Try setting directmedia=no in sip.conf. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@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 --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users