Have a look into the command logrotate
http://www.cyberciti.biz/faq/how-do-i-rotate-log-files/
On 05/08/10 21:26, Ujjval Karihaloo wrote:
Is there a setting to roll over the Master.csv CDR File in
/var/log/asterisk/cdr-csv, from and ZIP the older file once its gets
a certain size?
--
Is
Steve-
> On 08/06/2010 05:40 AM, Jeff Brower wrote:
>> Miguel-
>>
>>> El 05/08/10 14:50, Tim Nelson escribió:
- "michel freiha" wrote:
> Dear Sir,
>
> I tried to convert ilbc to ulaw and get the same result...Bad Voice
Quality
> Regards
>
Again, iLBC is p
Please can anyone help me with this ?!
I have tried renaming the sip.conf file, or tried including another file
into sip.conf like sippy.conf and then add sippy.conf =>
mysql,AsteriskDB,ast_config to extconfig.conf but all this is not working.
The only thing that changes something is my examp
On 08/05/2010 06:25 PM, Roderick A. Anderson wrote:
> Kevin P. Fleming wrote:
>> On 08/05/2010 03:52 PM, Roderick A. Anderson wrote:
>>> I have a Linux-Vserver guest running CentOS 5.5 with Asterisk 1.6
>>> installed from the asterisk.org and digium.com repositories.
>>>
>>> I have Asterisk starti
Hi,
I have followed steps which were mentioned on forum and given below. Still
couldn't get speex working. On test calls getting error "chan_sip.c:
sip_call: No audio format found to offer."
# yum install speex
# yum install speex-devel
# cd /usr/src/asterisk
# make clean
# make
# ser
On Fri, Aug 6, 2010 at 5:29 PM, Deepika Nijhawan <
deepika.nijha...@oxygen8.com> wrote:
> Hi,
>
>
>
> I have followed steps which were mentioned on forum and given below. Still
> couldn’t get speex working. On test calls getting error “chan_sip.c:
> sip_call: No audio format found to offer.”
>
>
On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote:
>> MELPe is patent encumbered,
>
>Not if used for govt/defense purposes. For commercial-only purposes, TI will
>waive royalty fees if their chip is used
>in the product. It would have been nice if Digium had considered the many
>adv
On Fri, 06 Aug 2010 07:40:44 -0500, Michael Graves wrote:
>On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote:
>
>
>
>>> MELPe is patent encumbered,
>>
>>Not if used for govt/defense purposes. For commercial-only purposes, TI will
>>waive royalty fees if their chip is used
>>in the produ
Hi,
May you also need to install *speex-tools* . if problem retain then let us
know about your Linux distribution and Asterisk version.
Regards
On Fri, Aug 6, 2010 at 4:59 PM, Deepika Nijhawan <
deepika.nijha...@oxygen8.com> wrote:
> Hi,
>
>
>
> I have followed steps which were mentioned on fo
To provide a reliable fax solution for users connected to a Asterisk 1.6.2.6
server i have tested a few T.38 capable ATA's:
- Patton M-ATA
- Grandstream HandyTone 486
- Fritz!Box 7170
I have tried Asterisk 1.6.2.6 compiled with SpanDSP-0.0.6pre17 and also
Asterisk 1.6.2.6 with Fax for Asterisk
2010/8/5 Dario Quiroz
> Hi all!
> Are someone using a CDR report? I have an Asterisk 1.6 running perfect but
> I need a web based report of CDRs.
> Nothing big, only the basic. Have anybody a how-to or a link?
>
Look for asterisk-stats (or cdr-stats as it has been renamed) : users are
rather ple
Hi Chandrakant
I have checked and it shows func_speex module is enabled.
Where can I install speex-tools from ?
Asterisk version 1.6.2.10 and Centos 5.5 are installed.
