Now we are using asterisk13.27.0 and ./configure --with-pjproject-bundled for building the pjsip
When I do the incoming and outgoing call for a long time, there is no UPDATE or re-INVITE happening from our asterisk server to sip trunk. Japanese sip trunk we uses requires us to send UPDATE or re-INVITE session-timers in specific intervals. So, we need this.
Is there any bug or issues with pjsip session-timer itself?
I saw the jira issue tracking
But, I can’t find the related issues.
I used this configuration in pjsip-endpoints
type = endpoint transport=transport-udp context=from-trunk ;allow=!all,g722,ulaw allow=!all,ulaw from_domain=xxx ;force_rport=yes ;rewrite_contact=yes ;rtp_symmetric=yes timers=required timers_min_se=1800 timers_sess_expires=1800 accept_multiple_sdp_answers=yes direct_media=yes connected_line_method=update direct_media_method=update direct_media_glare_mitigation=update disable_direct_media_on_nat=no force_rport=yes rewrite_contact=yes rtp_symmetric=yes