Yves A.
2014-Nov-22 13:05 UTC
[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 gepr?ft. http://www.avast.com
Eric Wieling
2014-Nov-22 17:50 UTC
[asterisk-users] SIP call drops after 32 seconds, but only when....
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 --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren gepr?ft. 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
Rafael Visser
2014-Nov-22 18:01 UTC
[asterisk-users] SIP call drops after 32 seconds, but only when....
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 > > --- > Diese E-Mail wurde von Avast Antivirus-Software auf Viren gepr?ft. > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141122/1b7585ac/attachment.html>
Ron Wheeler
2014-Nov-22 18:19 UTC
[asterisk-users] SIP call drops after 32 seconds, but only when....
You might check your phones as well. We had this problem early on with a softphone and it was a setting in the phone that was set to hang up after 30 seconds of inactivity "in case of network disruption". For some reason it was detecting "network disruption" in every call even when the calls were proceeding normally. Unchecking this box solved the problem. It may not be related to your problem but if it is the cause, you will spend a lot of time trying to fix this in Asterisk. :-D At least I did! On the bright side, it does force people to get point in a hurry! Ron On 22/11/2014 12:50 PM, Eric Wieling wrote:> 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 > > --- > Diese E-Mail wurde von Avast Antivirus-Software auf Viren gepr?ft. > http://www.avast.com > >-- Ron Wheeler President Artifact Software Inc email: rwheeler at artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102