asterisk users - Dec 2015 - Deutsche Telekom: calls dropped after 15 minutes

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 [+39015222222 at default:1]
Set("SIP/00493511111111-00000125",
"newNumber=0039015222222") in new stack
    -- Executing [+39015222222 at default:2]
Verbose("SIP/00493511111111-00000125", "2,Rewrite number
+39015222222 to 0039015222222") in new stack
  == Rewrite number +39015222222 to 0039015222222
    -- Executing [+39015222222 at default:3]
Dial("SIP/00493511111111-00000125", "local/0039015222222")
in new stack
    -- Called local/0039015222222
    -- Executing [0039015222222 at default:1] Verbose("Local/0039015222222
at default-0000003c;2", "2,DEFAULT") in new stack
  == DEFAULT
    -- Executing [0039015222222 at default:2] Set("Local/0039015222222 at
default-0000003c;2", "CHANNEL(musicclass)=default") in new stack
    -- Executing [0039015222222 at default:3] GotoIf("Local/0039015222222
at default-0000003c;2", "0?dialrebvoice") in new stack
    -- Executing [0039015222222 at default:4] GotoIf("Local/0039015222222
at default-0000003c;2", "0?dialluca") in new stack
    -- Executing [0039015222222 at default:5] GotoIf("Local/0039015222222
at default-0000003c;2", "1?dialluca") in new stack
    -- Goto (default,0039015222222,13)
    -- Executing [0039015222222 at default:13] Verbose("Local/0039015222222
at default-0000003c;2", "2,Outgoing call for 0039015222222 using
pbxluca") in new stack
  == Outgoing call for 0039015222222 using pbxluca
    -- Executing [0039015222222 at default:14] Dial("Local/0039015222222 at
default-0000003c;2", "SIP/pbxluca/0039015222222,,RXx") in new
stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/pbxluca/0039015222222
    -- SIP/pbxluca-00000126 is ringing
    -- SIP/pbxluca-00000126 is making progress passing it to Local/0039015222222
at default-0000003c;2
    -- Local/0039015222222 at default-0000003c;1 is ringing
    -- Local/0039015222222 at default-0000003c;1 is making progress passing it
to SIP/00493511111111-00000125
    -- SIP/pbxluca-00000126 answered Local/0039015222222 at default-0000003c;2
    -- Local/0039015222222 at default-0000003c;1 answered
SIP/00493511111111-00000125
  == Spawn extension (default, 0039015222222, 14) exited non-zero on
'Local/0039015222222 at default-0000003c;2'
    -- fixed jitterbuffer created on channel SIP/00493511111111-00000125
  == Spawn extension (default, +39015222222, 3) exited non-zero on
'SIP/00493511111111-00000125'
    -- fixed jitterbuffer destroyed on channel SIP/00493511111111-00000125

My number is the 00493511111111 and I called the 0039015222222.
Any idea?

Thanks
Luca Bertoncello
(lucabert at lucabert.de)

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 [+39015222222 at default:1]
Set("SIP/00493511111111-00000125",
"newNumber=0039015222222") in new stack
>     -- Executing [+39015222222 at default:2]
Verbose("SIP/00493511111111-00000125", "2,Rewrite number
+39015222222 to 0039015222222") in new stack
>   == Rewrite number +39015222222 to 0039015222222
>     -- Executing [+39015222222 at default:3]
Dial("SIP/00493511111111-00000125", "local/0039015222222")
in new stack
>     -- Called local/0039015222222
>     -- Executing [0039015222222 at default:1]
Verbose("Local/0039015222222 at default-0000003c;2",
"2,DEFAULT") in new stack
>   == DEFAULT
>     -- Executing [0039015222222 at default:2] Set("Local/0039015222222
at default-0000003c;2", "CHANNEL(musicclass)=default") in new
stack
>     -- Executing [0039015222222 at default:3]
GotoIf("Local/0039015222222 at default-0000003c;2",
"0?dialrebvoice") in new stack
>     -- Executing [0039015222222 at default:4]
GotoIf("Local/0039015222222 at default-0000003c;2",
"0?dialluca") in new stack
>     -- Executing [0039015222222 at default:5]
GotoIf("Local/0039015222222 at default-0000003c;2",
"1?dialluca") in new stack
>     -- Goto (default,0039015222222,13)
>     -- Executing [0039015222222 at default:13]
Verbose("Local/0039015222222 at default-0000003c;2", "2,Outgoing
call for 0039015222222 using pbxluca") in new stack
>   == Outgoing call for 0039015222222 using pbxluca
>     -- Executing [0039015222222 at default:14]
Dial("Local/0039015222222 at default-0000003c;2",
"SIP/pbxluca/0039015222222,,RXx") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/pbxluca/0039015222222
>     -- SIP/pbxluca-00000126 is ringing
>     -- SIP/pbxluca-00000126 is making progress passing it to
Local/0039015222222 at default-0000003c;2
>     -- Local/0039015222222 at default-0000003c;1 is ringing
>     -- Local/0039015222222 at default-0000003c;1 is making progress passing
it to SIP/00493511111111-00000125
>     -- SIP/pbxluca-00000126 answered Local/0039015222222 at
default-0000003c;2
>     -- Local/0039015222222 at default-0000003c;1 answered
SIP/00493511111111-00000125
>   == Spawn extension (default, 0039015222222, 14) exited non-zero on
'Local/0039015222222 at default-0000003c;2'
>     -- fixed jitterbuffer created on channel SIP/00493511111111-00000125
>   == Spawn extension (default, +39015222222, 3) exited non-zero on
'SIP/00493511111111-00000125'
>     -- fixed jitterbuffer destroyed on channel SIP/00493511111111-00000125
> 
> My number is the 00493511111111 and I called the 0039015222222.
> Any idea?
> 
> Thanks
> Luca Bertoncello
> (lucabert at 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

Karsten Wemheuer <kwem at gmx.de> 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
(lucabert at lucabert.de)