---
Kind Regards,
Deepika Nijhawan
VoIP Engineer
Oxygen8 Communications
--
___
I have a rather simple setup running under Asterisk 1.4. I'd like to
move it to a new install of 1.6. Before I bring it online are there any
gotchas I should look for? A Gotcha README would be better but
searching with Google and the forums, for me, gets hits that deal with
hardware issues -
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A.
Anderson
Subject: [asterisk-users] Using a 1.4 config with 1.6
I have a rather simple setup running under Asterisk 1.4. I'd like to
move it to a new install of 1.6. Before I b
I have been seeing some attempts to register devices on my Asterisk
and I want to reconfigure it so that devices will be registered only
if they are from the correct address, ie 192.168.1.8/255.255.255.255.
I thought using a config like
deny=0.0.0.0/0.0.0.0
permit=192.168.1.8/255.255.255.255
but
Hi,
Currently CentOS yum repository does not provide speex-tools so you have to
install it manaully. follow the steps given below.
1. first remove existing speex pacages yum remove speex*
2. run following to install required rpms
rpm -ivh
http://www.lfarkas.org/linux/packages/centos/5/i386/gstre
Danny Nicholas wrote:
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roderick A.
> Anderson
> Subject: [asterisk-users] Using a 1.4 config with 1.6
>
> I have a rather simple setup running under Asterisk 1.4. I'd like to
> move it
On Fri, Aug 6, 2010 at 10:36 AM, Roderick A. Anderson
wrote:
> I have a rather simple setup running under Asterisk 1.4. I'd like to
> move it to a new install of 1.6. Before I bring it online are there any
> gotchas I should look for? A Gotcha README would be better but
> searching with Google
Hi,
Do any of you have these phones ? How have you found it ? Are you using them
over WiFi or hard wired ? Does it play nicely with Asterisk ?
Need to replace my Snom M3s and this phone maybe a contender.
--
Thanks, Phil
--
_
On 08/06/2010 07:45 AM, Frank Church wrote:
> I have been seeing some attempts to register devices on my Asterisk
> and I want to reconfigure it so that devices will be registered only
> if they are from the correct address, ie 192.168.1.8/255.255.255.255.
>
> I thought using a config like
>
> deny
I have a site running 1.4.17 with Zaptel. They want to add a TE420P for
additional T1 capacity. I'm 99% sure this will work, anyone aware of a
reason it wont?
Thanks,
James
--
_
-- Bandwidth and Colocation Provided by http://w
On Fri, Aug 6, 2010 at 10:33 AM, James Texter wrote:
> I have a site running 1.4.17 with Zaptel. They want to add a TE420P for
> additional T1 capacity. I'm 99% sure this will work, anyone aware of a
> reason it wont?
>
> Thanks,
>
> James
>
>
I've got a client running a TE420P with asterisk 1.4
>>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
>>Subject: Re: [asterisk-users] Asterisk 1.4 and TE420P
>On Fri, Aug 6, 2010 at 10:33 AM, James Texter
wrote:
>I have a site running 1.4.17 with Zaptel. They want to add
You cannot use realtime static and the other realtime tables at the
same time. You will need to use realtime and then use something like
the EXEC command in sip.conf to execute a script that then pulls the
register statement from your database. Or use the realtime static table
for everyth
On Fri, Aug 6, 2010 at 11:24 AM, Danny Nicholas wrote:
> *>>From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren Selby
> >>*Subject:* Re: [asterisk-users] Asterisk 1.4 and TE420P
>
>
>
> >On Fri, Aug 6, 2010 at 10:33 AM, James
On 08/06/2010 04:43 PM, Jeff Brower wrote:
> Steve-
>
>>On 08/06/2010 05:40 AM, Jeff Brower wrote:
>>> Miguel-
>>>
El 05/08/10 14:50, Tim Nelson escribió:
> - "michel freiha" wrote:
>> Dear Sir,
>>
>> I tried to convert ilbc to ulaw and get the same result...Bad Vo
- Original Message -
> On Fri, 6 Aug 2010, --[ UxBoD ]-- wrote:
>
> > Hi,
> >
> > Do any of you have these phones ? How have you found it ? Are you
> > using them over WiFi or hard wired ? Does it play nicely with
> > Asterisk ?
