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

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lucabert at lucabert.de
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Posted: Mon Dec 21, 2015 12:52 pm Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

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=0039015222222") 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)=default") in new stack -- Executing [0039015222222@default:3] GotoIf("Local/0039015222222@default-0000003c;2", "0?dialrebvoice") in new stack -- Executing [0039015222222@default:4] GotoIf("Local/0039015222222@default-0000003c;2", "0?dialluca") in new stack -- Executing [0039015222222@default:5] GotoIf("Local/0039015222222@default-0000003c;2", "1?dialluca") in new stack -- Goto (default,0039015222222,13) -- Executing [0039015222222@default:13] Verbose("Local/0039015222222@default-0000003c;2", "2,Outgoing call for 0039015222222 using pbxluca") in new stack == Outgoing call for 0039015222222 using pbxluca -- Executing [0039015222222@default:14] Dial("Local/0039015222222@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@default-0000003c;2 -- Local/0039015222222@default-0000003c;1 is ringing -- Local/0039015222222@default-0000003c;1 is making progress passing it to SIP/00493511111111-00000125 -- SIP/pbxluca-00000126 answered Local/0039015222222@default-0000003c;2 -- Local/0039015222222@default-0000003c;1 answered SIP/00493511111111-00000125 == Spawn extension (default, 0039015222222, 14) exited non-zero on 'Local/0039015222222@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@lucabert.de) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

kwem at gmx.de
Guest

Posted: Mon Dec 21, 2015 1:11 pm Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

Hi Luca,

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

Quote:

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=0039015222222") 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)=default") in new stack
-- Executing [0039015222222@default:3] GotoIf("Local/0039015222222@default-0000003c;2", "0?dialrebvoice") in new stack
-- Executing [0039015222222@default:4] GotoIf("Local/0039015222222@default-0000003c;2", "0?dialluca") in new stack
-- Executing [0039015222222@default:5] GotoIf("Local/0039015222222@default-0000003c;2", "1?dialluca") in new stack
-- Goto (default,0039015222222,13)
-- Executing [0039015222222@default:13] Verbose("Local/0039015222222@default-0000003c;2", "2,Outgoing call for 0039015222222 using pbxluca") in new stack
== Outgoing call for 0039015222222 using pbxluca
-- Executing [0039015222222@default:14] Dial("Local/0039015222222@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@default-0000003c;2
-- Local/0039015222222@default-0000003c;1 is ringing
-- Local/0039015222222@default-0000003c;1 is making progress passing it to SIP/00493511111111-00000125
-- SIP/pbxluca-00000126 answered Local/0039015222222@default-0000003c;2
-- Local/0039015222222@default-0000003c;1 answered SIP/00493511111111-00000125
== Spawn extension (default, 0039015222222, 14) exited non-zero on 'Local/0039015222222@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@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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lucabert at lucabert.de
Guest

Posted: Mon Dec 21, 2015 1:57 pm Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

Karsten Wemheuer <kwem@gmx.de> schrieb:

Hi Karsten!

Quote:

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

session-refresher=uac
session-timers=refuse

Quote:

(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@lucabert.de)

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bc at iptel.co
Guest

Posted: Mon Dec 21, 2015 5:02 pm Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

sip trace?

On Mon, Dec 21, 2015 at 6:56 PM, Luca Bertoncello <lucabert@lucabert.de (lucabert@lucabert.de)> wrote:

Quote:

Karsten Wemheuer <kwem@gmx.de (kwem@gmx.de)> schrieb:

Hi Karsten!

Quote:

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

session-refresher=uac
session-timers=refuse

Quote:

(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@lucabert.de (lucabert@lucabert.de))

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lucabert at lucabert.de
Guest

Posted: Tue Dec 22, 2015 1:20 am Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

"Brian ::" <bc@iptel.co> schrieb:

Quote:

sip trace?

Could you please explain? I'm not a VoIP-expert...

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

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sebastian_ml at gmx.net
Guest

Posted: Tue Dec 22, 2015 4:28 am Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

On Tue, Dec 22, 2015 at 07:19:47AM +0100, Luca Bertoncello wrote:

Quote:

"Brian ::" <bc@iptel.co> schrieb:

Quote:

sip trace?

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

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lucabert at lucabert.de
Guest

Posted: Tue Dec 22, 2015 4:31 am Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

Zitat von Sebastian Kemper <sebastian_ml@gmx.net>:

Hi Sebastian

Quote:

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.

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)

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sebastian_ml at gmx.net
Guest

Posted: Tue Dec 22, 2015 4:35 am Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

On Tue, Dec 22, 2015 at 09:30:52AM +0000, Luca Bertoncello wrote:

Quote:

Zitat von Sebastian Kemper <sebastian_ml@gmx.net>:

Hi Sebastian

<snip>

Quote:

I tried with

sip set debug 42
sip set verbose 42

The result was in my first E-Mail...

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

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lucabert at lucabert.de
Guest

Posted: Tue Dec 22, 2015 4:42 am Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

Zitat von Sebastian Kemper <sebastian_ml@gmx.net>:

Quote:

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.

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)

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sebastian_ml at gmx.net
Guest

Posted: Tue Dec 22, 2015 4:50 am Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

On Tue, Dec 22, 2015 at 09:42:04AM +0000, Luca Bertoncello wrote:

Quote:

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

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lucabert at lucabert.de
Guest

Posted: Tue Dec 22, 2015 4:53 am Post subject: [asterisk-users] Deutsche Telekom: calls dropped after 15 mi

Zitat von Sebastian Kemper <sebastian_ml@gmx.net>:

Quote:

OK, I'll try and report to the list

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

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