Hi Gary,
> I not using anything to create my dialplan. I'm trying to add a softphone to
> a dialplan
> that was created a couple years ago by someone that knew what they were doing.
> Everything else in the dialplan works. As you can see I don't understand how
> to
> create a dialplan and I'm
On 12/08/2010 02:48 PM, Alex Saavedra wrote:
Jonas,
I've been using H.264 and H.263+ with a few Grandstream GVX3140. When
using H.264 the image quality was better, and required bandwidth
appeared lower compared with H.263+. In fact H.264 is expected to
consume less bandwidth for as much as 50
On 9 Dec 2010 at 22:32, Gary (Gary Kuznitz ) commented
about [asterisk-users] Audio ports:
> I see in sip debug it says Audio is at port 10342
> Express Talk suggests Audio at UDP 8000-8020
> I've tried changing Express Talk to 1 and forwarded 1-10400.
> Is there a possibility Express T
I see in sip debug it says Audio is at port 10342
Express Talk suggests Audio at UDP 8000-8020
I've tried changing Express Talk to 1 and forwarded 1-10400.
Is there a possibility Express Talk won't work in the 1 range?
Is it possible to limit Asterisk to 8000-8020?
Thank you,
Gary
--
Thanks for the reply.
On 9 Dec 2010 at 20:56, Steve (Steve Edwards )
commented about Re: [asterisk-users] (Fwd) Re: Configuring Softp:
> On Thu, 9 Dec 2010, Gary Kuznitz wrote:
>
> > I'm getting closer. Express Talk is now making the call.
> > I'm getting an error on the cmd line:
> >--
On Thu, 9 Dec 2010, Gary Kuznitz wrote:
> I'm getting closer. Express Talk is now making the call.
> I'm getting an error on the cmd line:
>-- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro("SIP/120-
> b6003810", "trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|") in
> new
On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz ) commented
about
[asterisk-users] (Fwd) Re: Configuring Softphone:
> Thank you for the reply.
>
> On 8 Dec 2010 at 13:38, Danny (Danny Nicholas ) commented
> about RE: [asterisk-users] Configuring Softphone:
>
> > -Original Message-
> > Fro
http://john8802.ru.gg/Hcthealtsor.htm
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas ) commented
about RE: [asterisk-users] Configuring Softphone:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
> Sent:
On Thu, Dec 9, 2010 at 10:02 AM, Bruce McAlister <
bruce.mcalis...@blueface.ie> wrote:
> Hi RR,
>
>
>
> I’ve not tried compiling 1.8.1-rc1 on Solaris yet and I’ve not come across
> this issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build
> error’s though. I’m not sure if the co
BTW, the issue was created yesterday, but I didn't think there was a need to
post it here but nevertheless for posterity, the Issue ID is: 18442
Thanks
\RR
On Wed, Dec 8, 2010 at 6:57 PM, RR wrote:
> On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher wrote:
>
>> On Wednesday 08 December 2010 14
On Thu, Dec 9, 2010 at 10:31 AM, Daniel Tryba wrote:
>
> You could use SIPVicious to run attacks on your own servers:
> http://code.google.com/p/sipvicious/
>
Yeah, why not? All the criminals on the internet are using it, too! ;^)
I'm seeing 1-4 scans per day on the average. And it's pretty cl
On Thu, Dec 09, 2010 at 07:57:37AM -0600, Joe Greco wrote:
> Specifically looking for examples of (or how to generate)
>
> 1).*No registration for peer '.*' (from )
> 2).*Host failed MD5 authentication for '.*' (.*)
> 3).*Failed to authenticate user .*@.*
>
> If anyone who is more fa
The AsteriskExchange.com is a directory of products and services that
complement, extend, or enhance Asterisk. As announced at AstriCon 2009, Digium
answered the demand for a marketplace that would guide Asterisk users to
available solutions. Free listings are available for free products and
se
On Tue, Dec 07, 2010 at 12:48:43AM +0100, Giuseppe D'alessio wrote:
> Hi, i have context in a dialplan, I want to "execute" this context
> without insert the Answer Application (s? ..without call any ext).
[snipped]
I'm puzzeled:
What is the question? Is something going wrong or unexpected? (I'm
Hi,
This is it: http://www.asteriskexchange.com/
It's also good to know that people from such respectful community may not know
it at all. Besides, the ones from Digium who read and moderate also don't reply
to my post - good to know that too :)
- Original Message -
From: Goke M Ar
On 12/09/2010 08:15 AM, Bruce McAlister wrote:
> I have now logged issue number 0018447 relating to this query.
The real question here is how you define 'not responding to INVITEs'.
According to the RFC, Asterisk must wait 64*T1 for a response to an
outbound INVITE, which is 32 seconds. If 'qual
On Thu, Dec 9, 2010 at 1:29 AM, Daniel Knoll wrote:
> Hey Guys,
> for debugging i need to read the Events from AMI. But i have a lot of
> unwanted "RTCPSent" Events.
> How can i filter this Events in Asterisk 1.6.2.x Version of Asterisk?
>
You can control to some extent on what kind of events a
can someone give more eduaction to me about what the asterisk exchange is
all about?
thanks
On Thu, Dec 9, 2010 at 5:43 AM, Sevana Oy wrote:
> Hi,
>
> A couple of months ago we registered our product AQuA at Asterisk Exchange.
> We were told that it collects like 14K visitors per month and kno
I'm not sure if this is the log entry you are looking for. I had many of these
last
night.
[Dec 9 06:47:51] NOTICE[5630]: chan_sip.c:15593 handle_request_register:
Registration from '"106" ' failed for '121.11.158.174' -
Wrong password
If you need more information from this Asterisk box let
Hi RR,
I've not tried compiling 1.8.1-rc1 on Solaris yet and I've not come across this
issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build error's
though. I'm not sure if the code has changed that much between 1.8.0-rc5 and
1.8.1-rc1.
I'm no coding guru by anyone's standards
On Dec 9, 2010, at 5:57 AM, Joe Greco wrote:
>
> > Hello,
> >
> > We had been seeing SIP-guessing attacks on our Asterisk server here.
> >
> > While it wasn't that hard to write a once-a-minute cron job to spank
> > the lusers, that runs once a minute and creates little spikes in the
> > usage a
sorry i am not familiar with sshguard, but you can also try ossec by
trend micro http://www.ossec.net/ it can auto-block an IP address using
iptables. you can also follow this howto for asterisk:
http://sysbrain.wordpress.com/2010/05/24/asterisk-ossec-part-ii/
hope this helps.
regards
Ron
On
I do not have log examples to provide but do have info about other issues.
There is a nocolor option in asterisk.conf that can turn off color.
logger.conf has a provision to use syslog directly.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 9, 2010, at 5:57 AM,
I have now logged issue number 0018447 relating to this query.
Thanks all for your responses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: 03 December 2010 22:53
To: Asterisk Users Mai
On 12/08/2010 11:46 PM, mayamatakeshi wrote:
> Hello,
> does anyone know a way to load test the SkypeForAsterisk module without
> actually generating calls to Skype Network? (only inside test
> environment). I mean, is there any way to simulate the endpoint SFA
> talks to?
No, there is not. It tal
Hello,
We had been seeing SIP-guessing attacks on our Asterisk server here.
While it wasn't that hard to write a once-a-minute cron job to spank
the lusers, that runs once a minute and creates little spikes in the
usage and I/O graphs, and is slower to respond than I'd really prefer.
I felt that
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