> >
> > Need to replace my Snom M3s and this phone maybe a co
Hi all,
i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for
OCS2007 R2 following the HowTo
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.
I can call the OCS from Asterisk and vice versa.
The only thing that do
My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card to
put into my Asterisk box.
I know from the wonderful O'Reilly book that the proper installation is Zaptel
à libpri à Asterisk. Is it possible to simply reinstall in that order once I
have installed the card and hav
>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
jeremy.hellst...@synovate.com
>Subject: [asterisk-users] Reinstalling Asterisk due to hardware changes
>My purely SIP experiment has failed so I am purchasing a Digium E1/T1 card
to put
Steve-
> El 05/08/10 14:50, Tim Nelson escribió:
>> - "michel freiha" wrote:
>>> Dear Sir,
>>>
>>> I tried to convert ilbc to ulaw and get the same result...Bad Voice
>> Quality
>>> Regards
>>>
>> Again, iLBC is poor quality to begin with. You can't take
Hi all,
i finally got it working to use Asterisk 1.6.1.20 as my PSTN gateway for
OCS2007 R2 following the HowTo
http://blogs.technet.com/b/gclark/archive/2008/10/09/asterisk-1-6-with-office-communications-server-2007.aspx.
I can call the OCS from Asterisk and vice versa.
The only thing tha
Oliver,
Did you happen to use a Dialog Media gateway in the mix, or is
this straight Asterisk to OCS? We are implementing this in my shop, and running
a bit of a ground (trying to use the MediaGateway). Any help anyone could
provide would be wonderful. Thanks all.
Lyle J. McKarn
I did it straight Asterisk to OCS using the OCS Mediation Server. We do have
Dialogic Diva Server Cards which are able to be used as Media Gateway too using
an additional software called SipControl (not 100% sure about the name) but as
this software needs to be licensed separately I prefer a dir
On 6 August 2010 16:21, Bruce Ferrell wrote:
> On 08/06/2010 07:45 AM, Frank Church wrote:
>> I have been seeing some attempts to register devices on my Asterisk
>> and I want to reconfigure it so that devices will be registered only
>> if they are from the correct address, ie 192.168.1.8/255.255.
Sorry for asking this after 6 week of staleness on this post, but do you
know which FCC registration is needed? We already have a 214. Is that it?
Thanks.
Regards
HASSAN
On Thu, Jun 24, 2010 at 00:43, Tarek Sawah wrote:
> i consuleted didforsale.com regarding the wholesale thing and their
On Thu, 5 Aug 2010, Faheem wrote:
Hey, Is there any way to share MySQL connection between different agi's.
No.
Each AGI is executed as a separate process. While debugging (not
interacting with a "real" call), you can executed your AGI completely
independent from Asterisk from the command li
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
> You cannot use realtime static and the other realtime tables at the
> same time. You will need to use realtime and then use something like
> the EXEC command in sip.conf to execute a script that then pulls the
> register statement from your data
On 08/06/2010 06:45 PM, Carlos Chavez wrote:
> Or use the realtime static table for everything.
What do you mean by "everything" ?! What is this "everything" ?!
You mean all the sip options in a database and so no sip.conf file ?!
Kind regards,
Jonas.
--
> You cannot use realtime static and the other realtime tables at the
> same time. You will need to use realtime and then use something like
> the EXEC command in sip.conf to execute a script that then pulls the
> register statement from your database. Or use the realtime static table
> f
This works. I have tested with the following settings:
In regards to the specifics of your question:
In sip.conf:
dynamic_exclude_static=yes
In users.conf, for each user (changing the permit statement to the ip of
each user):
hassip=yes
host=dynamic
registersip=yes
deny=0.0.0.0/0.0.0.0
permit=192.
