Hello
In app_meetme.c, there are two configuration file loaded i.e. meetme.conf
and sla.conf..
I want to know that if i removed whole code of sla_* and sla.conf from
app_meetme.c file..
Is this create problem for MeetMe application and register action/event...
--
Regards,
Chandrakant Solanki
Hello all,
Assuming I have 1 asterisk with 24 channels fxo and another 2 asterisk boxes
all connected iax2,
I want to grand the first asterisk box to use all the 24 channels on the
main, but I want the 2nd asterisk to use only 8 port, how can limit the
second box from receiving more than
B.Masoud @ SH wrote:
I want to grand the first asterisk box to use all the 24 channels on the
main, but I want the 2^nd asterisk to use only 8 port, how can limit the
second box from receiving more than 8 simultaneous calls?? (even if the
main have available ports)
This can be done using the
David @ULC wrote:
Looking for Genuine VPS Server for 250 ports on Rent.
Ask on the biz list instead.
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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AstriCon 2009 -
Pablo Bernasconi wrote:
My Asterisk version is 1.6.0.15, but I`ve tried it in 1.6.0.6 and
1.6.1.6 version and the same happens.
I dont know what I am missing...
Please help me.
Pablo, I did not answer in the first place because I am not completely
sure, but just guessing, PlayDTMF just
Barton Fisher wrote:
I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP.
I have one user that is having problems once he connects to asterisk.
He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk)
which goes to my asterisk IVR.
If he presses a dtmf
Carlo Dimaggio wrote:
I have a new installation with asterisk 1.6.1.6 but I'm unable to
receive calls from a SIP trunk:
[Oct 2 14:30:09] NOTICE[21554]: chan_sip.c:18523
handle_request_invite: Call from 'user001' to extension 'user001'
rejected because extension not found.
You are
We've started to use Asterisk for conferencing and have been getting some
complaints. Our configuration is that some people call in from home, but
we have a physical conference room with a Polycom. When somebody was giving
a presentation in the physical conference room, we were told that the
Hey, all. I'm seriously thinking about doing the VoIP thing at home. The
perfect platform seemed to be the Sheeva wall wart
(http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp).
It's a cute little doohicky with USB, SD-card, Ethernet, and runs on an
ARM CPU.
Hey list,
I have a problem when I host 2 SIP-accounts on the same Asterisk-server.
Asterisk picks out the SIP-account on alphabetic order A -- Z.
In my sip.conf :
register = user1:pass...@server/user1
register = user2:pass...@server/user2
[YOCAN-3starsnet]
type=peer
host=server
username=user1
On Thu, 8 Oct 2009 08:40:37 -0400 (EDT), Ken D'Ambrosio
k...@jots.org wrote:
Hey, all. I'm seriously thinking about doing the VoIP thing at home. The
perfect platform seemed to be the Sheeva wall wart
I can't help you with the issue you had compiling for IAX, but I'm
very interested in your
Ivan,
first of all thank you for your answer.
The manager function PlayDTMF only generates sound, and the dialplan
function SendDTMF only generates sound too, I´d prove it and the same
result...
So, how can I really send a DTMF to a channel?? and not just the audio..
Thank you very much, Pablo
On Thu, Oct 08, 2009 at 08:40:37AM -0400, Ken D'Ambrosio wrote:
Hey, all. I'm seriously thinking about doing the VoIP thing at home. The
perfect platform seemed to be the Sheeva wall wart
(http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp).
It's a cute
-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI(Local/3...@default-6cc4,2,
recordingcheck|20091008-093826|1255009106.184) in new stack
-- Executing [3...@default:1] Set(Local/3...@default-e393,2,
__RINGTIMER=10) in new stack
-- Executing [3...@default:2] Macro(Local/3
How do I properly quote things when I want to use the IF function on
something returning a string with blanks (e.g, CALLERID(name))?
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Hi,
Some IP Phones (Aastra) are able to send a custom HTTP request just after
registration completion.
Using this, it is possible to update phone's screen with messages like Do
Not Disturb or Forwarded To VM.
RFC 3680 (http://www.faqs.org/rfcs/rfc3680.html) provides a mecanism to
support these
You should be able to do this either via a system command or an AGI.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Thursday, October 08, 2009 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial
Hello Jonas,
I had the same problem and from my own research I found that you
could not made a distinction.
