[asterisk-users] SIP calls dropping at 15 minutes

Steve Edwards asterisk.org at sedwards.com
Sat Nov 21 14:10:28 CST 2015

> On 11/20/15 11:13 AM, Steve Edwards wrote:

>> I have a problem where SIP calls from some providers are dropping at 15 
>> minutes.
>> The topology is: Client sends calls to a host running OpenSIPS, 
>> OpenSIPS sends calls to an Asterisk server.

>> 1) Is a 'ds_select_dst()' followed by a 'forward()' the right way to 
>> route calls in OpenSIPS? It works most of the time.
>> 2) Can (or should) I configure Asterisk to not send the INVITE at 15 
>> minutes?

On Sat, 21 Nov 2015, Andres wrote:

> Looks like session timers are kicking in and a Re-Invite is being sent. 
> I would disable them in sip.conf and try again:
> session-timers=refuse
> http://doxygen.asterisk.org/trunk/sip_session_timers.html

>> 3) Should OpenSIPS be responding differently to the INVITE at 15 
>> minutes?

This appears to work, but it feels wrong. Shouldn't I be configuring 
Asterisk or OpenSIPS  to respond or receive the re-invite correctly?

Thanks in advance,
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST

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