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@default:1] Set(“SIP/00493511111111-00000125”, “newNumber 39015222222”) 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)
sip trace?
“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)
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
Zitat von Sebastian Kemper:
OK, I’ll try and report to the list
Thanks Luca Bertoncello
(lucabert@lucabert.de)