On Tue, Dec 22, 2015 at 09:42:04AM +, Luca Bertoncello wrote:
> Is it not this:
>
> http://lists.digium.com/pipermail/asterisk-users/2015-December/288078.html
>
> ?
>
> sip set debug 42 should be a little trick to enable more debugging...
> So I got in this list some months ago...
No,
On Tue, Dec 22, 2015 at 09:30:52AM +, Luca Bertoncello wrote:
> Zitat von Sebastian Kemper :
>
> Hi Sebastian
>
>
> I tried with
>
> sip set debug 42
> sip set verbose 42
>
> The result was in my first E-Mail...
Hi Luca,
I don't remem
On Tue, Dec 22, 2015 at 07:19:47AM +0100, Luca Bertoncello wrote:
> "Brian ::" schrieb:
>
> > sip trace?
>
> Could you please explain? I'm not a VoIP-expert...
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
Hi Luca,
Brian suggests to check the SIP traces. You can either enable SIP
deb
Hello all,
My Asterisk is between my ITSP and a SIP phone. I cannot do direct media
between the provider and the SIP phone, but I would like Asterisk to
locally RTP bridge the two channels using native_rtp.
Example:
> Bridge cfb56606-6b40-4da7-a6fa-6499e183cdbb: switching from simple_bridge
> t
Am 16. September 2015 18:48:16 MESZ, schrieb Daniel Heckl
:
>Sebastian,
>
>If I have understood you correctly, the SIP communication is now via
>NAT instead forwarded ports. For safety, it is much better.
>
>I think it is not because of a UDP timeout, but rather because of a NAT
>timeout. For this
On Tue, Apr 14, 2015 at 08:26:07AM +0200, Sebastian Kemper wrote:
> On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote:
> > I do not want set allowguest=yes. The problem is, there is no official
> > list with ip addresses of Telekom Germany. But I think all ip
> >
Am 3. Juli 2015 13:17:34 MESZ, schrieb Jerry Geis :
>alsa_card_init^[[0m: snd_pcm_open failed: Connection refused
>soundcard_init^[[0m: Problem opening alsa capture device
>
>These are the errors I get.
>
>I changed the following:
>chown -R myuser:myuser /var/log/asterisk
>chown -R myuser:myuser /
Am 5. Juni 2015 16:29:21 MESZ, schrieb Luca Bertoncello :
>Hi again!
>
>I just noticed, that my Asterisk (running on an OpenWRT-Switch) writes
>
>the logs using GMT...
>On the Switch the time is right configured and a "date" says me the
>current LOCAL time.
>
>I didn't found in logger.conf or ot
Am 31. Mai 2015 10:58:56 MESZ, schrieb Luca Bertoncello :
>-BEGIN PGP SIGNED MESSAGE-
>Hash: SHA1
>
>Guenther Boelter schrieb:
>
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA256
>>
>> On 05/31/2015 02:31 PM, Luca Bertoncello wrote:
>> > Hi list!
>> >
>> > Now all works as expected,
Hi Luca,
It's not the A number you have to look at if you want to know how a call comes
into the dialplan and then goes out again. You want do know in which context a
call arrives. That depends on things like the IP address (peer),
username/password (friend) or other things.
I suggest to read
On Fri, May 29, 2015 at 07:24:45AM +0200, Luca Bertoncello wrote:
>
> Since I think, I have a problem in my dialplan, how can I debug it?
> It would be very useful a command in Asterisk CLI to ask Asterisk what it
> would do if the number X call the number Y.
> Something like "exim -bt", if someon
t; had no changes any more. But we should be prepared for changes.
>
> You must enable the dnsmgr. If DNS resolves a new ip, the peer is
> updated.
Hello Daniel,
Thanks for the tip. I've enabled the DNS manager. Let's see how it goes.
Kind regards,
Sebastian
>
>
On Thu, Apr 02, 2015 at 11:33:38PM +0200, Daniel Heckl wrote:
> I do not want set allowguest=yes. The problem is, there is no official
> list with ip addresses of Telekom Germany. But I think all ip
> addresses comes from the ip range 217.0.0.0/13.
Hello Daniel,
Judging by the lists I found I thi
On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote:
> On 4/1/15 10:48 AM, Daniel Heckl wrote:
> > John,
> >
> > thank you four your answer. I think you have misunderstood the
> > problem. It’s about a ip address change of the sip trunk, not of my
> > asterisk server.
> You would probably benefi
On Tue, Mar 31, 2015 at 12:36:34PM +0200, Daniel Heckl wrote:
> Hello Sebastian,
>
> I had already seen this list of the hosts, but it is not active. All
> servers with which my Asterisk has been communicated are not listed.
>
> A port scan, to eventually update the list, found hundreds of server
On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:
> Hello
>
> I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom
> Germany. We have sometimes problems with incoming and outgoing calls.
> I hope I can explain it understandable.
Hello Daniel,
I'll find myself in the same si
Hello list,
I'm hoping that you could read through this mail and give me some tips
on how to improve my setup (functionality, security, really anything).
It's my first Asterisk installation and meant for simple home use.
I installed Asterisk 11 on an OpenWrt Barrier Breaker router. Currently
it's
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