[asterisk-bugs] [JIRA] (ASTERISK-22380) Inbound SIP call to a valid extension results in segfault in multicast_rtp_new at res_rtp_multicast.c

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22380?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=209567#comment-209567 ] 

Rusty Newton commented on ASTERISK-22380:
-----------------------------------------

Specifying rtpengine 

{noformat}
[6001]
type=endpoint
context=from-internal
disallow=all
allow=ulaw
transport=transport-udp
auth=6001
aors=6001
rtpengine=asterisk
{noformat}

Results in it not being recognized and then failing endpoint configuration totally.

{noformat}
  == Parsing '/etc/asterisk/pjsip.conf': Found
19:20:46.695   udp0x3e49e40 !SIP UDP transport started, published address is 192.168.1.55:5060
  == Parsing '/etc/asterisk/pjsip.conf': Found
[Aug 24 19:20:46] ERROR[6317]: config_options.c:681 aco_process_var: Could not find option suitable for category '6001' named 'rtpengine' at line 18 of 
[Aug 24 19:20:46] ERROR[6317]: config_options.c:681 aco_process_var: Could not find option suitable for category '6002' named 'rtpengine' at line 38 of 
  == Parsing '/etc/asterisk/pjsip.conf': Found
  == Parsing '/etc/asterisk/pjsip.conf': Found
{noformat}