Why am I booted out of Asterisk?
thufir@aws:
thufir@aws: sudo asterisk -rvvvv
Asterisk 13.1.0~dfsg-1.1ubuntu4.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.1.0~dfsg-1.1ubuntu4.1 currently running on ip-172-31-14-19 (pid = 15999)
== Parsing '/etc/asterisk/pjsip.conf': Found
== Parsing '/etc/asterisk/pjsip.conf': Found
== Manager registered action PJSIPShowSubscriptionsInbound
== Manager registered action PJSIPShowSubscriptionsOutbound
== Manager registered action PJSIPShowResourceLists
== res_pjsip_pubsub.so => (PJSIP event resource)
Loading res_format_attr_silk.so.
== Registered format interface for codec 'silk'
== res_format_attr_silk.so => (SILK Format Attribute Module)
Loading res_format_attr_opus.so.
== Registered format interface for codec 'opus'
== res_format_attr_opus.so => (Opus Format Attribute Module)
Loading res_pjsip_exten_state.so.
== res_pjsip_exten_state.so => (PJSIP Extension State Notifications)
Loading res_pjsip_outbound_authenticator_digest.so.
== res_pjsip_outbound_authenticator_digest.so => (PJSIP authentication resource)
Loading chan_motif.so.
== Parsing '/etc/asterisk/motif.conf': Found
== Registered RTP glue 'Motif'
== Registered channel type 'Motif' (Motif Jingle Channel Driver)
== chan_motif.so => (Motif Jingle Channel Driver)
Loading chan_sip.so.
SIP channel loading...
== Parsing '/etc/asterisk/sip.conf': Found
== Parsing '/etc/asterisk/users.conf': Found
== SIP Listening on 0.0.0.0:5060
== Using SIP CoS mark 4
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Message technology 'sip' registered.
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered RTP glue 'SIP'
== Registered application 'SIPDtmfMode'
== Registered application 'SIPAddHeader'
== Registered application 'SIPRemoveHeader'
== Registered custom function 'SIP_HEADER'
== Registered custom function 'SIPPEER'
== Registered custom function 'CHECKSIPDOMAIN'
== Manager registered action SIPpeers
== Manager registered action SIPshowpeer
== Manager registered action SIPqualifypeer
== Manager registered action SIPshowregistry
== Manager registered action SIPnotify
== Manager registered action SIPpeerstatus
== WebSocket registered sub-protocol 'sip'
== chan_sip.so => (Session Initiation Protocol (SIP))
Loading res_pjsip_sdp_rtp.so.
[Nov 13 20:29:39] ERROR[15999]: astobj2.c:116 INTERNAL_OBJ: user_data is NULL
[Nov 13 20:29:39] ERROR[15999]: astobj2.c:116 INTERNAL_OBJ: user_data is NULL
[Nov 13 20:29:39] ERROR[15999]: astobj2.c:116 INTERNAL_OBJ: user_data is NULL
[Nov 13 20:29:39] ERROR[15999]: res_pjsip_sdp_rtp.c:1330 load_module: Unable to register SDP handler for audio stream type
[Nov 13 20:29:39] ERROR[15999]: astobj2.c:116 INTERNAL_OBJ: user_data is NULL
[Nov 13 20:29:39] ERROR[15999]: astobj2.c:116 INTERNAL_OBJ: user_data is NULL
ip-172-31-14-19*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
thufir@aws:
this is on an Amazon EC2 instance.