Luca Bertoncello
2015-Dec-21 17:52 UTC
[asterisk-users] 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)
Karsten Wemheuer
2015-Dec-21 18:10 UTC
[asterisk-users] Deutsche Telekom: calls dropped after 15 minutes
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
Luca Bertoncello
2015-Dec-21 18:56 UTC
[asterisk-users] Deutsche Telekom: calls dropped after 15 minutes
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)