[EMAIL PROTECTED] ha scritto:
On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote:
[EMAIL PROTECTED] ha scritto:
context = from-sccp-intenal
I guess intenal is not the righe context :-)
Sergio
The from-sccp-internal is almost an exact copy of my
On 4/8/06, Joe [EMAIL PROTECTED] wrote:
Remove the SIP /400 entry from the Asterisk DB.
Database del At asterisk prompt.
Or look at the wiki for info on how to remove it.
Or make sure the SIP/500 uses a different IP address than the old SIP/400.
Joe
thanks for your reply i've
On 4/10/06, William Harrison [EMAIL PROTECTED] wrote:
How is the 9133i configured, through the .cfg file, the WebUI, or the
Phone's own interface? The PhoneUI WebUI take precedence over the
.cfg file.
You can look at the WebUI and see what the current settings are, and
clear them out if
I want to playback sound file loaded in memory not from a file...is
this possible?
_
FREE pop-up blocking with the new MSN Toolbar - get it now!
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/
12 apr 2006 kl. 18.38 skrev Ronald Lewis:
I was alerted the other day by of all people, my mom, that she
wasn't hearing a ring when she dialed my number. Puzzled, I tried
calling myself. The call connects, but there's dead silence until
voicemail picks up. Calling internally, extensions
[EMAIL PROTECTED] wrote:
Hi All,
We are making a VOIP application for Mobiles (PDA's) and we are using
Asterisk
for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP
router and routes everything to Asterisk. We also have rtpproxy for SER. Our
packet delivery
On Thu, 2006-04-13 at 09:00 +0200, nik600 wrote:
On 4/10/06, William Harrison [EMAIL PROTECTED] wrote:
How is the 9133i configured, through the .cfg file, the WebUI, or the
Phone's own interface? The PhoneUI WebUI take precedence over the
.cfg file.
You can look at the WebUI and see
For the moment, if you need FAX tone detection, you will need to use
'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp
module; this will not disable the echo canceler, just stop using it for
tone detection.
Any idea if/when this will be addressed? We had been going
As a clarification for further posts, its wise to delete both the asterisk modules and header directory when having problems upgrading from 1.0 to 1.2 and depracated modules are in the way. As Rob T. pointed out the best way to do this is:
# mv /usr/lib/asterisk/modules
[EMAIL PROTECTED] wrote:
On Wed, 12 Apr 2006, Leo Ann Boon wrote:
I'm not sure tmpfs is the right solution for the OP's problem - disk
access slowing down the system. My understanding of tmpfs is that it
will swap pages in and out to/from disk. Wouldn't that be as bad as
directly writing to
Hi Joe,
In your mail you wrote that
I've heard a few stories that reported partial success with an Eicon
Diva Server card, but always with the caveat that it doesn't work quite
right or something along those lines.
I can ensure you that this is not the case. We are implementing a Diva
Server
Hi,so how do I contribute the translation? (this would be one of my first contrib to an open source project).For the translation, I used Translation2.php from the pear repository and put the translated text in an xml file.
Please informe me on how to contribute.benqOn 4/3/06, Dan Austin [EMAIL
The thing I absolutely need is. To play a background
music in call.
If I have the opportunity to stop it via entering a
dtmf combination is would be very very nice also.
Does anybody know some application do this.
NZR
__
Do You Yahoo!?
Tired of
I'm seeing Diva Server V-BRI running close to $1K/card. There are other
Diva cards running around $700. A little pricy but not impossible to
do. I remember back in the 90's I had ISDN into my home for internet
access. The netgear router I used cost me about $350 back then, and it
worked
is it possibile to set up an external smtp server for the relay to the
users of the mails?
thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Typical user error, one user forwards his calls to another using CFwdAll on
Cisco 7940, but the user receiving the call has done the reverse.
-- Called 117
-- Got SIP response 302 Moved Temporarily back from 10.139.2.15
-- Now forwarding Local/[EMAIL PROTECTED],2 to 'Local/[EMAIL
Hi !
