On 14:42, Fri 02 Oct 09, Michelle Dupuis wrote:
> Has anyone written an app that monitors SIP/IAX registration attempts? A
> couple of clients are being flooded with SIP registrations (but the source
> IP changes every few hours so IPtables won't do)..
>
> I would think that any attempt to reg 5
On Fri, Oct 02, 2009 at 01:45:59PM -0700, Chris Miller wrote:
> Darrin Henshaw wrote:
> > add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
> > depending on what you are running. zaptel or dahdi.
>
> Can someone confirm when the default was changed from "never" to
> 3600 seconds? Acc
Hi,
I want to use asterisk with jack audio.
I tried the next configurations :
1 ) app_jack.so -> does not work.
2 ) chan_alsa.so without jackd started
my .asoundrc :
pcm.!default
{
type hw
card 1
}
pcm.jack1 {
type jack
playback_ports {
0 alsa_pcm:playback_1
1 a
Hi everyone,
I hope someone can help me with a problem I'm having with Cisco 7940
phones on the SIP 8.12 image. When I place a call from one of the
handsets, the call proceeds as normal for 20 seconds and is then
terminated by Asterisk (1.4.26.2):
[Oct 3 10:08:55] WARNING[1650]: chan_sip
Hello there !
I have successfully installed Asterisk on a normal server and on a
Linksys WRT610N with DD-WRT running from connected USB-stick. This is
not my first Asterisk-installation. I'm always finding a smaller and
smaller platform to install Asterisk on.
But now I encounter a problem.
I'm
On Friday 02 October 2009 14:58:51 Danny Nicholas wrote:
> Sipregisterattempts would seem to be the simplest way to do this. It is 0
> by default, changing it to 5 would stop the hacker after 5 tries.
That configuration setting is for OUTBOUND register attempts, not inbound.
The idea there is if
On Saturday 03 October 2009 04:13:40 Tzafrir Cohen wrote:
> On Fri, Oct 02, 2009 at 01:45:59PM -0700, Chris Miller wrote:
> > Darrin Henshaw wrote:
> > > add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
> > > depending on what you are running. zaptel or dahdi.
> >
> > Can someone con
On Sat, Oct 03, 2009 at 03:19:36PM +0200, jonas kellens wrote:
> Hello there !
>
> I have successfully installed Asterisk on a normal server and on a
> Linksys WRT610N with DD-WRT running from connected USB-stick. This is
> not my first Asterisk-installation. I'm always finding a smaller and
> sma
[Dropping -dev . This is not a -dev question]
On Sat, Oct 03, 2009 at 11:36:59AM +0200, Fabien COMTE wrote:
> Hi,
>
> I want to use asterisk with jack audio.
>
> I tried the next configurations :
>
> 1 ) app_jack.so -> does not work.
That should be the way to use. What do you mean by "doesn't
On Sat, Oct 3, 2009 at 6:17 AM, James Stocks wrote:
> Hi everyone,
>
> I hope someone can help me with a problem I'm having with Cisco 7940
> phones on the SIP 8.12 image. When I place a call from one of the
> handsets, the call proceeds as normal for 20 seconds and is then
> terminated by Asteri
There was still something that I did not do to debug things and that is
"sip set debug".
This showed me that the registration was rejected.
I did not saw this first but in my sip.conf I had [NSLU] in capital and
username=nslu in lower case.
This has to be the same !! So I've set both as followed :
Hello list !
SETUP :
Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk
(VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone
PROBLEM :
I've noticed that when I put down the horn of my Grandstream it still
takes a while for my GSM/CellPhone to stop ringing.
INFORMATION
Yes, it was build.
The problem is when I do a JACK_HOOK.
In app_jack.c the line jack_data->client = jack_client_open(client_name,
jack_options, &status); fails
It returns jack_data->client = NULL and status = 1
When I look in jack_sources/libjack/client.c
jack_client_open return this values when
Ideasip is down today ?
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On 3 Oct 2009, at 16:37, Jonathan Thurman wrote:
> On Sat, Oct 3, 2009 at 6:17 AM, James Stocks
> wrote:
>> Hi everyone,
>>
>> I hope someone can help me with a problem I'm having with Cisco 7940
>> phones on the SIP 8.12 image. When I place a call from one of the
>> handsets, the call proceed
I am running asterisk 1.4.26.1 and using ALSA not oss
dahdi 2.2.0
and libpri-1.4.10
I am calling into console/dsp I hear the audio just fine then after the
hangup I hear ringing
on the console/dsp.
Why would that be?
I found this bug for OSS https://issues.asterisk.org/view.php?id=13686
Does th
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