[asterisk-users] SIP call drops after 32 seconds, but only when....

[Yves A.]

Hi,

the useragents nothing to do with the problem... i tried numeric, alpha 
and alphanumeric... no difference.
they work all.... as long as I only use ONE registrar...
as soon as I register at more than one registrar... the line drops after 
32 seconds.... really strange.

yves

Am 22.11.2014 um 19:01 schrieb Rafael Visser:
>
> Hi Yves..
> This may be silly... but what is the useragent of your sip configuration?
> In the case that useragent has some special characters like "(.", 
> please remove it and tell us if there is any change!!.
> Regards.
> rv
>
>
> 2014-11-22 14:50 GMT-03:00 Eric Wieling <EWieling at nyigc.com 
> <mailto:EWieling at nyigc.com>>:
>
>     Try setting directmedia=no in sip.conf.
>
>     -----Original Message-----
>     From: asterisk-users-bounces at lists.digium.com
>     <mailto:asterisk-users-bounces at lists.digium.com>
>     [mailto:asterisk-users-bounces at lists.digium.com
>     <mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Yves A.
>     Sent: Saturday, November 22, 2014 8:06 AM
>     To: asterisk-users at lists.digium.com
>     <mailto:asterisk-users at lists.digium.com>
>     Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but
>     only when....
>
>     Am 22.11.2014 um 12:51 schrieb Andreas Sikkema:
>     >> but as soon as I configure another sip registration on another
>     server,
>     >> outgoing
>     >> calls  drop after 32 seconds.
>     > Are both your servers behind the same NAT router?
>     >
>     thanks for taking part...
>
>     I don´t know...
>     one is
>
>     siptrunk.ovh.net <http://siptrunk.ovh.net>
>
>     and the other one is
>
>     sip.ovh.fr <http://sip.ovh.fr>
>
>     how can i determine and how could that affect... I mean... why do they
>     interfere at all?
>
>     thanks,
>     yves
>
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