Deutsche Telekom: Calls Dropped After 15 Minutes

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Asterisk Users 10 Comments

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@default:1] Set(“SIP/00493511111111-00000125”, “newNumber39015222222”) in new stack
— Executing [+39015222222@default:2] Verbose(“SIP/00493511111111-00000125”, “2,Rewrite number +39015222222 to 0039015222222”) in new stack
== Rewrite number +39015222222 to 0039015222222
— Executing [+39015222222@default:3] Dial(“SIP/00493511111111-00000125”, “local/0039015222222”) in new stack
— Called local/0039015222222
— Executing [0039015222222@default:1] Verbose(“Local/0039015222222@default-0000003c;2”, “2,DEFAULT”) in new stack
== DEFAULT
— Executing [0039015222222@default:2] Set(“Local/0039015222222@default-0000003c;2”, “CHANNEL(musicclass)

10 thoughts on - Deutsche Telekom: Calls Dropped After 15 Minutes

  • Hi Luca,

    Am Montag, den 21.12.2015, 18:52 +0100 schrieb Luca Bertoncello:

    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 schrieb:

    Hi Karsten!

    Sorry, I forgot to mention that… I already have this setting:

    session-refresher=uac session-timers=refuse

    I use chan_sip.

    Thanks Luca Bertoncello
    (lucabert@lucabert.de)

  • “Brian ::” schrieb:

    Could you please explain? I’m not a VoIP-expert…

    Thanks Luca Bertoncello
    (lucabert@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

  • 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…

    Thanks Luca Bertoncello
    (lucabert@lucabert.de)

  • 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

  • Zitat von Sebastian Kemper :

    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
    (lucabert@lucabert.de)

  • Zitat von Sebastian Kemper :

    OK, I’ll try and report to the list

    Thanks Luca Bertoncello
    (lucabert@lucabert.de)