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(num)
What other fields or variables would an attacker be able to use in the
packets that they s
Hi Faheem,
You need to build some daemonized application, here FastAGI will help you
Regards
On Fri, Aug 6, 2010 at 10:54 AM, Faheem wrote:
> Hey, Is there any way to share MySQL connection between different agi's.
> Actually when call comes to asterisk box it executes various agi scripts
> se
Hey, Is there any way to share MySQL connection between different
agi's.Actually when call comes to asterisk box it executes various agi scripts
sequentially. Each script checks various values by making a
new MySQL connection and then execute query and then disconnects.
So, Ideally there should
Suddenly a couple days ago all of our SIP registrations are missing the
Mailbox entry. We are using MySQL Add-on for realtime.
Anyone have any idea why? Mailbox is still in the mysql tables.
--
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-- Bandwidth and Colocation Prov
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 poor quality to begin
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 starting (service asterisk start) but see errors about
>> dahdi
>> This just made me remember some comment on the iax.conf sample file...
>>
>> disallow=lpc10; Icky sound quality... Mr. Roboto.
>>
> LPC10 is a very old codec, from early 1980s. LPC10 doesn't do a good job
> with pitch detection so it tends to have a
> 'robotic' sound. With
logrotate
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Thu, Aug 5, 2010 at 4:26 PM, Ujjval Karihaloo
w
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 poor quality to begin with. You can't take a poor audio
>> sa
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 starting (service asterisk start) but see errors about
> dahdi in /var/log/asterisk/messages.
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 starting (service asterisk start) but see errors about
dahdi in /var/log/asterisk/messages.
... ERROR[25658] codec_dahdi.c: Failed to open /dev/dahdi/tr
Manmohan wrote:
> I commented locale.php in defines.php and it perfectly worked well.
> Now i am wondering what is this invite participants for, while adding
> conference. wherein it asks for first name, lastname, emailaddress &
> telephone number..
The 'Invite Others' option is mostly for insta
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 a poor audio
sample and make it better by converting it t
Michel-
> I tried to convert ilbc to ulaw and get the same
> result...Bad Voice Quality
I think you have to be more specific when you say "bad voice quality". Like
what? Worse than a cellphone call? Gaps
of audio missing? Robotic or "cyborg" sound? Static? A background tone or
buzzing?
i
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?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to A
1.7 for ASteriskNOw
I will investigate..Thx for the ideas!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, August 05, 2010 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
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?
Thanks in advance!!
--
Atenciosamente,
---
Hi Dan,
I commented locale.php in defines.php and it perfectly worked well.
Now i am wondering what is this invite participants for, while adding
conference. wherein it asks for first name, lastname, emailaddress &
telephone number..
Let me brief you how i had done this setup. I had created a SI
- "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 a poor audio sample
and make it better by converting it to a codec with better 'resolution'.
Dear Sir,
I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality
Regards
On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson wrote:
> - "michel freiha" wrote:
> >
> > Dear All,
> >
> > i would like to ask please if someone tried to make a codec conversion
> from ilbc to g729,
Danny Nicholas wrote:
>> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
> Lesher
>> Subject: Re: [asterisk-users] AMI Command
>
>> Actually, what you probably want is the CoreShowChannels command.
>
>> Tilghman Lesher
>
> To
On Thu, Aug 5, 2010 at 11:14 AM, Felipe Figueiredo <
felipe.figueired...@gmail.com> wrote:
> Yes. Unless you use "make samples" while compiling the new Asterisk, you
> won't lose your confg files.
> I'm afraid there's no 1.7 version of Asterisk. [?]
>
>
>
But there is a 1.7 version of Asterisk
>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval
Karihaloo
>Subject: Re: [asterisk-users] COnfig File question
>Hi All:
> If we upgrade asteriskNow from 1.4.18 to 1.7.0; just want to make sure
all the config file functio
Yes. Unless you use "make samples" while compiling the new Asterisk, you
won't lose your confg files.