The problem is that the peer is identified based on the IP (or
IP+PORT) information found in INVITE. And you (and me) have same IP
(in my case same port as well) for several SIP accounts.
) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI(Local/3...@default-6cc4,2,
recordingcheck|20091008-093826|1255009106.184) in new stack
-- Executing [3...@default:1] Set(Local/3...@default-e393,2,
__RINGTIMER=10) in new stack
Richard Kenner wrote:
How do I properly quote things when I want to use the IF function on
something returning a string with blanks (e.g, CALLERID(name))?
Use double quotes around your variable
See:
http://www.voip-info.org/wiki/view/Asterisk+Expressions
--
Iván Stepaniuk
Alba Fotónica S.L.
Hi,
I need to get a new router for private/SOHO use of *, especially when
the kids are on internet:(
Any a good advice?
Thank you!
Best regards
HB
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AstriCon 2009 - October 13
Quick reminder before Astricon (from which we will be reporting from live):
Tomorrow's guest will be VoIP author Alex Robar. Alex has worked with
open source telephony solutions for the past four years, and has
collaborated on the development and growth of an international
Asterisk-based VoIP
A linksys wrt54g flashed with the Tomato firmware provides the best
bang for the buck when it comes to QoS for voip. Installing the
firmware is now a very easy process that can be done via the linksys
web gui on the router. If you plan to use this method, I would set up
QoS rules for full
The idea of using ICMP unreachables as a method to shut down RTP
streams (and corresponding signaling sessions) is a good one, and I'd
like to see discussion on it.
There is the rtptimeout option in sip.conf which will possibly solve
some of those symptoms (and has dangerous side-effects,
Hi,
I need to build a simple, command-line method to generate a legal and
perfect RTP stream across a network link, and analyse it on the other side
and measure network performance. Want to do this for a number of links and
over long periods. I'm trying to characterise performance of various
I like the Qos functionality. Is that a linux based package available for
other distros?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Knight
Sent: Thursday, October 08, 2009 11:15 AM
To: Asterisk Users List
Subject:
You know, I'm not entirely sure. I've never thought about using it
outside the context of Tomato. Does anyone else know if that's a
standalone (and hopefully architecture independent) package?
Michelle Dupuis wrote:
I like the Qos functionality.
Is that a linux based package
On Oct 8, 2009, at 11:41 AM, Michelle Dupuis wrote:
I like the Qos functionality. Is that a linux based package
available for other distros?
...A linksys wrt54g flashed with the Tomato firmware provides the
best bang for the buck when it comes to QoS for voip. Installing
the
I've found this to be a pretty powerful little device for $55.
http://67.210.200.94/oscommerce/store/product_info.php?cPath=24products_id=37osCsid=9f8d0c58b1f4c62761f0baf8854dd04d
Cory J. Andrews
Director New Market Initiatives
Sayers Media Group
VoIP Supply, LLC
454 Sonwil Drive
Buffalo,
2009/10/8 Danny Nicholas da...@debsinc.com
You should be able to do this either via a system command or an AGI.
How ?
When a phone registers for the first time, IMHO, no part of dialplan is
launched, is it ?
Using AMI to be notified of such registrations must be possible but I don't
know if
At 07:52 AM 10/8/2009, you wrote:
I need to get a new router for private/SOHO use of *, especially when
the kids are on internet:(
Any a good advice?
I use a cheap Linksys router that lets me set port priority and set
the port to the Asterisk box at high priority and all the others at
the
You are correct that the dialplan isn't called on a registration. The
correct way to handle this would be to modify chan_sip.c to do an action
when a phone registers. The hacky way would be to capture sip debug to a
log and process that with a daemon.You could always post a bounty.
H Oliver,
You may be able use AMI to catch the Register event (PeerStatus or Registry -
depending on what you're trying to do and how often you need to do it) and
SIPpeers to get the device address.
YMMV.
For example, when a SIP device registers to asterisk for the first time, you
see the
Sono fuori ufficio, rientrero' il 08102009
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Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or
- DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote:
STUFF
any help appreciated
There is no *FREE* G.729 codec...
