[asterisk-r2] Hang up after 5 to 10 minutes of call

[Leon Ramos]

Hi guys!

Thank you for doublechecking the trace! As everybody expected, there was a
timeout configured in the system.

[2016-10-07 11:28:42] NOTICE[17513]: chan_sip.c:29115 check_rtp_timeout:
Disconnecting call 'SIP/2188-000080f9' for lack of RTP activity in 301
seconds

It is related to the SIP leg of the call, and it arised only when a call is
in MOH. This is present in Complete PBX version 3.x, not sure about the 4.x
default settings.

Doing a quick search I found two interesting options:


   - *rtptimeout
   <http://www.voip-info.org/wiki/view/Asterisk+sip+rtptimeout>* = seconds
   : Terminate call if x seconds of no RTP activity when we're not on hold.
   Valid only in [general] section and *type=peer*.
   - *rtpholdtimeout
   <http://www.voip-info.org/wiki/view/Asterisk+sip+rtpholdtimeout>* =
   seconds : Terminate call if x seconds of no RTP activity when we're on hold
   (must be larger than rtptimeout). Valid only in [general] section and
   *type=peer*.


I hope this helps and thanks again!

Leon Ramos

León Ramos,

2016-10-05 19:35 GMT-05:00 Alexandre Cavalcante Alencar <
alexandre.alencar at gmail.com>:

> Check your dialplan for Set(TIMEOUT(absolute) and a Dial with L option.
>
> On Wed, Oct 5, 2016 at 7:44 PM Moises Silva <moises.silva at gmail.com>
> wrote:
>
>> On Wed, Oct 5, 2016 at 6:07 PM, Leon Ramos <leon.ramos at gmail.com> wrote:
>>
>> Hi everybody!
>>
>> Long time without writting, hope everyone is doing fine.
>>
>> I have a customer with a full E1 working "fine", calls can be received
>> and placed. The only problem is a random hangout of an ongoing call between
>> 5 to 10 minutes. It does not matter if the call is inbound or outbound.
>>
>> The service provider is using audiocodes at the other end.
>>
>>
>> Yes that's your pbx hanging up (Asterisk). You should enable Asterisk
>> debugging and see where the hangup comes from.
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-r2 mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-r2
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-r2 mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-r2
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-r2/attachments/20161007/ffcfb6bc/attachment.html>