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Added support for PJSIP in manager scripts (requires patched Asterisk 18+)

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git-svn-id: svn://192.168.202.10@3974 3d104415-ff17-0410-8863-d5cf3c621b8a
This commit is contained in:
mattf
2026-03-27 19:29:08 +00:00
parent 8d8e366c07
commit 11b535beba
2 changed files with 19 additions and 10 deletions
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agc_2-X/trunk/bin/AST_manager_listen_AMI2.pl
+11 -5
View File
@@ -17,7 +17,7 @@
# the ADMIN_keepalive_ALL.pl script, which makes sure it is always running in a
# screen, provided that the astguiclient.conf keepalive setting "2" is set.
#
# Copyright (C) 2025 Matt Florell <vicidial@gmail.com> LICENSE: AGPLv2
# Copyright (C) 2026 Matt Florell <vicidial@gmail.com> LICENSE: AGPLv2
#
# CHANGES
# 170915-2106 - Initial version based off the orginal AST_manager_listen.pl script
@@ -26,6 +26,7 @@
# 190121-1505 - Added RA_USER_PHONE On-Hook CID to solve last RINGAGENT issues
# 210407-2009 - Added Event handlers for new manager events
# 251007-2110 - Added code for recording_dtmf_detection and recording_dtmf_muting
# 260327-1516 - Added support for PJSIP (requires patched Asterisk 18+)
#
# constants
@@ -570,16 +571,21 @@ sub handle_event
# SIPCriticalTimeout event
case "SIPCriticalTimeout" { return handle_sip_crit_timeout_event( %event_hash ); }
# PJSIPRetransmitTimeout event
case "PJSIPRetransmitTimeout" { return handle_sip_crit_timeout_event( %event_hash ); }
# PeerRegistered event
case "PeerRegistered" { return handle_peer_registered_event( %event_hash ); }
# SIPRTPDisconnect evett
# SIPRTPDisconnect event
case "SIPRTPDisconnect" { return handle_sip_rtp_disconnect_event( %event_hash ); }
# PJSIPRTPDisconnect event
case "PJSIPRTPDisconnect" { return handle_sip_rtp_disconnect_event( %event_hash ); }
# PeerStatus event
case "PeerStatus" { return handle_peer_status_event( %event_hash ); }
#case "" { return handle__event( %event_hash ); }
else { return 2; }
@@ -610,7 +616,7 @@ sub handle_sip_crit_timeout_event
}
else
{
print STDERR "SIPCriticalTimeout event does not have a Type, CallID, SeqNo, Host, or Timeout ?!!!\n";
print STDERR "SIPCriticalTimeout or PJSIPRetransmitTimeout event does not have a Type, CallID, SeqNo, Host, or Timeout ?!!!\n";
return 3;
}
@@ -669,7 +675,7 @@ sub handle_sip_rtp_disconnect_event
}
else
{
print STDERR "SIPRTPDisconnect event does not have a Channel or RTPLastRX ?!!!\n";
print STDERR "(PJ)SIPRTPDisconnect event does not have a Channel or RTPLastRX ?!!!\n";
return 3;
}
}
agc_2-X/trunk/bin/AST_manager_sip_AMI2.pl
+8 -5
View File
@@ -6,10 +6,11 @@
# connects to the Asterisk Manager Interface version 2 and inserts records into
# the vicidial_sip_log for SIPInvite and SIPResponse events.
#
# Copyright (C) 2019 Matt Florell <vicidial@gmail.com> LICENSE: AGPLv2
# Copyright (C) 2026 Matt Florell <vicidial@gmail.com> LICENSE: AGPLv2
#
# CHANGES
# 190528-1304 - Copied AST_manager_listen_AMI2.pl to make the initial version
# 260327-1509 - Added support for PJSIP (requires patched Asterisk 18+)
#
# constants
@@ -528,16 +529,18 @@ sub handle_event
# Asterisk is shutting down so should we
case "Shutdown" { return 0; }
# DTMF event
case "DTMFBegin" { return handle_dtmf_begin_event( %event_hash ); }
case "DTMFEnd" { return handle_dtmf_end_event( %event_hash ); }
# SIPInvite event
case "SIPInvite" { return handle_sip_invite ( %event_hash ); }
# PJSIPInvite event
case "PJSIPInvite" { return handle_sip_invite ( %event_hash ); }
# SIPResponse event
case "SIPResponse" { return handle_sip_response ( %event_hash ); }
# PJSIPResponse event
case "PJSIPResponse" { return handle_sip_response ( %event_hash ); }
else { return 2; }
}