rendered paste body[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:11194 handle_response_invite: SIP response 407 to standard inviteTransmitting (NAT) to 217.114.78.114:5060:ACK sip:01716344354@sip.pepphone.de SIP/2.0Via: SIP/2.0/UDP 88.198.230.110:5060;branch=z9hG4bK450f0efe;rportFrom: "507430" <sip:507430@sip.pepphone.de>;tag=as083c447cTo: <sip:01716344354@sip.pepphone.de>;tag=as1a5bd0abContact: <sip:507430@88.198.230.110>Call-ID: 7749805236360fcc2f2e838d3f336f49@sip.pepphone.deCSeq: 102 ACKUser-Agent: Asterisk PBXMax-Forwards: 70Content-Length: 0---[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:10620 do_proxy_auth: Auth attempt 1 on INVITE[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5820 add_sdp: ** Our capability: 0x2 (gsm) Video flag: False[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5821 add_sdp: ** Our prefcodec: 0x2 (gsm)Audio is at 88.198.230.110 port 12072Adding codec 0x2 (gsm) to SDPAdding non-codec 0x1 (telephone-event) to SDP[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5973 add_sdp: -- Done with adding codecs to SDP[Aug 30 14:56:29] DEBUG[9111]: chan_sip.c:5974 add_sdp: XXX p->owner=='0'Segmentation fault (core dumped)