I still haven't figured out what is the best practices or Asterisk-way
to do traditional switching between channels in Asterisk. I come from
traditional computer telephony where there are buses such as MVIP, with
streams and timeslots.
Asterisk, being born as a PBX solves most of the problems
Jean-Michel Hiver wrote:
Which makes me think, what is the real use of AEL. I have taken a look
at it, and it makes asterisk's config file almost as unreadable as SER.
What exactly does AEL do that a well written AGI / FastAGI app doesn't?
I would think (but I'm surely wrong) that it would be
On 2006-08-17 23:12:29 -0700, Barzilai [EMAIL PROTECTED] said:
I still haven't figured out what is the best practices or
Asterisk-way to do traditional switching between channels in Asterisk.
I come from traditional computer telephony where there are buses such
as MVIP, with streams and
Steve Murphy wrote:
...
[a lot of well-written arguments]
...
And, pardon the shameless plug here, but for all you fence sitters, I
invite you to try AEL in/for your dialplans, and give me feedback! If
the majority of those who use it feel it's useless, I'll drop it and do
other useful things
Barzilai wrote:
I still haven't figured out what is the best practices or
Asterisk-way to do traditional switching between channels in Asterisk.
I come from traditional computer telephony where there are buses such
as MVIP, with streams and timeslots.
Asterisk, being born as a PBX solves most
On 2006-08-17 18:54:00 -0700, Ferguson, Michael [EMAIL PROTECTED] said:
This is a multi-part message in MIME format.
Hello All,
=20
I am quite frustrated at my lack of knowledge here and so I seek
pointers from you, the wise ones.
Repeated scouring of my .conf files is unfruitfull.
=20
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barzilai
Sent: Friday, August 18, 2006 2:55 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: what is the real use of AEL?
Steve Murphy
17 aug 2006 kl. 18.38 skrev Ferguson, Michael:
G'Day List;
I hoping for some direction here:
The following message is scrolling without end on my asterisk box,
continuously: (NOTE: date and time changes accordingly and IP
addresses are not real)
Aug 17 11:49:53 NOTICE[1034]:
17 aug 2006 kl. 21.33 skrev Douglas Garstang:
Does realtime support include = yet?
Do you mean the realtime switch or static load of extensions.conf
from realtime?
If you mean the realtime switch, how would you suggest include=
should work?
Curious.
/Olle
17 aug 2006 kl. 22.03 skrev kjcsb:
How to I access the URI from an Invite:
INVITE sip:[EMAIL PROTECTED]
I want to set a variable to equal 5556678. The variable ${SIPURI}
returns the From URI.
The extension processed in the dialplan is the userinfo part of the
URI - the part before
the
18 aug 2006 kl. 07.48 skrev kjcsb:
I have read the wiki about the SIP_HEADER function (http://www.voip-
info.org/wiki/index.php?page=Asterisk+func+sip_header). Where can I
get a list of the names that are available to be used with the
function e.g. TO is one name as in ${SIP_HEADER(TO)}.
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
Given that all you have on the client side is:
exten = _X.,1,AGI(agi://server.gumby.com)
... how do you send commands?
The other thing you can do, if you want the same FastAGI server to do
different things from
Hi.
I have the following two extensions:
exten = _71405XXX,1,Dial(Zap/g1/${EXTEN:5}|20,tr)
exten = _71.,1,Dial(Zap/g2/${EXTEN:2}|20,tr)
I have an external application that generates dialstrings, it generates
the 71 prefix so that the call can go through the T1 cards. As you can
see the 71
In article [EMAIL PROTECTED],
Jan du Toit [EMAIL PROTECTED] wrote:
I have the following two extensions:
exten = _71405XXX,1,Dial(Zap/g1/${EXTEN:5}|20,tr)
exten = _71.,1,Dial(Zap/g2/${EXTEN:2}|20,tr)
I have an external application that generates dialstrings, it generates
the 71 prefix
Hi Leo,Thank you for your quick response. In Internet, I came to know that1) In India, we have to give dtmf and ring for cidsignallling and cidstart respectively. 2) Default Asterisk setup doesn't recognise callerid in India. To recognize callerid in India, we have to do or change some
Olle,
Thanks
,preciate it.