Go to Digium.com and purchase the licensed G.729 codec. Yes, it's $10 per
channel but as a nice side effect, you'll also get a supported, working G.729
implementation where
Maybe you can bum a license or two off of the Astricon attendees.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, October 08, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial
there is free implementation of g729 codec
you can get it from http://asterisk.hosting.lv
On Thu, Oct 8, 2009 at 10:53 PM, Danny Nicholas da...@debsinc.com wrote:
Maybe you can bum a license or two off of the Astricon attendees…
--
*From:*
Asterisk-1.4.23.1
Dahdi-linux-2.1.0.4
Dahdi-tools-2.1.0.2
Libpri-1.4.7
Digium Wildcard TE212P
Fedora core 8 (2.6.23.1-42)
Switchtype=dms100
Im having a bit of trouble finding documentation on this, has anyone got it
working? I found this document:
Anyone pls
I have seen this message stopped sounds while I am watching asterisk
debug:
-- Called 9/0532828384
-- Call accepted by 192.168.10.220 (format ulaw)
-- Format for call is ulaw
-- IAX2/9-69 stopped sounds
-- IAX2/9-69 answered
On Thu, Oct 8, 2009 at 1:33 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:
there is free implementation of g729 codec
you can get it from http://asterisk.hosting.lv
I'm not an expert on patents, but even when you have access to the g729
implementation, the algorithm is patented, so, as
Where do I add these commands? To which file?
[macro-stdvoip]
; ${ARG1} - full dial string
; Return ${DIALSTATUS} = CHANUNAVAIL if ${VOIPMAX} exceeded
exten = s,1,Set(GROUP()=trunkgroup1) ;Set Group
exten = s,2,GotoIf($[${GROUP_COUNT(trunkgroup1)} ${VOIPMAX}]?103)
Cory Andrews wrote:
I've found this to be a pretty powerful little device for $55.
http://67.210.200.94/oscommerce/store/product_info.php?cPath=24products_id=37osCsid=9f8d0c58b1f4c62761f0baf8854dd04d
Cory J. Andrews
Director New Market Initiatives
Unfortunately there is sparse
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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database 4491 (20091008) __
The message
4491 (20091008) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
__ Information from ESET NOD32 Antivirus, version of virus signature
database 4491 (20091008) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
Hello.
I have a server installed with asterisk 1.6. I have a PSTN line that comes in
to one of those clone cards. Everything seem to be working fine. The only
problem I have is that I can't get voicemails coming from the PSTN line. All
other: SIP, IAX work fine. I can hear those ok but, when
Spinning off from another topic...what are people using for QoS / Shaping?
I'm using Wondershaper script with OK results...but I'd like better. Ideas?
___
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More specificallyI'm looking for a Linux package to allow shaping, QoS,
prioritization by port, etc.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 08, 2009 4:03 PM
To: Asterisk
On Thu, Oct 08, 2009 at 12:43:00PM -0700, Landy Landy wrote:
Hello.
I have a server installed with asterisk 1.6. I have a PSTN line that
comes in to one of those clone cards. Everything seem to be working
fine. The only problem I have is that I can't get voicemails coming
from the PSTN
On 9/10/09 2:37 AM, Pablo Bernasconi wrote:
Ivan,
first of all thank you for your answer.
The manager function PlayDTMF only generates sound, and the dialplan
function SendDTMF only generates sound too, I´d prove it and the same
result...
So, how can I really send a DTMF to a channel?? and
Antivirus, version of virus signature
database 4491 (20091008) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
__ Information from ESET NOD32 Antivirus, version of virus signature
database 4491 (20091008) __
The message was checked by ESET NOD32
I've been spending the day trying to get IMAP_STORAGE on my test box, to
evaluate for production, but I'm having no luck getting uw-imap to
build. I've tried installing it from an upstream package, but Asterisk
still isn't finding it to compile -with-imap. My google searches have
turned up very
Noah I. Engelberth wrote:
I’ve been spending the day trying to get IMAP_STORAGE on my test box, to
evaluate for production, but I’m having no luck getting uw-imap to
build. I’ve tried installing it from an upstream package, but Asterisk
still isn’t finding it to compile –with-imap. My google
On Thu, 2009-10-08 at 16:07 -0400, Michelle Dupuis wrote:
More specificallyI'm looking for a Linux package to allow shaping,
QoS, prioritization by port, etc.
snip
Spinning off from another topic...what are people using for QoS /
Shaping?
