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 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...


But now somewhat other: yesterday evening I spoke with Telekom. They  
tried to "reset my DSL port" (whatever it means).
As result I was without Internet and phone for over an hour... Then I  
tried to call my cousin in Italy and the call was NOT dropped after 15  
minutes...


I'll try this evening again. Maybe it was a problem by Deutsche Telekom...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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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 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.

Regards,
Sebastian

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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 to the mailing list,
make sure to clean it up first (remove passwords, user names, phone
numbers, digest authentication info etc).


OK, I'll try and report to the list

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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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...

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 to the mailing list,
make sure to clean it up first (remove passwords, user names, phone
numbers, digest authentication info etc).

Regards,
Sebastian

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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 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.

Regards,
Sebastian

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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

   sip set debug 42
   sip set verbose 42

The result was in my first E-Mail...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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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 forgot to mention that...
I already have this setting:

session-refresher=uac
session-timers=refuse

> (I assume You are using chan_sip. I don't know how to disable session
> timer in pj sip).

I use chan_sip.

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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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
> > session-timers = refuse
> > into the general context of sip.conf. Reload the sip stack with "sip
> > reload".
>
> Sorry, I forgot to mention that...
> I already have this setting:
>
> session-refresher=uac
> session-timers=refuse
>
> > (I assume You are using chan_sip. I don't know how to disable session
> > timer in pj sip).
>
> I use chan_sip.
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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[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 [+3901522@default:1] Set("SIP/004935-0125", 
"newNumber=003901522") in new stack
-- Executing [+3901522@default:2] 
Verbose("SIP/004935-0125", "2,Rewrite number +3901522 to 
003901522") in new stack
  == Rewrite number +3901522 to 003901522
-- Executing [+3901522@default:3] Dial("SIP/004935-0125", 
"local/003901522") in new stack
-- Called local/003901522
-- Executing [003901522@default:1] 
Verbose("Local/003901522@default-003c;2", "2,DEFAULT") in new stack
  == DEFAULT
-- Executing [003901522@default:2] 
Set("Local/003901522@default-003c;2", "CHANNEL(musicclass)=default") in 
new stack
-- Executing [003901522@default:3] 
GotoIf("Local/003901522@default-003c;2", "0?dialrebvoice") in new stack
-- Executing [003901522@default:4] 
GotoIf("Local/003901522@default-003c;2", "0?dialluca") in new stack
-- Executing [003901522@default:5] 
GotoIf("Local/003901522@default-003c;2", "1?dialluca") in new stack
-- Goto (default,003901522,13)
-- Executing [003901522@default:13] 
Verbose("Local/003901522@default-003c;2", "2,Outgoing call for 
003901522 using pbxluca") in new stack
  == Outgoing call for 003901522 using pbxluca
-- Executing [003901522@default:14] 
Dial("Local/003901522@default-003c;2", 
"SIP/pbxluca/003901522,,RXx") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/pbxluca/003901522
-- SIP/pbxluca-0126 is ringing
-- SIP/pbxluca-0126 is making progress passing it to 
Local/003901522@default-003c;2
-- Local/003901522@default-003c;1 is ringing
-- Local/003901522@default-003c;1 is making progress passing it to 
SIP/004935-0125
-- SIP/pbxluca-0126 answered Local/003901522@default-003c;2
-- Local/003901522@default-003c;1 answered 
SIP/004935-0125
  == Spawn extension (default, 003901522, 14) exited non-zero on 
'Local/003901522@default-003c;2'
-- fixed jitterbuffer created on channel SIP/004935-0125
  == Spawn extension (default, +3901522, 3) exited non-zero on 
'SIP/004935-0125'
-- fixed jitterbuffer destroyed on channel SIP/004935-0125

My number is the 004935 and I called the 003901522.
Any idea?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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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 connection will
> be dropped:
> 
>   == Using SIP RTP CoS mark 5
> -- Executing [+3901522@default:1] Set("SIP/004935-0125", 
> "newNumber=003901522") in new stack
> -- Executing [+3901522@default:2] 
> Verbose("SIP/004935-0125", "2,Rewrite number +3901522 to 
> 003901522") in new stack
>   == Rewrite number +3901522 to 003901522
> -- Executing [+3901522@default:3] Dial("SIP/004935-0125", 
> "local/003901522") in new stack
> -- Called local/003901522
> -- Executing [003901522@default:1] 
> Verbose("Local/003901522@default-003c;2", "2,DEFAULT") in new stack
>   == DEFAULT
> -- Executing [003901522@default:2] 
> Set("Local/003901522@default-003c;2", "CHANNEL(musicclass)=default") 
> in new stack
> -- Executing [003901522@default:3] 
> GotoIf("Local/003901522@default-003c;2", "0?dialrebvoice") in new 
> stack
> -- Executing [003901522@default:4] 
> GotoIf("Local/003901522@default-003c;2", "0?dialluca") in new stack
> -- Executing [003901522@default:5] 
> GotoIf("Local/003901522@default-003c;2", "1?dialluca") in new stack
> -- Goto (default,003901522,13)
> -- Executing [003901522@default:13] 
> Verbose("Local/003901522@default-003c;2", "2,Outgoing call for 
> 003901522 using pbxluca") in new stack
>   == Outgoing call for 003901522 using pbxluca
> -- Executing [003901522@default:14] 
> Dial("Local/003901522@default-003c;2", 
> "SIP/pbxluca/003901522,,RXx") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called SIP/pbxluca/003901522
> -- SIP/pbxluca-0126 is ringing
> -- SIP/pbxluca-0126 is making progress passing it to 
> Local/003901522@default-003c;2
> -- Local/003901522@default-003c;1 is ringing
> -- Local/003901522@default-003c;1 is making progress passing it 
> to SIP/004935-0125
> -- SIP/pbxluca-0126 answered Local/003901522@default-003c;2
> -- Local/003901522@default-003c;1 answered 
> SIP/004935-0125
>   == Spawn extension (default, 003901522, 14) exited non-zero on 
> 'Local/003901522@default-003c;2'
> -- fixed jitterbuffer created on channel SIP/004935-0125
>   == Spawn extension (default, +3901522, 3) exited non-zero on 
> 'SIP/004935-0125'
> -- fixed jitterbuffer destroyed on channel SIP/004935-0125
> 
> My number is the 004935 and I called the 003901522.
> Any idea?
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
> 

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".

(I assume You are using chan_sip. I don't know how to disable session
timer in pj sip).

HTH,

Karsten



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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)

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