Best Wishes
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, August 18, 2006 3:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Registration Error
Is there a maximum length of allowable CID information when using the AGI
command SET CALLERID ?
We're having some issues with our outbound minutes provider, and I want to
make sure it's not just a limitation on our side that's causing the problem.
We set caller ID info based on the user's
In my case I'm able to see the callerid digits being
read by the chan_zap module. However, the number does
not get through to the ${CALLERID} variable. please
see debug trace below.
zaptel.conf:
fxsls=1
fxsls=2
loadzone= us
defaultzone = us
in zapata.conf, i have:
signalling=fxsls
Hi all,I'm developing an app with dialout .call files, and when one of the legs of my call is busy i get this msg: -- Called g1/2132 -- Channel 0/1, span 1 got hangup -- Channel 0/1, span 1 received AOC-E charging 149502040 units
-- Zap/1-1 is busyIt seems to me OK, but i'm wondering the meaning
I also want to add that I saw a great improvement from versions 1.0.x to
versions 1.2.x. Let's see what 1.4 will bring, but I hope a 2.0 version
with a complete rearchitecturing could finally make Asterisk the Apache
of telephony as I read somewhere. (Or wait for the OpenPBX guys to
awake from
Hello Han!
I'm from Rio de Janeiro and I'm using Tmais (www.tmais.com.br).
I'm very happy with the service and they accept credit card payment.
Regards;
Leonardo Kamache
On 8/17/06, Han van Hulst [EMAIL PROTECTED] wrote:
Who can help me out i am looking for a Brazilaan telephone number
On Friday 18 August 2006 07:57, Marco Mouta wrote:
-- Channel 0/1, span 1 received AOC-E charging 149502040 units
-- Zap/1-1 is busy
It seems to me OK, but i'm wondering the meaning of received AOC-E charging
149502040 units ???
Asterisk ignores AOC (advice of charge) messages, so
Hi Marco, as good?
Well, you are use libpri-1.2.3? Believe that this is a bug of this version. Look at link´s below, contains patchs for this problem.
I wait to have helped.
Best Regards
Josué
http://bugs.digium.com/file_download.php?file_id=7499type=bug
Sergio:
Hola tanto tiempo!!
cuando te conectás? asi charlamos un poco...
Un tema que quedó pendiente es que íbamos a hacer algo con una cooperativa
amiga tuya..
¿te acordás?
chei, ando necesitando un gw gsm para conectar a la central.
Creo que vos tenés el 2N VoiceBlue Lite. ¿Lo tenés en
Disculpen, mandé un mensaje particular a la lista.
Salu2
Leonardo
Leonardo F. Bauchwitz
__
Preguntá. Respondé. Descubrí.
Todo lo que querías saber, y lo que ni imaginabas,
está en Yahoo! Respuestas (Beta).
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Hi Marco, as good?
Well, you are use libpri-1.2.3?
Believe that this is a bug of this version. Look at link´s below, contains
patchs for this problem.
I wait to have helped.
Best Regards
Josué
Barzilai wrote:
Steve Murphy wrote:
...
[a lot of well-written arguments]
...
And, pardon the shameless plug here, but for all you fence sitters, I
invite you to try AEL in/for your dialplans, and give me feedback! If
the majority of those who use it feel it's useless, I'll drop it and do
other
I'd appreciate some feedback on the behaviour of some tests relating to
presence SUBSCRIBE/NOTIFY. In the tests no NAT or proxies are involved.
We have a client using a SIP stack accepting requests on one port (eg 5060)
but handling responses on a 'temporary' port. In other words it sends a
18 aug 2006 kl. 16.04 skrev Shaun Bailey:
I'd appreciate some feedback on the behaviour of some tests
relating to
presence SUBSCRIBE/NOTIFY. In the tests no NAT or proxies are
involved.
We have a client using a SIP stack accepting requests on one port
(eg 5060)
but handling responses
Hi Nathan -
The problem occurs during transfer and hold retrieval, answering the
call is fine, the call is put on hold then either a transfer is
attempted or the call is retrieved from hold. When this is attempted
the remote party (i.e. the caller in the case of a hold retrieval)
cannot hear
Hello everyone,
I am still on my quest to build a g729 compatible (yet license free)
Asterisk system.
