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

[Eric Wieling]

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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