On Mon, Sep 27, 2010 at 5:04 AM, Rogelio scubac...@gmail.com wrote:
Please excuse the ignorance of these questions. (I still have yet to
install and configure a VoIP solution.) Depending on the feasibility
of some of these questions, I was going to try my hand at installing
something...
It
Dear All,
As per this article
http://www.voip-info.org/wiki/view/Asterisk+cmd+Smsasterisk support
PSTN to SMS and from SMS support endpoints to PSTN.
In my scenario we have SS7 based E'1 on which our SMS provider sending SMS
on our DID numbers and my all DID's are registered on OPENSER.
What I
Hello,
Is RFC3329(Security Mechanism Agreenment for SIP) supported in Asterisk ?
Regards,
Sijan
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Hello
Change your a2b 1.8 version to a2b 1.7.1 version
1.7.1 version is a stable.
And forum for a2b is http://forum.asterisk2billing.org/index.php
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Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
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On Fri, Sep 24, 2010 at 02:30:12PM -0400, Paul Belanger wrote:
Am I missing something?
DEBUG_THREADS
Thanks, I guess I should have RTFM :)
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Daniel Tryba
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On Mon, 27 Sep 2010 15:00:36 +0200, Roberto Piola wrote:
Hard drive speed may differ between 5400rpm and 7200rpm.
in high performance server environment, you can get 1 and 15000rpm
drives as well... the fastest, the better (and the more expensive).
moreover, if you have 6 disks in raid
Hi all,
Has anyone tried Asterisk on Arch? I am currently running the latest Arch and I
am thinking about installing Asterisk or is some other distro better?
I have been using Debian in the passed. Many thanks for any reply!
Christian--
Michelle Dupuis mdup...@ocg.ca writes:
Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com
(also on the voip wiki)
You get the cluster/heartbeat replication without needing to add openSER
or full HAlinux. A simpler approach - easier to config and manage
How do you handle
bruce bruce bruceb...@gmail.com writes:
Other than the price difference (2.5 is more expensive and can't find
many of the 1TB or so) is there any preference, advantage, or
disadvatage of chosing 2.5 HDD or 3.5 when it comes to the server
operations or Asterisk operation?
There is no
Hi!
thank you for your good answers.
Another related question:
I tried using Page() and it works perfectly, but I need to implement a
slightly different behavior, and I'm looking into ways of implementing it.
When a user picks up the phone and chooses to page the speakers, the
call should
HAAST runs a sync script a regular intervals (time to sync data prior to a
failover check etc)
HAAST includes a sample script which syncs voicemail (and config, etc) files
using rsync from master to slave. After a master/slave reversal the process
automatically reverses.
MD
is there a trick to get asterisk (1.6.2.13) to propagate
codec-changing sip reinvites when directrtpsetup=yes?
i'm trying to route calls to a gateway without keeping asterisk in the
rtp stream.
the gateway is first routing the call to a media server. when
connecting the call to the downstream
Hello,
I'm trying to compile DAHDI on DEBIAN but i have the following error:
r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make
echo You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
On 09/27/2010 11:02 AM, Eugene Oden wrote:
is there a trick to get asterisk (1.6.2.13) to propagate
codec-changing sip reinvites when directrtpsetup=yes?
i'm trying to route calls to a gateway without keeping asterisk in the
rtp stream.
You are looking for a SIP proxy; Asterisk is not a SIP
Source files aren't automatically installed on every install.
This link should help:
http://www.cyberciti.biz/faq/howto-install-kernel-headers-package/
Dean Hoover
Milwaukee, Wisconsin
On 9/27/2010 11:09 AM, Danny Dias wrote:
Hello,
I'm trying to compile DAHDI on DEBIAN but i have the
Thanks Dean,
I've done it before, that's why i'm here asking :( take a look:
r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# apt-cache search
linux-headers-$(uname -r)
linux-headers-2.6.26-2-amd64 - Header files for Linux 2.6.26-2-amd64
r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# sudo
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote:
r...@sangoma-testing:/usr/src/dahdi-linux-2.1.0.4# make
echo You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
exit 1
make:
Done my friend:
r...@sangoma-testing:/lib/modules/2.6.26-2-amd64/kernel/drivers#
module-assistant prepare
Getting source for kernel version: 2.6.26-2-amd64
apt-get install linux-headers-2.6.26-2-amd64
Reading package lists... Done
Building dependency tree
Reading state information... Done
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote:
What should i do?
aptitude install module-assistant
m-a a-i dahdi
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ok, thanks. i was beginning to suspect as much but was hoping to
limit the number of components in our configuration.
thanks,
gene
On Mon, Sep 27, 2010 at 11:13 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 09/27/2010 11:02 AM, Eugene Oden wrote:
is there a trick to get asterisk
¿ahhh?