Anybody know if 1800 free termination services from trxtel are in troubles?
I can´t reach it, and don´t know why.
Thanks a lot
gus
At 06:49 a.m. 26/03/2006, you wrote:
Hi there,
Thanks for the tip ! I am happily using this service now.
One question though : I cannot get DTMF to work.
Hi,
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does anyone have some idea what could be wrong?
No verbose data...
Tks,
Daniel Korndorfer
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
What version of Asterisk?
On Thu, 2006-04-13 at 12:38, Daniel Korndorfer wrote:
Hi,
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does anyone have some idea what could be wrong?
No verbose data...
Tks,
Daniel Korndorfer
Hi,I've been debuging the call disconnection problem in our
architecture:PSTN---E1---OldPBX---E1---AsteriskThis is our
problem:-SIP user agent A calls a pstn phone B.-B hangs up the
call.-SIP user agent A starts listenning busytones... But the call still
on. (and being payed).- Call only ends
Just noticed that I occasionally get these messages:-
Apr 12 09:27:03 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq
ran 281 scheduled tasks all at once
Apr 13 09:13:18 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq
ran 1987 scheduled tasks all at once
Apr 13 12:47:56 WARNING[11390]
I had this problem with 1.2.5, 1.2.6 and now with 1.2.7...
On 4/13/06, Gareth Blades [EMAIL PROTECTED] wrote:
What version of Asterisk?
On Thu, 2006-04-13 at 12:38, Daniel Korndorfer wrote:
Hi,
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does
Hi,I need to have the information about the current IP address of the user. I want to know IP address from which user is registered to Asterisk server. Is it possible with Asterisk to log this information to the database or file? Does anyone can give me some info about this issue? Thanks in
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
On Wed, Apr 12, 2006 at 11:16:10PM -1000, Mark Coccimiglio said:
I'm seeing Diva Server V-BRI running close to $1K/card. There are other
Diva cards running around $700. A little pricy but not impossible to
do. I remember back in the 90's I had ISDN into my home for internet
access. The
On Wednesday 12 April 2006 18:40, Jim Rice wrote:
Had I have documented the process and included config files and log
files and tcpdump traces, wouldn't I have received the TMI lecture
instead?
Depends on how verbose you were. The [macid].cfg file is very, very short
though (1-3 lines IIRC)
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not to mention extremely cool.
dbc.
Hi Joe,
In your mail you wrote that
I've heard a few stories that reported partial success with an Eicon
Diva Server card, but always with the caveat that it doesn't work quite
right or something along those lines.
I can ensure you that this is not the case. We are implementing a Diva
On Thursday 13 April 2006 09:02, David Cook wrote:
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not to mention extremely
If the phones are registering twice, and you are 100% sure they are only
configured via the WebUI, then you must have the settings in two places.
There are SIP configs for Global SIP, and also for every available line.
You may have one set of settings in Global SIP, and a different one for
one of
Andrew,
This is written to the asterisk database.
Use database show from the CLI.
That will show the IP addresses of the
people that are registered in the /SIP/Registry/(username).
William Piper
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew Nowrot
Hi,
I'm writing a Java client/server application that talks to the Asterisk
manager interface via the asterisk-java stuff. The idea being it will
give you an app to run on your desktop that monitors your phone
essentially. Once I've got something vaguely working it will be released
under the
Daniel Korndorfer wrote:
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does anyone have some idea what could be wrong?
No verbose data...
Occasionally I am seeing music on hold stop playing for parked calls, so
this isn't unique for queues. Maybe it is as
Hi I'm a little confused here... trying to setup a parking lot...
lot is setup but how do I send calls to the parkinglot? If I
allow #700 transfer, it seems I can only transfer on inbound calls...
if I use a T in my dialplan I can only transfer on outbound calls...
additionally pressing #
Currently, compiling asterisk on an Itanium fails with the GSM codec.
All I could find on Google was a hack to basically remove GSM from the
build which is not an option for some. This patch will allow it to
compile and seems to work perfectly.
