Zitat von Sebastian Kemper :
I don't remember seeing anything looking like a SIP trace in your first
mail. Try
sip set debug on
instead of
sip set debug 42
I don't think there's a sip debugging level like 42 in Asterisk. You can
either switch it on or off.
Is it not
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 remember seeing anything
Zitat von Sebastian Kemper :
No, that's not it. SIP debugging should show you all the SIP messages
like INVITEs, ACKs and the likes. See this link:
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Big fat warning: If you want to paste a SIP trace
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...
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
Zitat von Sebastian Kemper :
Hi Sebastian
Brian suggests to check the SIP traces. You can either enable SIP
debugging in Asterisk like so:
sip set debug on
Or you could run tcpdump and capture the SIP traffic.
The first option is probably the easiest.
I tried with
Karsten Wemheuer schrieb:
Hi Karsten!
> the timeout value of 15 minutes directs me to an issue with session
> timer. Try to refuse them by putting the line
> session-timers = refuse
> into the general context of sip.conf. Reload the sip stack with "sip
> reload".
Sorry, I
sip trace?
On Mon, Dec 21, 2015 at 6:56 PM, Luca Bertoncello
wrote:
> Karsten Wemheuer schrieb:
>
> Hi Karsten!
>
> > the timeout value of 15 minutes directs me to an issue with session
> > timer. Try to refuse them by putting the line
> >
Hi list!
My Problem: all calls to international numbers will be dropped after exactly
15 minutes...
I have a VoIP-account by Deutsche Telekom.
This is what I see when I call someone (my parents) and the connection will
be dropped:
== Using SIP RTP CoS mark 5
-- Executing
Hi Luca,
Am Montag, den 21.12.2015, 18:52 +0100 schrieb Luca Bertoncello:
> Hi list!
>
> My Problem: all calls to international numbers will be dropped after exactly
> 15 minutes...
> I have a VoIP-account by Deutsche Telekom.
> This is what I see when I call someone (my parents) and the
"Brian ::" schrieb:
> sip trace?
Could you please explain? I'm not a VoIP-expert...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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