2010/9/27 Roger Burton West ro...@firedrake.org
On Mon, Sep 27, 2010 at 06:09:15PM +0200, Danny Dias wrote:
What should i do?
aptitude install module-assistant
m-a a-i dahdi
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On Mon, Sep 27, 2010 at 06:28:04PM +0200, Danny Dias wrote:
You do not appear to have the sources for the 2.6.26-2-amd64 kernel
installed.
exit 1
make: *** [modules] Error 1
The same result :(
Where did you get the dahdi source? There is no official 2.1.0.4
in any debian version. If you
On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias ing.diasda...@gmail.com wrote:
What should i do?
Try with the lastest DAHDI version, 2.4.0.
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Paul Belanger | dCAP
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blog.polybeacon.com
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I've these versions of DAHDI running into another Server with no
problem...it seems to be a problem with dependencies, but i can't find the
trick :(
2010/9/27 Paul Belanger paul.belan...@polybeacon.com
On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias ing.diasda...@gmail.com
wrote:
What should i
On 9/27/10 8:57 PM, Michelle Dupuis wrote:
HAAST runs a sync script a regular intervals (time to sync data prior to a
failover check etc)
HAAST includes a sample script which syncs voicemail (and config, etc) files
using rsync from master to slave. After a master/slave reversal the
The same problem!
r...@sangoma-testing:/usr/src/dahdi-linux-complete-2.4.0+2.4.0# make all
make install make config
make -C linux all
make[1]: Entering directory
`/usr/src/dahdi-linux-2.1.0.4/dahdi-linux-complete-2.4.0+2.4.0/linux'
make -C drivers/dahdi/firmware firmware-loaders
make[2]:
On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias ing.diasda...@gmail.com wrote:
The same problem!
What is the output from the following?
$ ls -la /lib/modules/
$ ls -la /usr/src/linux
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Paul Belanger | dCAP
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Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
That's exactly what we recommend for DB/realtime installs. HAAST's focus is
the failover, promotion, assignment of IP, etc. but links to standard tools for
file/db sync. In line with the philosophy of try to not be everything to
everybody...
From:
On Sep 27, 2010, at 1:29 PM, Michelle Dupuis wrote:
That's exactly what we recommend for DB/realtime installs. HAAST's focus is
the failover, promotion, assignment of IP, etc. but links to standard tools
for file/db sync. In line with the philosophy of try to not be everything to
Thanks Dave. I'll do a complete system restart when I can, and see if it
helps. (an Asterisk restart didn't).
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Platt
Sent: Monday, September 27, 2010
Greetings:
I have a working configuration for SCCP on our LANS which doesn't
route RTP correctly to a skinny phone behind NAT registering from
a remote public IP.
Configuration:
asterisk 1.4.35 servicing only skinny phones trunked to
asterisk 1.2.40 which services chan_phone FXS, zap FXO
and
Hi list,
I'm trying to test an IVR system with recorded prompts and would
like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3
is slin; Need it the other way so I can do DAHDI-- IAX testing.
On Mon, Sep 27, 2010 at 01:02:04PM -0500, Danny Nicholas wrote:
I'm trying to test an IVR system with recorded prompts and would
like to be able to call 1234 and have the codec be gsm, 2234 slin, 3234
ulaw, etc. I know I can set up 3 users where #1 is gsm, #2 is ulaw and #3
is slin;
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Tryba
Sent: Monday, September 27, 2010 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to pick a codec
Has any of you tested Vyatta Load balancing and fail over solution with
Asterisk? It uses heartbeat and works like magic with regular traffic but
didn't have the time nor chance to test it with VoIP traffic.. but I think
it's the same way.
Anyone?
-Original Message-
From:
i think it's SIP_CODEC now .. and not _SIP_CODEC?
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
From: da...@debsinc.com
To: dan...@tryba.nl;
From: da...@debsinc.com
To: dan...@tryba.nl; asterisk-users@lists.digium.com
Date: Mon, 27 Sep 2010 13:30:08 -0500
Subject: Re: [asterisk-users] How to pick a codec on the fly
-Original Message-
From:
Thank you guys for the responses, actually that piece that was missing is
logger reload, only when I issue this command i can see the dedug turning
on, i though reload is enough to reload everything but it seems like
logger reload is specifically needed to apply changes, Thanks a lot Steve
and
From: da...@debsinc.com
To: dan...@tryba.nl; asterisk-users@lists.digium.com
Date: Mon, 27 Sep 2010 13:30:08 -0500
Subject: Re: [asterisk-users] How to pick a codec on the fly
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Thanks guys. Amazing feedback.
Sounds like 2.5 is a better choice for being less in size (easier access
for voicemail for example), as fast as 3.5 HDD in RPM, and allows 6 HDDs
per Node which allows more RAID choice.
