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 <ewiel...@nyigc.com>: > 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 >
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