Thanks,
Steve Totaro
On 4/13/06, Alex Brett [EMAIL PROTECTED] wrote:
Hi,
I'm writing a Java client/server application that talks to the Asterisk
manager interface via the asterisk-java stuff. The idea being it will
give you an app to run on your desktop that monitors your phone
essentially. Once I've got
Hi,Thanks for so fast replyOk I know about this but actually I am thinking about logging the IP address of a user in realtime. Each time the user changes his location and register Asterisk will log the time and IP address. Is it possible?
Best wishes
___
Just an update found a few bug
tickets regarding it and a change to page.trunk.php which allows the w.
Apparently it will be fixed by version 2.1
Thanks
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Wednesday, April 12, 2006
9:15 PM
hi,
can you someone explain this bug? (or point me to number from
bugs.digium.com)
2006-03-28 19:07 + [r15699] Olle Johansson [EMAIL PROTECTED]
* channels/chan_sip.c: Fix breakage of NAT support for peers with
qualify=yes. Thanks Damin for access to your system, sorry folks.
thanks
Don Pobanz wrote:
Daniel Korndorfer wrote:
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does anyone have some idea what could be wrong?
No verbose data...
Occasionally I am seeing music on hold stop playing for parked calls, so
this isn't unique for
Currently, compiling asterisk on an Itanium fails with the GSM codec.
All I could find on Google was a hack to basically remove GSM from the
build which is not an option for some. This patch will allow it to
compile and seems to work perfectly.
Thanks,
Steve Totaro
Hi all,
I'm using Asterisk 1.2.5 and , for some reason, when I install it, the
module app_meetme.so didn't install. Is there some way to download
that module, and add it to asterisk without re-install it?
Thanks in advance
Sebastian
___
--Bandwidth
I seem to be having the same problems. Is anyone from trxtel reading
this? I guess you get what you pay for :)
- Waldo
On Apr 13, 2006, at 6:36 AM, Gustavo Hernandez wrote:
Hi !
Anybody know if 1800 free termination services from trxtel are in
troubles?
I can´t reach it, and don´t know
Hi,
I have an asterisk 1.2.1 on a debian box with a tdm400p card and a
monoBRI card.
I tryed to use fxotune: turned off asterisk leaving modules active as
seen from lsmod:
zaptel224132 1 wctdm
crc_ccitt 2432 1 zaptel
I launched fxotune:
*pbxtest:/etc/asterisk#
At 01:19 PM 04/11/2006, you wrote:
The last point also brings up a question. Does anyone know how
gracefully Asterisk handles attempting to write leg files to a full disk?
For some number of days my * box was running with the disk set to
read only and I only discovered it when I noticed some
Un-comment ztdummy and build re-zaptel and then re-build asterisk
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Sebastian Milioto [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 13, 2006 10:31 AM
To: asterisk-users@lists.digium.com
Subject:
Are you guys using native music on hold, or MP3 music on hold?
On Thu, 13 Apr 2006 10:09:26 -0400, Alex Brett [EMAIL PROTECTED]
wrote:
Don Pobanz wrote:
Daniel Korndorfer wrote:
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does anyone have some idea
Mmmm.. sorry... I'm rookie. Coudl you be a litlle more specific please?
Where I have to uncomment ztdummy?
On 4/13/06, Steve Totaro [EMAIL PROTECTED] wrote:
Un-comment ztdummy and build re-zaptel and then re-build asterisk
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
Google is your friend.
http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLD,GGLD:2
004-48,GGLD:enq=uncomment+ztdummy
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Sebastian Milioto [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 13, 2006
marek cervenka wrote:
can you someone explain this bug? (or point me to number from
bugs.digium.com)
2006-03-28 19:07 + [r15699] Olle Johansson [EMAIL PROTECTED]
* channels/chan_sip.c: Fix breakage of NAT support for peers with
qualify=yes. Thanks Damin for access to your system,
On 4/13/06, David Cook [EMAIL PROTECTED] wrote:
My cell vm goes to asterisk, not the carrier. Apparently MWI is turned
on/off with specially formatted SMS messages. Anyone know how to do this
on a Treo 600? Having the phone light from Asterisk would be HUGE ...
not to mention extremely cool.
