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

Karsten Wemheuer kwem at gmx.de
Mon Dec 21 12:10:22 CST 2015


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





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