However, it does come to be more expensive.
Thanks again,
On Mon, Sep 27,
On 10-09-26 01:00 PM, bilal ghayyad wrote:
First of all, I am looking to have the H323 Gatekeeper service available at
Asterisk, and really I do not know if 1.4 or 1.6 or 1.8 started implementing
H323 gatekeeper functionality or not?
Until 1.4.26.2 version, there is no h323 gatekeeper
On 10-09-26 02:55 PM, Ira wrote:
At 10:37 PM 9/24/2010, you wrote:
You probably need to install libssl-dev then rerun ./configure. At
least I did (Debian Lenny). Seems chan_sip needs res_crypto which
needs libssl.
Thanks, I tried to figure out what I needed but I failed. That was
it,
On 09/22/2010 05:49 PM, Matteo Fortini wrote:
I'm building a paging system composed of roughly 10 switches in daisy
chain, with an embedded box with a speaker and a microphone for each
switch. The embedded box runs my software.
I'm not sure if this is much use to your setup - but I've
How do you handle replicating voice mails?
I do that by putting the voicemails into MYSQL and replicating that.
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Check out HAAST (High Availability ASTerisk) at [1]www.generationd.com
Bit out of my pricing. It must be possible to do it using downloadable
open-source.
Dan
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Hello Paul,
Here is the output of the commands:
r...@sangoma-testing:/home# ls -la /lib/modules/
total 12
drwxr-xr-x 3 root root 4096 2010-09-24 10:21 .
drwxr-xr-x 13 root root 4096 2010-09-27 12:57 ..
drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64
r...@sangoma-testing:/home# ls
Did you install the header files after ./configure was run? If so redo the
./configure command and see what that does.
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Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2010, at 3:57 PM, Danny Dias wrote:
Hello Paul,
Here is the output of the commands:
Thanks Jim,
What do you mean with redo ?
I did not run the ./configure, i'm installing dahdi-linux and just need :
make make install
The problem is when i issue make
Thanks for your answer my friend!
2010/9/28 Jim Dickenson dicken...@cfmc.com
Did you install the header files after
Sorry Paul,
My mistake...i repeat the commands again:
r...@sangoma-testing:/usr/src# ls -la /lib/modules/
total 28
drwxr-xr-x 7 root root 4096 2010-09-27 19:35 .
drwxr-xr-x 13 root root 4096 2010-09-27 12:57 ..
drwxr-xr-x 3 root root 4096 2010-09-27 19:29 2.6.26-1-amd64
drwxr-xr-x 4 root root
On Mon, Sep 27, 2010 at 6:57 PM, Danny Dias ing.diasda...@gmail.com wrote:
r...@sangoma-testing:/home# ls -la /lib/modules/
total 12
drwxr-xr-x 3 root root 4096 2010-09-24 10:21 .
drwxr-xr-x 13 root root 4096 2010-09-27 12:57 ..
drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64
$
On Mon, Sep 27, 2010 at 7:36 PM, Danny Dias ing.diasda...@gmail.com wrote:
Is that Ok?
$ uname -r
Also what version of Debian?
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Paul Belanger | dCAP
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Do you not need to do a ./configure command before make make install? If so
issue the ./configure command again and see if that fixes the problem.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2010, at 4:32 PM, Danny Dias wrote:
Thanks Jim,
What do you
On Mon, Sep 27, 2010 at 9:05 PM, Jim Dickenson dicken...@cfmc.com wrote:
Do you not need to do a ./configure command before make make install? If
so issue the ./configure command again and see if that fixes the problem.
No, it does not exist for DAHDI.
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Paul Belanger | dCAP
Polybeacon |
Hi All.
got this problem that IP phones could not re-register to my server. even
if device is power cycled it still would not register. the solution i
found was to change the SIP port settings on the phone and it will
register. but after registration expires and its time to re-register the
packet tracing RTP at the border router shows (on the LAN segment
of the asterisk host), e.g.
11:43:54.380567 IP (tos 0x0, ttl 64, id 737, offset 0, flags [DF], proto
UDP (17), length 200)
pbx1.cybertheque.net.20002 192.168.0.3.29112: UDP, length 172
Shouldn't 'nat=yes' in skinny.conf
27.09.2010 16:25, Philipp von Klitzing пишет:
Hi!
Well, only problem I see, is to how pass channel name from macro to h
extension...
SHARED() or CDR(userfield)
Philipp
Looks like I still don't understand how SHARED works :-(
Let's say, I dial my softphone:
The AstLinux Team is happy to announce the release of AstLinux 0.7.3.
This update contains mostly bug fixes and security updates. All current
users of AstLinux are encouraged to update to this release.
Updating can be performed from the web interface or from the command
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