I'm using Wondershaper script with OK
On Thu, 2009-10-08 at 15:45 -0500, Jason Parker wrote:
Noah I. Engelberth wrote:
I’ve been spending the day trying to get IMAP_STORAGE on my test box, to
evaluate for production, but I’m having no luck getting uw-imap to
build. I’ve tried installing it from an upstream package, but
Richard Kenner wrote:
How do I properly quote things when I want to use the IF function on
something returning a string with blanks (e.g, CALLERID(name))?
Use double quotes around your variable
Thanks. That was my second try, but I thought that it didn't work
because I introduced a typo
- Original Message -
From: jonas kellens
To: Asterisk Mailing
Sent: Thursday, October 08, 2009 15:20
Subject: [asterisk-users] How to keep difference between 2
SIP-accounts/trunks from same server ??
Hey list,
I have a problem when I host 2 SIP-accounts on the same
Starting with Asterisk 1.2 I have always used realtime static to load
my extensions.conf into Asterisk. It worked perfectly up to version
1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I
can see that the extensions.conf file is mapped to the database:
== Parsing
,
recordingcheck|20091008-093826|1255009106.184) in new stack
-- Executing [3...@default:1] Set(Local/3...@default-e393,2,
__RINGTIMER=10) in new stack
-- Executing [3...@default:2] Macro(Local/3...@default-e393,2,
exten-vm|novm|3005) in new stack
-- Executing [...@macro-exten-vm:1] Macro(Local/3
On Thursday 08 October 2009 16:37:59 Carlos Chavez wrote:
Starting with Asterisk 1.2 I have always used realtime static to load
my extensions.conf into Asterisk. It worked perfectly up to version
1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I
can see that the
Anahi Ludueña wrote:
Hi people,
I have the following dialplan, but it doesn't have the behavior that I think
it should have.
[default]
exten = 2001,1,Answer
exten = 2001,n,Dial(local/3005)
exten = 2001,n,Hangup
exten = 3005,1,Set(__RINGTIMER=10)
exten = 3005,n,Macro(exten-vm,novm,3005)
Thanks, the answers helped me...
I was thinking to execute a macro or another context which performs a DIAL
command to a particular number. First I checked how it was working doing DIAL
directly... that is the reason why I put that context.
Thanks again...
Anahi Ludueña
Date: Fri, 9
You may want to ask on the trixbox forums.
- Original Message -
From: mickael ropars
To: asterisk-users@lists.digium.com
Sent: Tuesday, October 06, 2009 18:18
Subject: [asterisk-users] adding modules
Hi,
I am working on Trixbox. I want to create my own dial() function
Just because something is available doesn't mean that it is legal. You can get
TV Shows, Movies etc. on the internet but just because it's there it doesn't
mean that you should use it. Digium supports the Asterisk project. Shouldn't
you show your appreciation back to them ?
- Original
John Knight wrote:
You know, I'm not entirely sure. I've never thought about using it
outside the context of Tomato. Does anyone else know if that's a
standalone (and hopefully architecture independent) package?
Michelle Dupuis wrote:
I like the Qos functionality. Is that a linux based
I believe that Intel placed a 729 codec into the public domain (free), and
someone wrapped it in a nice Asterisk package for use.
No idea where - but I do recall that it is out there, and legal. Of course
it's nice to support a vendor, but free alternatives can't be shunned...
_
From:
On 9/10/09 3:31 PM, Michelle Dupuis wrote:
I believe that Intel placed a 729 codec into the public domain (free),
and someone wrapped it in a nice Asterisk package for use.
No idea where - but I do recall that it is out there, and legal. Of
course it's nice to support a vendor, but free
Does any one know about a SIP hard phone capable of sending SMS messages
(Or a SIP MESSAGE) that could be read from Asterisk dial plan??
Or at least with J2ME support, to run a little program?
Regards,
Juan
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Dudes. just use G723...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Thursday, October 08, 2009 8:47 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] g729 free
On Thu, Oct 08, 2009 at 10:31:40PM -0400, Michelle Dupuis wrote:
I believe that Intel placed a 729 codec into the public domain (free),
While you may believe what you want, Intel actually hasn't done that.
and someone wrapped it in a nice Asterisk package for use.
That package requires an
On Thu, Oct 08, 2009 at 09:23:44PM -0700, Ujjval Karihaloo wrote:
Dudes. just use G723...
Which has basically the same problems, if not worse (license-wise).
Just stay out of patent-encumbered algorithms (ones with greedy patent
holders, that is).
--
Tzafrir Cohen
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