I'm afraid there's no 1.7 version of Asterisk. [?]
On Thu, Aug 5, 2010 at 1:01 PM, Ujjval Karihaloo wrote:
> Any answers would be appreciated
>
>
>
> Thx
>
> UK
>
>
>
>
>
>
>
> *From:* aster
Any answers would be appreciated
Thx
UK
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Thursday, July 29, 2010 4:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] COnfi
>From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
>Subject: Re: [asterisk-users] AMI Command
>Actually, what you probably want is the CoreShowChannels command.
>Tilghman Lesher
To second this; core show channels doesn't r
On Thursday 05 August 2010 06:05:48 Ron wrote:
> Thank you. i think i would go for this solution.
>
> On 8/5/10 4:53 PM, Gareth Blades wrote:
> > Ron wrote:
> >> Hi,
> >>
> >> Is there a way to check on AMI if a user is currently engage on the
> >> phone? i would like to display on my portal whethe
- "michel freiha" wrote:
>
> Dear All,
>
> i would like to ask please if someone tried to make a codec conversion from
> ilbc to g729, because i did that but the voice quality was too bad and a lot
> of disconnection..
>
> Can i get your feedback regarding this issue please?
>
> rega
On Wed, Aug 4, 2010 at 10:12 PM, Philipp von Klitzing
wrote:
> Ok, here's the challenge:
>
> I would like to be able to find, match - and then react - upon prompts
> that are presented by the outbound/remote side of a call. Think mobile
> phone and "This user is temporarily unavailable".
>
> Colle
Thank you. i think i would go for this solution.
On 8/5/10 4:53 PM, Gareth Blades wrote:
> Ron wrote:
>> Hi,
>>
>> Is there a way to check on AMI if a user is currently engage on the
>> phone? i would like to display on my portal whether a user is calling or
>> not.
>>
>> thank you
>>
>> regards
>
> Only when I configure my Grandstream to use only G726 (I have 8
> choices), I see that the g726-codec is used.
> When I configure 7 x g726 and 1 x alaw, then again alaw is used !
>
> Is it normal that Asterisk has such a great preference for alaw ?! The
> moment the peer suggests codec alaw (ev
Hi!
> Although my previous posts in this forum have not received satisfying
> answers, here is another question from me.
You might want to consider to reqest a refund. ;->
> my question is can i use ChanIsAvail function to get the status of a user
> in the format SPI/user-id if i provide user i
Dear All,
i would like to ask please if someone tried to make a codec conversion from
ilbc to g729, because i did that but the voice quality was too bad and a lot
of disconnection..
Can i get your feedback regarding this issue please?
regards
--
_
Ron wrote:
> Hi,
>
> Is there a way to check on AMI if a user is currently engage on the
> phone? i would like to display on my portal whether a user is calling or
> not.
>
> thank you
>
> regards
> Ron
>
You could get it to run a command and do 'core show hints' and parse the
result. You wi
On Thu, Aug 5, 2010 at 11:28 AM, Ron wrote:
> Hi,
>
> Is there a way to check on AMI if a user is currently engage on the
> phone? i would like to display on my portal whether a user is calling or
> not.
>
# Asterisk Manager API Action CoreShowChannels: List currently active
channels (Priv: syst
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
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-- Bandwidth and Colocation Provided by http://
hello,
Although my previous posts in this forum have not received satisfying
answers, here is another question from me.
my question is can i use ChanIsAvail function to get the status of a user in
the format SPI/user-id if i provide user in sip uri like this
ChanIsAvail(SIP/u...@153.18.x.x:5062)
On 08/03/2010 04:21 PM, Philipp von Klitzing wrote:
> Also:
>
> There are at least two implementations of the g726 codec, i.e. g726 and
> g726aal2. For this also look at the g726nonstandard setting in sip.conf.
> It is quite possible that your problem is here.
>
I have the following setting in
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