Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes

2014-01-19 Thread David Cunningham
If it's a reinvite problem, check the domain in the request URI. Might be
something it shouldn't be, eg something to do with the VPN.



On 11 September 2013 05:28,  wrote:

> Some providers send a reinvite after 15 min and if asterisk doesn't
> respond will disconnect the call
> Maybe playaround with canreinvite
>
> --Original Message--
> From: Jeremy Kister
> Sender: asterisk-users-boun...@lists.digium.com
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes
> Sent: Sep 10, 2013 10:23 PM
>
> On 9/10/2013 7:05 AM, Administrator TOOTAI wrote:
> > I face the subject strange behavior: calls arre dropped after 15 minutes
> > on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk
>
> Just for kicks, I would disable session-timers to see if the problem
> goes away.  in the general section and/or each peer in sip.conf:
> session-timers=refuse
>
>
>
> --
>
> Jeremy Kister
> http://jeremy.kister.net./
>
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Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes

2013-09-10 Thread isrlgb
Some providers send a reinvite after 15 min and if asterisk doesn't respond 
will disconnect the call
Maybe playaround with canreinvite

--Original Message--
From: Jeremy Kister
Sender: asterisk-users-boun...@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes
Sent: Sep 10, 2013 10:23 PM

On 9/10/2013 7:05 AM, Administrator TOOTAI wrote:
> I face the subject strange behavior: calls arre dropped after 15 minutes
> on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk

Just for kicks, I would disable session-timers to see if the problem 
goes away.  in the general section and/or each peer in sip.conf:
session-timers=refuse



-- 

Jeremy Kister
http://jeremy.kister.net./

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Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes

2013-09-10 Thread Jeremy Kister

On 9/10/2013 7:05 AM, Administrator TOOTAI wrote:

I face the subject strange behavior: calls arre dropped after 15 minutes
on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk


Just for kicks, I would disable session-timers to see if the problem 
goes away.  in the general section and/or each peer in sip.conf:

session-timers=refuse



--

Jeremy Kister
http://jeremy.kister.net./

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Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes

2013-09-10 Thread Asghar Mohammad
hi,
it seems your vpn connection drop.
is you vpn on WiFi of any other high latency network?


On Tue, Sep 10, 2013 at 1:05 PM, Administrator TOOTAI wrote:

> Hi all,
>
> I face the subject strange behavior: calls arre dropped after 15 minutes
> on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk
> through OpenVPN seems to have the problem.
>
> From CDR, I see for 3 calls from this morning I'm aware of, that asterisk
> hangup after respectively 899s 894s 898s
>
> In logs I see
>
> WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission
> 522eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See
> https://wiki.a
> Packet timed out after 6401ms with no response (or 6399ms or 6401ms)
>
> Qualify freq being 6 ms for the peers.
>
> For the SIP peers I see
>
>   Sess-Timers  : Accept
>   Sess-Refresh : uas
>   Sess-Expires : 1800 secs
>   Min-Sess : 90 secs
>
> I tried to use originate for session-timer in global SIP conf, no changes.
>
> Any hint about this matter would be appreciate.
>
> --
> Daniel
>
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[asterisk-users] Asterisk 1.8 drop calls after 15 minutes

2013-09-10 Thread Administrator TOOTAI

Hi all,

I face the subject strange behavior: calls arre dropped after 15 minutes 
on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk 
through OpenVPN seems to have the problem.


From CDR, I see for 3 calls from this morning I'm aware of, that 
asterisk hangup after respectively 899s 894s 898s


In logs I see

WARNING[8213] chan_sip.c: Retransmission timeout reached on transmission 
522eec628683-uy8xshd6wc21 for seqno 102 (Critical Request) -- See 
https://wiki.a

Packet timed out after 6401ms with no response (or 6399ms or 6401ms)

Qualify freq being 6 ms for the peers.

For the SIP peers I see

  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs

I tried to use originate for session-timer in global SIP conf, no changes.

Any hint about this matter would be appreciate.

--
Daniel

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