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

Ron Wheeler rwheeler at artifact-software.com
Sat Nov 22 12:19:30 CST 2014

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!


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
> ---
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Ron Wheeler
Artifact Software Inc
email: rwheeler at artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

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