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,
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Steve Edwards [email protected] Voice: +1-760-468-3867 PST
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