asterisk users - Nov 2014 - SIP call drops after 32 seconds, but only when....

hi,

I have a really strange problem which is driving me crazy for days now.

If I register my asterisk (tried all versions from 1.6 up to 13.x) with 
one sip registrar,
everything works... calls go out and call come in... no 32 seconds limit.

but as soon as I configure another sip registration on another server, 
outgoing
calls  drop after 32 seconds.

as far as I know, there is no firewall in between...

I tried to "work around" this by increasing the settings for
"timerb"...
but I
realized that asterisk does not care at all, what I set this value to...
"sip show settings" always gives me 32000ms, and it does not make any
difference if I configure timerb in the general context or in the phone 
context...

any ideas?

thanks,
yves

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> 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?

-- 
Andreas Sikkema

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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On 22.11.2014, at 13:40, Yves A. <yves030 at gmx.de>
wrote:
> I have a really strange problem which is driving me crazy for days now.
> 
> If I register my asterisk (tried all versions from 1.6 up to 13.x) with one
sip registrar,
> everything works... calls go out and call come in... no 32 seconds limit.
> 
> but as soon as I configure another sip registration on another server,
outgoing
> calls  drop after 32 seconds.
Do a 'sip set debug on' and see what they (Asterisk and the registrar)
are talking about just before the call drops.

-- 

marie

Call drop after 30+sec happens if RTP is not received by asterisk for 30 
seconds (RTP Timeout).
You should look for media IP address in SDP. If there is firewall, apart 
from port UDP/5060, you also need to open port UDP/10000-UDP/20000 
(standard RTP ports)
You should try with RTP debug. It should show bidirectional traffic. If 
not, you surely have an issue with media IP or ports.

*Thanks & Regards,*
Amit Patkar


On 11/27/2014 10:01 AM, Marie Fischer wrote:
> On 22.11.2014, at 13:40, Yves A. <yves030 at gmx.de> wrote:
>> I have a really strange problem which is driving me crazy for days now.
>>
>> If I register my asterisk (tried all versions from 1.6 up to 13.x) with
one sip registrar,
>> everything works... calls go out and call come in... no 32 seconds
limit.
>>
>> but as soon as I configure another sip registration on another server,
outgoing
>> calls  drop after 32 seconds.
> Do a 'sip set debug on' and see what they (Asterisk and the
registrar) are talking about just before the call drops.
>

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