Here is the latest... It seems that when Asterisk needs to indicate
ringing or busy to a SIP channel that has already been answered (like
with an IVR) it plays back audio using the
What are trying to do is bridge 2 existing channels, the only thing
that can do it right now is meetme. There has been a lot of talk on
this list about this, but it still doesn't exist (at least in stable)
in Asterisk.
On 8/18/06, Barzilai [EMAIL PROTECTED] wrote:
I still haven't figured out
Hi
i have running a call center and I have
5 agents , and I m using vcdial. I have 512kb 1:1 bandwidth and also I m using codec
g729 license.but then also I m facing a voice brakeage.
Can I monitor(barge) an agent and client conversation ?
If u have the solution pls let me know
Is your server located locally ?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of support
Sent: Friday, August 18, 2006
11:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] call
barge
Hi
i have running a call center and I
have 5 agents ,
Hi,
is there any way to monitor a video call coiming from IAX2/SIP ..or
video voice mail with asterisk?
thanks
atik
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I just bought a Te207P, and I was wondering if there is anything special
that I have to do in Asterisk's zapata.conf or the zaptel.conf to enable
the echo canceller, or if it is automatically enabled.
Thanks in advance!
Steve
___
--Bandwidth and
In article [EMAIL PROTECTED],
Kristian Kielhofner [EMAIL PROTECTED] wrote:
Hello everyone,
Hi Kris
I am still on my quest to build a g729 compatible (yet license free)
Asterisk system.
Here is the latest... It seems that when Asterisk needs to indicate
ringing or busy to a
Steven Ringwald wrote:
I just bought a Te207P, and I was wondering if there is anything special
that I have to do in Asterisk's zapata.conf or the zaptel.conf to enable
the echo canceller, or if it is automatically enabled.
Make sure echocancel=yes is in zapata.conf. Also, you can look in
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort
of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
exten = _1877NXX, Dial,
exten =
I am running Asterisk 1.2.9.1 in a call center with 26
agents placing outbound calls using SIP soft phones going out a Diginum 4 port
T1 card (All 4 spans have PRI t1s).
All of the calls run through
[macro-record-call]
exten =
-Original Message-
From: Moises Silva [mailto:[EMAIL PROTECTED]
Sent: Friday, August 18, 2006 5:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Recent additions to the Digium
Asteriskdevelopment team
I also want to add that I saw
On 8/18/06, David Cook [EMAIL PROTECTED] wrote:
Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort
of like the SPA's can?
Tollfree numbers for example. I can have a line for each combination:
exten = _1800NXX, Dial,
exten = _1866NXX, Dial,
Douglas Garstang wrote:
Or wait for free-switch (http://www.freeswitch.org) :)
I don't see any docs on the web site. That's a great start.
Unless I missed them somewhere.
Huh? Tony writes lots and lots and heaps of documentation.
http://www.freeswitch.org/docs/
Lee.
Hi all,I'm working with dialout call files and i've noticed that with MaxRetries: 1 ,many times the call is already established successfully and asterisk dials a second call.I mean Asterisk dials party A then party B then Bridges the call and after a while starts trying to do the same...
Can someone tell me what this is about? Asterisk seems to be 'losing' peers.
Usually when a peer isn't known (such as when you first start Asterisk),
Asterisk will do a database lookup and find the peer, and then seed them.
I tried to dial 3254101, and I get the error below. I ran an ngrep and
Sounds like a poweredge server and the 2850 we
have...hyperthreading has caused all kinds of crazy stuff...and turning it off
just in grub.conf doesn't solve it until it is turned off in the
bios.
- Original Message -
From:
Adam
Kavan
To:
Douglas Garstang wrote:
Can someone tell me what this is about? Asterisk seems to be 'losing' peers.
Usually when a peer isn't known (such as when you first start Asterisk),
Asterisk will do a database lookup and find the peer, and then seed them.
