pence after the credit
remaining prompt, besides the dollar and cents file?
/Obelix
This message was sent using IMP, the Internet Messaging Program
Is it possible to play or generate some white noise, down an Asterisk call? Some
calls I am making are terminating if there is an RTP timeout.
Is there some file I can play during the call to fix this?
/Obelix
This message
Is there a way to config a sip user so that he appears to be connecting from a
different IP address?
I want to use different IP addresses to authenticate different accounts with
service providers rather than the username/password combo.
Are there SIP settings to allow that?
/Obelix
Quoting René Enskat [Teamware GmbH] [EMAIL PROTECTED]:
In my experience most AGI problems I had came from other info sent to the
terminal via verbose commands and other stdout output. There is some info on
the voip-info wiki about using AGI.
I use the phpagi 2 library, and carefully setting up
Quoting Chuck Bunn [EMAIL PROTECTED]:
Check your firewall configuration. New versions of Linux come with tighter
default firewall configurations.
Check these notes from Redhat to see what processes if any are listening on the
relevant ports.
variables and see if the hashes are what you expect them
to be.
I have the phpagi 2 library too.
So what did you change in details there to mute the vebrose things?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Obelix
Gesendet: Montag, 17
I have compiled the OH323 module for my system.
When can I find some info on how to properly configure it?
I haven't read any info for its configuration, and I need some starting info.
Were do I start?
Obelix
This message
I want to add H323 support to my asterisk setup. What are the pros and cons of
the available modules, h323, oh323 and ooh323 and which is the best one to go
for?
Obelix
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I have been receiving a lot these 488 Not Acceptable Here from a number of
providers. What could the problem be?
What is the most common cause of this message?
This message was sent using IMP, the Internet Messaging Program.
My Asterisk PBX seems unable to receive DTMF information via SIP. I have tried
all the various methods, rfc2833, inband and info and they all don't seem to
work. IAX2 works fine. Is there something I must be missing
?
/Obelix
Quoting Ray Van Dolson [EMAIL PROTECTED]:
How can you determine which codecs are acceptable to them?
Do they have a way of indicating it?
Perhaps they dont' like the codec you're offering in your INVITE message?
Ray
On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote:
I have been
How does one check what codec translations are in use in a call?
I am connecting to sip system which says 488 4XX Not Acceptable Here. I don't
know what is stopping the call from being accepted and I'd like to know if
there are codec issues involved.
/Obelix
When I try to dial through a pbx I receive this message
to 216.127.66.119:0
Oct 8 23:53:48 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81331cc
(len 670) to 216.127.66.119 returned -1: Invalid argument
Retransmitting #5 (no NAT):
The line is silent and nothing happens.
/Obelix
I have been trying to connect via sip and things don't seem to work. What do
messages like this mean?
Oct 9 00:33:57 WARNING[22849]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x81ab834
(len 361) to 216.127.66.119 returned -1: Invalid argument
Oct 9 00:33:58 WARNING[22849]: chan_sip.c:694
If you have configured Asterisk to remote to a SIP provider, how do you verify
that the registration has been successful?
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I want to backup Asterisk based on 1.07 to install 1.09 and 1.2beta on another
server.
Which files and folders do I have to backup, in order to restore if things don't
work right?
This message was sent using IMP, the Internet
Quoting Anders Svensson [EMAIL PROTECTED]:
this page might help.
http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning
Hi all!
Is it possible to have a setup with a server only dedicated for transcoding
from ulaw/alaw to G729. What is the capacity of a server like that in
voipreach.net - are they functioning, I sent them a few emails and they did not
reply.
are they operational?
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VoIP - VoIP?
All the above refer to a VoIP setting.
4. Is there a difference between bridging and transferring?
/Obelix
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on the Ethernet cable to allow it plug directly
into a proper TDM connection?
Will someone please enlighten me.
/Obelix
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) :-).
If you have a critical comment to make, by all means do so, but try to go some
way in helping with the question.
I also have no intention to skimp on the pennies.
Yours Obelix.
PS. You will get invited to dinner now.
That was long, wasn't it :-)
sure you have one sitting around somewhere you
I know this must look like a very trivial question, but what are SIP Proxies and
H323 Gatekeepers, and what do they add to Asterisk?
Why should Asterisk need them?
/Obelix
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I have been reading a number of the past threads about G.729 licensing., about
how the registration keys are linked to the network configurations, limited
number of registrations etc, etc.
Is there no reason why the decoding can't be done in with some Asterisk
compatible hardware, so that once
configuration that I don't know about?
IAX connects okay
/
Obelix
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Is it possible to use G729 on asterisk without the license?
It is to connect devices which use the codec to termination providers in a phone
card application.
Will decoding the DTMF tones from the caller require G729 processing?
considering everything is it only Microsoft's lib.exe which can do the
job?
Obelix wrote:
I want to compile the G729 codec to try it out with firefly.
I don't have Visual C++ 6 compiler. Is there a way I can obtain the
link.exe
alone for use with cygwin, or a substitute program?
I don't look
I want to organize my agi scripts by putting them in separate subdirectories.
Is this permissible, or it necessary for at list the initial file to be in the
agi-bin directory?
In case I prefer to move them outside the main folder what syntax should I use
for the folder? will it be worked off
to, and what else it can be used for.
Are these some info sources which go into these areas in depth?
Obelix
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Asterisk
to, and what else it can be used for.
Are these some info sources which go into these areas in depth?
Obelix
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Quoting Darren Wiebe [EMAIL PROTECTED]:
They never truly got their act together. I remember checking my CDR and
realising that they were charging my 0800 numbers in 1/100 of a cent instead of
cents. It is a pity their DTMF tones were not working for me. At least I would
have gained something from
Quoting Tony Hoyle [EMAIL PROTECTED]:
I think I installed the framework some time ago. I will hunt for the install
location and see if I will find it.
Thanks
Obelix wrote:
I want to compile the G729 codec to try it out with firefly.
I don't have Visual C++ 6 compiler. Is there a way I can
I want to compile the G729 codec to try it out with firefly.
I don't have Visual C++ 6 compiler. Is there a way I can obtain the link.exe
alone for use with cygwin, or a substitute program?
I don't look forward to installing the whole Visual C++ just for the link.exe
timestamp the events themselves.
Obelix
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When I check the received email, my user name does not appear on the From list.
All it says is To: asterisk-users@lists.digium.com.
Is there something configured wrongly in my mail client, or is it coming from
the mailing list configuration
Quoting Obelix [EMAIL PROTECTED]:
Is this question too difficult, or is it simply one that only a few users have
experienced?
My CDR is displaying wildly inaccurate results.
When I make a call the CDR records the time between connecting into the
server and hanging up, instead of recording
My CDR is displaying wildly inaccurate results.
When I make a call the CDR records the time between connecting into the
server and hanging up, instead of recording the time between dialling
from the server to the PSTN destination via VOIP termination.
It is alright to log the duration of the
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