Hi,
Im not able to set the outgoing number in filename for asterisk recordings
Following is what I have done in
/var/lib/asterisk/agi-bin/recordingcheck file
.
.
.
include("phpagi.php");
/***
On 08/06/2010 02:16 PM, Frank Church wrote:
> On 6 August 2010 16:21, Bruce Ferrell wrote:
>
>> On 08/06/2010 07:45 AM, Frank Church wrote:
>>
>>> I have been seeing some attempts to register devices on my Asterisk
>>> and I want to reconfigure it so that devices will be registered only
>>
On 08/07/2010 03:15 AM, Jeff Brower wrote:
> Steve-
>
>> El 05/08/10 14:50, Tim Nelson escribió:
>>> - "michel freiha"wrote:
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice
>>> Quality
Regards
>>>
What kind of attack can they reform calling in?
On Aug 6, 2010 1:12 AM, wrote:
> I am setting filters, etc. on variables that attackers can send asterisk
> when they call (for example when they initially call into asterisk).
>
> So far, I am filtering:
>
> exten
>
> CALLERID(name)
>
> CALLERID(nu
On 08/06/2010 07:30 PM, Bruce Ferrell wrote:
> On 08/06/2010 02:16 PM, Frank Church wrote:
>
>> On 6 August 2010 16:21, Bruce Ferrell wrote:
>>
>>
>>> On 08/06/2010 07:45 AM, Frank Church wrote:
>>>
>>>
I have been seeing some attempts to register devices on my Asterisk
On 7 August 2010 03:54, Bruce Ferrell wrote:
> On 08/06/2010 07:30 PM, Bruce Ferrell wrote:
>> On 08/06/2010 02:16 PM, Frank Church wrote:
>>
>>> On 6 August 2010 16:21, Bruce Ferrell wrote:
>>>
>>>
On 08/06/2010 07:45 AM, Frank Church wrote:
> I have been seeing some attempts
Well, I'm not sure actually. I was attacked in June by someone who racked up
between $800 and $900 in international calls to places in the middle of
Africa, Korea, etc. So, I am motivated to secure this. I have made it much
much more secure, definitely, but am looking for as many ways to further
lo
On Fri, Aug 6, 2010 at 10:53 PM, wrote:
> Someone from Amsterdam was trying to register yesterday using an automated
> program which tried roughly 1,000 or so username password combinations
> before I shut asterisk down and added his/her ip to iptables to drop it. I
> wonder if I can configure th
Hi
Can you tell me which Linux OS are you used and what is speex / speex-devel
version.
Can you give details for above?
--
Regards,
Chandrakant Solanki
On Fri, Aug 6, 2010 at 6:22 PM, Nasir Iqbal wrote:
> Hi,
>
> May you also need to install *speex-tools* . if problem retain then let us
> k
Am I really the only one having problems with this new "shrinkcallerid"? I
can't find anything on Google about it.
Was happening on 1.6.2.10 and now on 1.8.0-beta2
In sip.conf shrinkcallerid=no, yet a name like "Joe Smith" ends up being
"JoeSmith"
Whoever though this up anyway is stupid. Why wo
Hi All,
I have Sangoma A200 Card installed on my system,
I have centos 5.5 with 64 bit,
Here are the description for asterisk and dahdi.
Asterisk 1.6..2.9
Dahdi: 2.3.0.1
I have two issues with dahdi
1) I am not getting full callerid on my phones from sangoma card to asterisk
users. if i am connecti
> Use fail2ban. Also, read some of the security advisories from earlier
this year about being sure to always use a FILTER statement whenever you're
dialing using > a variable (most notably ${EXTEN}).
http://downloads.asterisk.org/pub/security/AST-2010-002.html
Thanks Warren!!
From: aster
53 matches
Mail list logo