George Pajari wrote:
For the moment, if you need FAX tone detection, you will need to use
'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp
module; this will not disable the echo canceler, just stop using it for
tone detection.
Any idea if/when this will be
Akpome Akpoguma wrote:
I want to playback sound file loaded in memory not from a
file...is this possible?
Akpome,
If the sound file is being played more than once, there is a good chance
that this is already happening. At one point, our production system had
100 calls in queue.
Hi,
When making a call from an Asterisk box over a PRI connection, I am
able to set the Caller ID phone number to what ever I want. This works find.
How to I make the called party callerid display Confidential or
unknown as we sometimes see ?
Andre
I've been working on this off and on for AGES. There are some SMS portal
sites that claim to be able to do this as well, but I have not managed to
find one.
I had found a company called bahamasystems which has an asterisk interface but
it's a service and it's expensive.
Another poster
Justin Tunney wrote:
Are you guys using native music on hold, or MP3 music on hold?
I believe I am using MP3. My musiconhold.conf file looks like this
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
--
Don Pobanz
___
--Bandwidth and
Hi,
I am trying to register SIP clients which are behind NAT on different
network. In order to achieve this goal I think I need STUN Server . I
downloaded STUN Server from
http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz
But I don't know how to install/configure it.
And
Andre Courchesne - Consultant wrote:
Hi,
When making a call from an Asterisk box over a PRI connection, I am
able to set the Caller ID phone number to what ever I want. This works
find.
How to I make the called party callerid display Confidential or
unknown as we sometimes see ?
In
It is in fact required for some implementations of callerid name. It
comes on a facility message that arrives after the call is setup. It
also can come in a display IE in the call setup. It really depends on
which way they are sending it.
Matthew Fredrickson
On Apr 11, 2006, at 12:49 PM,
Hi,
Have you try to set hidecallerid=yes in zapata.conf?
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre
Courchesne - Consultant
Sent: Friday, April 14, 2006 12:02 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Display
Prepend *67 if your carrier allows it
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Andre Courchesne - Consultant [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 13, 2006 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Display
Maybe hidecallerid=yes in Zapata.conf
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 13, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Rich Adamson wrote:
In the US, you can't.
Yes, you can. You just set the 'presentation' bits to show that the
number is not known or is restricted. However, you can't control the
actual words that show up on the recipient's device instead of the CNAM...
If you mean to have a private caller ID, I would think that the
phone company would need to update you record in the database.
On 4/13/06, Andre Courchesne - Consultant [EMAIL PROTECTED] wrote:
Hi,
When making a call from an Asterisk box over a PRI connection, I am
able to set the Caller
Don Pobanz wrote:
I believe I am using MP3. My musiconhold.conf file looks like this
[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3
Daniel and Don,
Try switching to native MOH. You'll eliminate the decoding of the MP3s
and the host of problems that come along with using
Just point your ATA to stun.fwdnet.net; it is a free service by free world
dialup. Sure beats creating your own stun server.
If you do need to create your own stun server, I suggest
http://www.voip-info.org/wiki/view/Vovida.org+STUN+server
-Original Message-
From: [EMAIL PROTECTED]
I'm running asterisk 1.2.6 with Sangoma A200 4 port FXO card.
Outbound calls are working fine. However, when I have an inbound
call.. asterisk will segfault.. and then start again ... then it will
take 1 call fine
I'm running asterisk with a -U and -G of asterisk. Any thoughts?
Here is what you would do for a sip call. I'm sure it isn't much different
for a PRI call.
[cid-block]
exten = _*67.,1,Ringing
exten = _*67.,2,goto(dial-cid-block,${EXTEN:3},1)
[dial-cid-block]
exten = _XXX,1,Macro(cid-block,1${CALLERIDNUM:-10:3}${EXTEN})
exten =
Anyone have any ideas why DTMF would not work on only one number? Looking
through the logs, anytime a button is pressed, this is what shows up:
2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1
2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on
We get these messages too, but they don't seem to cause any problems.