I tried to dial 3254101, and I get the error
Marco Mouta wrote:
Hi all,
I'm working with dialout call files and i've noticed that with
MaxRetries: 1 ,many times the call is already established successfully
and asterisk dials a second call.
I was having the same issue until I schedule the dialout a few minutes
after the .call file
Here's an idea...
Rather than writing your own multi-thread socket server for use with FastAGI,
has anyone tried to use an Apache web server instead? After all, it does all
that for you. I just gave it a shot, but Asterisk tries to send all the agi
params to the web server, which it doesn't
On 8/18/06, Barzilai [EMAIL PROTECTED] wrote:
Jean-Michel Hiver wrote:
Which makes me think, what is the real use of AEL. I have taken a look
at it, and it makes asterisk's config file almost as unreadable as SER.
What exactly does AEL do that a well written AGI / FastAGI app doesn't?
I
It is an valid option, but you have to build a HTTP header in your
request to your web server, which CGI programs or Java servlets on web
server could interpret your request from Asterisk.
Tielin
[EMAIL PROTECTED] 08/18/06 11:28 AM
Here's an idea...
Rather than writing your own multi-thread
Tony Mountifield wrote:
OK, create a script called samples.pl containing the following:
#!/usr/bin/perl
$times = shift || 1;
undef $/;
$data = ;
$data =~ /{\s*(.*?)\s*}/s;
@samples = split /[^0-9-]+/,$1;
print pack('v*',@samples) x $times;
#--end--
This will read busy.h or ringtone.h and
Jeremy McNamara wrote:
Steven Ringwald wrote:
I just bought a Te207P, and I was wondering if there is anything
special that I have to do in Asterisk's zapata.conf or the
zaptel.conf to enable the echo canceller, or if it is automatically
enabled.
Make sure echocancel=yes is in zapata.conf.
I have read the wiki about the SIP_HEADER function (http://www.voip-
info.org/wiki/index.php?page=Asterisk+func+sip_header). Where can I get
a list of the names that are available to be used with the function e.g.
TO is one name as in ${SIP_HEADER(TO)}. What are the others?
I would guess
Asterisk is what you make of it. If you don't want certain
applications to run on a certain instance/machine then you should
noload them in modules.conf.
Barzilai still has a point. Noloading various applications doesn't
address the underlying architectural issues. The fact of the matter
I tried this...
[test-in]
switch = Realtime/[EMAIL PROTECTED]
switch = Realtime/[EMAIL PROTECTED]
It didn't work. The select that Asterisk sent to the database was:
SELECT * FROM extensions_table WHERE exten = '1001' AND context = 'test1' AND
priority = '1'
So, it obviously ignores the second
I'm currently trying recent asterisk trunk with original chan_skinny
(chan_sccp isn't working with trunk) and phone ci$co 7920,
dialout direction from phone is OK,
but can't receive calls (dial command: Dial(Skinny/[EMAIL PROTECTED]) or
Dial(Skinny/${EXTEN})
phone ci$co 7920 (wifi),
- Pavel Jezek [EMAIL PROTECTED] wrote:
I'm currently trying recent asterisk trunk with original chan_skinny
(chan_sccp isn't working with trunk) and phone ci$co 7920,
dialout direction from phone is OK,
but can't receive calls (dial command: Dial(Skinny/[EMAIL PROTECTED]) or
I like the gumball analogy.
I'm more awake today. I wasn't complaining last night, I was just
stating my criticism in an emphatic way :-)
As you say, Mark Co just started writing a little soft PBX system to
scratch their own itches at the time.
Mark was so kind as to share it with the rest.
Can anyone tell me the proper way to NOT generate a CDR record for a call using Asterisk 1.2? I heard about the C option and tried it, but I still see the call details in Master.csv. It would be nice if there was a way to NOT log an incoming call as well. Is that possible?
Thanks for any help in
exten = blah,n,NoCdr()
On Fri, 2006-08-18 at 16:55 -0400, Nate Kapi wrote:
Can anyone tell me the proper way to NOT generate a CDR record for a
call using Asterisk 1.2? I heard about the C option and tried it, but
I still see the call details in Master.csv. It would be nice if there
was a way
On 8/18/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Here's an idea...
