similar to: 183 Session in Progress. Disconnecting channel for lack of RTP activity

Hello. Several months server working on asterisk 13.7 and pjproject 2.5 (installed separately). Once a day the server crashes or hangs and is familiar sores that written watchdogs. Yesterday I decided to upgrade to 13.11 and bundled pjproject (2.5.5). Solved all the problems with compilation I started asterisk several times and each time after 5-7 seconds was seg fault. So I didn't get

Hi! AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE - but how should I handle the SUBSCRIBE in the context? thanks klaus SUBSCRIBE sip:u+431234567 at foobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:u+431234567 at 11.111.11.11:39982> To:

Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1] before remote end send OK or ACK there is one way SDP, no any audio stream. PJSIP channel (option rtp_timeout) does not take this one. Isn't it a mistake? What could be workarounds? 19.12.2016 11:33, Jean Aunis ?????: > > This means the remote end was not sending any audio stream,

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief