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

[Yves A.]

Hi,

I know this Bug,,, at least when you´re talking about x-lite 3... quite 
annoying, but if you know it...
so no... its not the phone... tested with zoiper and 3cx ... both 
work...but the problem occurs ONLY,
as soon as I register at more than one registrar...

yves

Am 22.11.2014 um 19:19 schrieb Ron Wheeler:
> 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
>>
>> ---
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>
>


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