Rather than writing your own multi-thread socket server for use with FastAGI,
has anyone tried to use an Apache web server instead? After all, it does all
that for you. I just gave it a shot, but Asterisk tries to send
IIRC, You can use REGEXes in your extension matchingDon't have a handy
link, but if I find it, I'll forward
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
William Moore
Sent: Friday, August 18, 2006 1:04 PM
To: Asterisk Users Mailing List -
I have an Iaxy that I am using to access our overhead paging system.
It is ext 5480 and required a 1 (office), 2 (Shop), or 3 (all) DTMF tone
after it answers.
If I dial 5480, I hear a tone to let me know that it is ready for the
digit.
I made an extension 5481 that using a macro and sendDTMF to
At 01:22 PM 8/18/2006, you wrote:
The other side of the coin is that there's a risk of becoming a
Frankenstein system with lots of ugly patches and configuration
syntaxes with small variations and basically lines upon lines of
code where everybody adds something to scratch their own itch in
Thanks Anders. Couldn't get erlang to compile. :(
-Original Message-
From: Anders Nygren [mailto:[EMAIL PROTECTED]
Sent: Friday, August 18, 2006 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Apache for FastAGI
On 8/18/06,
Hi,
I just saw there is a branch in the SVN that support SLA. The latest trunk also seems to have some kind of SLA support. Is there any doc about how to setup a shared
line or any docs concerning this feature?
Thx
David
___
I don't know if I responded to the original poster before but if you are looking for a python fastAGI server, there already is one, its called starpy.Anders, since you know Erlang, do you know of any media processig libraries in Erlang, do the ericsson softswitches do the media processing
Hi,
Is there any option in asterisk to load all the prompt files into
memory on startup, so that it doesn;t have to hit the disk to read
prompts for any call.
Or any plugin / suggestion to avoid hitting the disk for prompt files?
Thanks in advance.
Nitin
Try This
exten = 5481,1,NoOp(${TIMESTAMP} paging Group 1 Office Page)
exten = 5481,2,DIAL(IAX2/5480/w1||)
SNIP
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Hi,
Take a look at ramfs (http://plume.bxlug.be/articles/7). All you
need then is to create a link (ln -s) in /var/lib/asterisk/sound to then
ramdrive you created using ramfs.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Nitin
On Fri, 2006-08-18 at 19:17 -0400, David Gagnon wrote:
I just saw there is a branch in the SVN that support SLA. The latest
trunk also seems to have some kind of SLA support. Is there any doc
about how to setup a shared line or any docs concerning this feature?
It is pretty straight forward to
Any further experience with the 3102? I'm looking for a solution to connect 2 CO lines and a set of 2-line phones to my asterisk server (along with a bunch of SIP phones). Would 2 of these work well for that?
Hopefully no echo problems! That would kill this project? I'm still searching for
Heya all,
what is the acceptable latency for VoIP calling? 200ms? 300ms?
Best Regards,
Freddy Setiawan
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Barry D. Hassler wrote:
Any further experience with the 3102? I'm looking for a solution to
connect 2 CO lines and a set of 2-line phones to my asterisk server
(along with a bunch of SIP phones). Would 2 of these work well for that?
Hopefully no echo problems! That would kill this project?
Such an objective question. Everyone, including different users will
have a different answer.
Is this within an enterprise? at home? with a paid service? what
codec? pure IP or TDM mix?
I would say anything over 200 is bad, now how close you get to that.
We try to engineer our on net
We got mix IP and TDM. Currently we are facing 170-200ms latency. we are
using g711 for onnet, and g729 for offnet.
Quite difficult to get IVSP that located in south east Asia, since most
of the IVSP are located somewhere in US, EURO.
Still looking for IVSP that has pop in Singapore or
On Sat, 2006-08-19 at 00:12 -0500, Rich Adamson wrote:
Barry D. Hassler wrote:
Any further experience with the 3102? I'm looking for a solution to
connect 2 CO lines and a set of 2-line phones to my asterisk server
(along with a bunch of SIP phones). Would 2 of these work well for that?
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