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

Amit Patkar amit at avhan.com
Wed Nov 26 23:19:23 CST 2014

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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