Segmentation Fault on inbound call

[cameron5906]

Hi there guys. I'm trying to get a phone call setup through Twilio's SIP functionality. I use my phone to call one of my Twilio numbers which is set up to then call my sipster script through SIP and once the call comes through I get a segmentation fault. Before the call, everything is fine (I saw someone was having issues with segfaults much earlier.)

Does anyone have an idea as to why this may be? I'm on pjproject 2.5... 2.4.5 yielded the same results and 2.6 just didn't seem to work at all.

`17:18:52.156 os_core_unix.c !pjlib 2.5 for POSIX initialized
17:18:52.157 sip_endpoint.c .Creating endpoint instance...
17:18:52.157 pjlib .select() I/O Queue created (0x221cd10)
17:18:52.157 sip_endpoint.c .Module "mod-msg-print" registered
17:18:52.157 sip_transport. .Transport manager created.
17:18:52.157 pjsua_core.c .PJSUA state changed: NULL --> CREATED
17:18:52.157 sip_endpoint.c .Module "mod-pjsua-log" registered
17:18:52.157 sip_endpoint.c .Module "mod-tsx-layer" registered
17:18:52.158 sip_endpoint.c .Module "mod-stateful-util" registered
17:18:52.158 sip_endpoint.c .Module "mod-ua" registered
17:18:52.158 sip_endpoint.c .Module "mod-100rel" registered
17:18:52.158 sip_endpoint.c .Module "mod-pjsua" registered
17:18:52.158 sip_endpoint.c .Module "mod-invite" registered
17:18:52.158 pjlib ..select() I/O Queue created (0x2229d88)
WARNING: no real random source present!
17:18:52.160 sip_endpoint.c .Module "mod-evsub" registered
17:18:52.161 sip_endpoint.c .Module "mod-presence" registered
17:18:52.161 sip_endpoint.c .Module "mod-mwi" registered
17:18:52.161 sip_endpoint.c .Module "mod-refer" registered
17:18:52.161 sip_endpoint.c .Module "mod-pjsua-pres" registered
17:18:52.161 sip_endpoint.c .Module "mod-pjsua-im" registered
17:18:52.161 sip_endpoint.c .Module "mod-pjsua-options" registered
17:18:52.161 pjsua_core.c .1 SIP worker threads created
17:18:52.161 pjsua_core.c .pjsua version 2.5 for Linux-4.4.0.62/x86_64/glibc-2.23 initialized
17:18:52.161 pjsua_core.c .PJSUA state changed: CREATED --> INIT
17:18:52.161 pjsua_aud.c Setting null sound device..
17:18:52.161 pjsua_aud.c .Opening null sound device..
17:18:52.161 pjsua_core.c SIP UDP socket reachable at 144.217.4.197:5060
17:18:52.162 udp0x2210a30 SIP UDP transport started, published address is 144.217.4.197:5060
17:18:52.162 pjsua_acc.c Adding account: id=sip:144.217.4.197
17:18:52.162 pjsua_acc.c .Account sip:144.217.4.197 added with id 0
17:18:52.162 pjsua_core.c PJSUA state changed: INIT --> STARTING
17:18:52.162 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
17:18:52.162 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
17:18:53.163 pjsua_aud.c !Closing sound device after idle for 1 second(s)
17:18:53.163 pjsua_aud.c .Closing null sound device..
17:19:00.255 pjsua_core.c .RX 1232 bytes Request msg INVITE/cseq=1 (rdata0x224b458) from UDP 54.172.60.0:5060:
INVITE sip:sip@144.217.4.197 SIP/2.0
Record-Route: sip:54.172.60.0:5060;lr;ftag=50198020_6772d868_027d460a-af4a-4411-94df-2751d3dbcaab
CSeq: 1 INVITE
From: "+12604378544" sip:+xxxxxxxxxx@sip.twilio.com;tag=50198020_6772d868_027d460a-af4a-4411-94df-2751d3dbcaab
To: sip:sip@144.217.4.197
Max-Forwards: 67
Date: Fri, 09 Jun 2017 21:19:00 GMT
Min-SE: 120
Call-ID: a275xxxxxxxxxxac7a52fc6d9a2@0.0.0.0
Via: SIP/2.0/UDP 54.172.60.0:5060;branch=z9hG4bK1dd5.b40f8f55.0
Via: SIP/2.0/UDP 172.18.7.44:5060;rport=5060;received=172.18.7.44;branch=z9hG4bK027d460a-af4a-4411-94df-2751d3dbcaab_6772d868_248-213380807225513847
Contact: "+12604378544" sip:+12604378544@172.18.7.44:5060;transport=udp
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
User-Agent: Twilio Gateway
X-Twilio-ApiVersion: 2010-04-01
X-Twilio-AccountSid: AC509ea3xxxxxxxxx78714f43c387c
Content-Type: application/sdp
X-Twilio-CallSid: CA02d94c0a9bf8exxxxxxxxxx34a460d76
Content-Length: 260

v=0
o=- 1787099311 1787099311 IN IP4 54.172.61.16
s=Twilio Media Gateway
c=IN IP4 54.172.61.16
t=0 0
m=audio 12000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--end msg--
17:19:00.256 pjsua_call.c .Incoming Request msg INVITE/cseq=1 (rdata0x224b458)
17:19:00.257 pjsua_media.c ..Call 0: initializing media..
17:19:00.257 pjsua_media.c ...RTP socket reachable at 144.217.4.197:4000
17:19:00.257 pjsua_media.c ...RTCP socket reachable at 144.217.4.197:4001
17:19:00.257 pjsua_media.c ...Media index 0 selected for audio call 0
Segmentation fault (core dumped)`