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

[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>:

> 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] On Behalf Of Yves A.
> Sent: Saturday, November 22, 2014 8:06 AM
> To: 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
>
> and the other one is
>
> 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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