Are you connecting 2 * (with different versions) via IAX2? Are these
messages only appear on the lower version one? I asked a similar
question on the list, and the suggestion was to upgrade them to the
same version.
Hope this
Kevin P. Fleming wrote:
Rich Adamson wrote:
In the US, you can't.
Yes, you can. You just set the 'presentation' bits to show that the
number is not known or is restricted. However, you can't control the
actual words that show up on the recipient's device instead of the CNAM...
Ops, I read
I am wondering if anyone has sample XML config for the Sipura 2100 ATA.
We have been autoprovisioning our 2002s with success and the 2100's take
the same XML that we have come up with, but I am not sure of the syntax
for specific things that I need these boxes to do, such as turning T.38 on.
I'm trying to set up a system so that I can record a conversation over
SIP. Monitor and the like don't work so well for me, because I need to
pipe the conversation to other programs in realtime, rather than record
to a file, so I've been trying to use EAGI instead. (if anyone has any
other
I am trying to link an asterisk box up to a SIP server on the same
subnet. The SIP server does not have a password (and is locked down by
IP number 'allow'). How do I specify this on the register line?
Based on the documentation, the line looks like this:
register = user[:secret[:[EMAIL
I just upgraded to Asterisk 1.2.7 from 1.2.5.
Page() is behaving differently.
I'm getting an error - Incomplete destination '' supplied.
-- Executing Page(SIP/2944093-5999, SIP/3254107SIP/3254105|) in new
stack
Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destination
Wai,
Please explain how the in and out channels are mixed first before
they are written to the disk using monitor with no mixing onto the
scsi drive. I'd love to implement this on our system to cut in half
the I/O associated with Monitor().
Also, what bug does MixMonitor() have? It is
Hello,
I use .call files in /var/spool/asterisk/outgoing to initiate calls
automatically. And I'd like to setup the language used for the call in
this file but I haven't found any way of doing this. I tried something
like Set: language=fr, Set: ${LANGUAGE}=fr, ... but nothing worked.
Is
Hi
I need help in setting up the CAS circuit using Asterisk and
Digium Dual Port
T1 card.
I tried it but without any luck.
Any one have experienced the problem with Feature Group D on
Asterisk 1.2.6
Thanks
Ali
___
--Bandwidth
Douglas Garstang wrote:
I just upgraded to Asterisk 1.2.7 from 1.2.5.
Page() is behaving differently.
I'm getting an error - Incomplete destination '' supplied.
This was a bug introduced in 1.2.7. I have just fixed it in Subversion,
so you can update to the latest branch-1.2 code from there if
Check your 'GENERAL' section and your 'LINE' sections.. you probably
only changed in one place.
On 4/13/06, William Harrison [EMAIL PROTECTED] wrote:
If the phones are registering twice, and you are 100% sure they are only
configured via the WebUI, then you must have the settings in two
Details:Asterisk 1.0.9 Zaptel 1.0 Dell P3 1ghz with X100P Clone Location: IndiaThis is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel doesn't know to hang up the line so after a couple of hours when the telecom cuts the line, everythign stops
The Asterisk Development Team has released version 1.2.7.1 of Asterisk.
This release contains only two fixes, one of which is that the Page()
application was entirely broken in version 1.2.7. If you have already
upgraded to 1.2.7 and you do not use the Page() application in your
dialplan, there is
Hi,
I've searched almost everywhere but have not come across a solution so I was
hoping one of your fine folks can help me out.
The problem is that a carrier is passing me early media on calls that
sometimes have problems connecting. For example, calls to India mobile might
play an early media
I have a TDM2400P with hardware echo cancel. We seem to have static on
some calls but not others and the receive audio appears 'choppy'.
Transmit side works fine and does not have any audio problems. I had to
turn up the RX gain to 18 or the receive audio volume is too low.
Can anyone shed
Tim Jackson wrote:
I have a TDM2400P with hardware echo cancel. We seem to have static on
some calls but not others and the receive audio appears 'choppy'.
