Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Luca Bertoncello
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

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Sebastian Kemper
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

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Luca Bertoncello
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

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Sebastian Kemper
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...

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Sebastian Kemper
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

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-22 Thread Luca Bertoncello
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

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-21 Thread Luca Bertoncello
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

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-21 Thread Brian ::
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 > >

[asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-21 Thread 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 connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-21 Thread Karsten Wemheuer
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

Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

2015-12-21 Thread Luca Bertoncello
"Brian ::" schrieb: > sip trace? Could you please explain? I'm not a VoIP-expert... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New