Transmit side works fine and does not have any audio problems. I had to
turn up the RX gain to 18 or the receive audio volume is too
On 4/13/06, Tim Jackson [EMAIL PROTECTED] wrote:
I have a TDM2400P with hardware echo cancel. We seem to have static on
some calls but not others and the receive audio appears 'choppy'.
Transmit side works fine and does not have any audio problems. I had to
turn up the RX gain to 18 or the
I did not install soxmix in my linux box. If you having issues with
mixmonitor, you can put both legs of the call into a conference and
record the conference
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: Thursday, April 13, 2006 1:20 PM
Early audio is played, as long as you do not have a r in your Dial
statement.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mohammed Salim
Sent: April 13, 2006 2:17 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
ESWG Consensus 12-month Report on Nomadic VoIP Technical and Operating
Impediments to 9-1-1/E9-1-1 Service Delivery in Canada
DRAFT
Executive Summary
Emergency Services Working Group (ESWG) recommends on a consensus
basis the Commission order the deployment of NENA Internet-2 (i2)
compliant
Tofik Suleymanov wrote:
Darrell Long wrote:
The resulting file is not going to sound any better and its going to
take up more space. What is the reason you need a WAV file? Perhaps
there is a more efficient way to do what you are trying to do.
Darrell S. Long
BestWeb Corporation
Hi Douglas.
The Asterisk Development Team has released version 1.2.7.1 of Asterisk.This release contains only two fixes, one of which is that the Page()
application was entirely broken in version 1.2.7. If you have alreadyupgraded to 1.2.7 and you do not use the Page() application in yourdialplan,
Thanks. I've upgraded.
-Original Message-From: Josué Conti
[mailto:[EMAIL PROTECTED]Sent: Thursday, April 13, 2006 1:48
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Asterisk 1.2.7
Page()
Hi Douglas.
The Asterisk
List,
I have recently downloaded installed Asterisk2Billing
from http://www.asterisk2billing.org/
which is a great billing program for prepaid calling cards as well as SIP/IAX
users.
The problem is that our Asterisk server seems to have DTMF sensitivity
too high. If you dial
I just check the source code, Monitor uses ast_writestream and it
eventurally goes down to au_write, g723_write, etc. They don't commit to
the disk. So, in effect, if you have a lot of ram, the audio should stay
in ram until it gets swap out or the file is closed.
-Original Message-
Hi,
Good: Setting up an STUN is easy.
Bad: I have only a link to an german tuto-site.
(http://www.asteriskpbx.de/index.php?stun)
You need at least 2 network-cards.
Get the File:
wget http://mesh.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz
unpack:
# tar zxf stund_0.96_Aug13.tgz
#
Thank you Matt!!!
Matt Roth wrote:
Try switching to native MOH. You'll eliminate the decoding of the MP3s
and the host of problems that come along with using mpg123. The MOH is
handled by the same thread that's handling the call, so you should see
an overall performance benefit. Memory
Looking to possibly use the spa-942 in a business environment as a
medium class sip phone. Customer absolutely wants support for Page() or
Intercom.
Does anyone know if this phone truly handles Page() with two-way audio
correctly?
___
--Bandwidth
Has anybody used the sangoma fxo cards with asterisk? Anybody using multiple
cards? Problems with irq and such (same as with digium ones)?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|John Novack
|Sent: Wednesday, April 12, 2006 10:29 AM
|To:
Hello All,
Here's the problem, we have happily set up several Asterisk servers to offer
commercial service in the UK, our wholesale SIP termination partner
(Magrathea - use SER/CiscoGW to provide us the service on a public IP
address) - till now we have used Asterisk to connect clients on private
Hello,
Maybe someone has connected Panasonic KX-TD500 to asterisk using the
KX-TD5029 ?
I would like to have a IVR-like setup with panasonic:
Telco-BRIx2[PABX ]
Telco-PRi--[ KX ]-(KX-TD5029)--Asterisk
Telco_2nd-POTSx4---[TD500]
Any help or your comments are welcome..
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