Some additional debugging after trying to simply run an agitest.php file with the following details:
agitest.php:
#!/usr/local/php5.5/bin/php
<?php
require_once('/var/lib/asterisk/agi-bin/phpagi-asmanager.php');
$agi = new AGI();
$agi->answer();
$agi->say_phonetic("Hello World");
$agi->hangup();
?>
extensions.conf:
[default]
exten => _X.,1,Answer()
exten => _X.,2,AGI(agitest.php)
exten => _X.,3,Hangup()
call logs using tail -f /var/log/asterisk/full:
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting '10.10.7.84' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host '10.10.7.84' and port ''.
[Jun 20 21:15:34] WARNING[1462][C-0000001e] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting '10.10.7.84' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host '10.10.7.84' and port ''.
[Jun 20 21:15:34] WARNING[1462][C-0000001e] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting '10.10.7.84' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host '10.10.7.84' and port ''.
[Jun 20 21:15:34] WARNING[1462][C-0000001e] sip/config_parser.c: nat=yes is deprecated, use nat=force_rport,comedia instead
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Not an IPv4 nor IPv6 address, cannot get port.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Not an IPv4 nor IPv6 address, cannot set port.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: -REALTIME- loading peer from database to memory. Name: 1*7403. Peer objects: 0
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x55b4e86a0620'
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e86a0620) RTP allocated port 16490
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: RTP instance '0x55b4e86a0620' is setup and ready to go
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x55b4e91d52a0'
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e91d52a0) RTP allocated port 10796
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: RTP instance '0x55b4e91d52a0' is setup and ready to go
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting 'ehtestvm' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host 'ehtestvm' and port ''.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: () RTCP setup on RTP instance
[Jun 20 21:15:34] VERBOSE[1462][C-0000001e] netsock2.c: Using SIP VIDEO CoS mark 6
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting 'ehtestvm' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host 'ehtestvm' and port ''.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: () RTCP setup on RTP instance
[Jun 20 21:15:34] VERBOSE[1462][C-0000001e] netsock2.c: Using SIP RTP TOS bits 184
[Jun 20 21:15:34] VERBOSE[1462][C-0000001e] netsock2.c: Using SIP RTP CoS mark 5
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Setting NAT on RTP to On
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Setting NAT on VRTP to On
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing session-level SDP o=sdp_admin 65546373 252805455 IN IP4 10.10.7.84... OK.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED OR FAILED.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting '10.10.7.84' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host '10.10.7.84' and port ''.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing session-level SDP c=IN IP4 10.10.7.84... OK.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Setting tx payload type 9 based on m type on 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Setting tx payload type 18 based on m type on 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Setting tx payload type 8 based on m type on 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Unsetting payload 9 on 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/16000... UNSUPPORTED OR FAILED.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Crossover copying tx to rx payload mapping 0 (0x55b4e918ddd8) from 0x7f58b2fd3310 to 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Crossover copying tx to rx payload mapping 8 (0x55b4e91da058) from 0x7f58b2fd3310 to 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Crossover copying tx to rx payload mapping 18 (0x55b4e930aea8) from 0x7f58b2fd3310 to 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Crossover copying tx to rx payload mapping 101 (0x55b4e930b6e8) from 0x7f58b2fd3310 to 0x7f58b2fd3310
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] acl.c: For destination '10.10.7.84', our source address is '10.10.7.31'.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e86a0620) RTCP setting address on RTP instance
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying rx payload mapping 0 (0x55b4e918ddd8) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying rx payload mapping 8 (0x55b4e91da058) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying rx payload mapping 18 (0x55b4e930aea8) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying rx payload mapping 101 (0x55b4e930b6e8) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying tx payload mapping 0 (0x55b4e918ddd8) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying tx payload mapping 8 (0x55b4e91da058) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying tx payload mapping 18 (0x55b4e930aea8) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Copying tx payload mapping 101 (0x55b4e930b6e8) from 0x7f58b2fd3310 to 0x55b4e86a07f8
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e86a0620) RTCP ignoring duplicate property
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e91d52a0) DTLS srtp - stopped timeout timer'
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e91d52a0) DTLS srtp - stopped timeout timer'
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: () RTP Stop
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: We're settling with these formats: (alaw|ulaw)
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Checking SIP call limits for device 1*7403
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Updating call counter for incoming call
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting '10.10.7.31' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host '10.10.7.31' and port ''.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: Splitting '10.10.7.31:5060' into...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] netsock2.c: ...host '10.10.7.31' and port ''.
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] channel_internal_api.c: <initializing>: Formats: (none)
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] channel_internal_api.c: Channel is being initialized or destroyed
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] stasis.c: Creating topic. name: channel:ehtestvm-1718903734.31, detail:
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] stasis.c: Topic 'channel:ehtestvm-1718903734.31': 0x55b4e91903e0 created
[Jun 20 21:15:34] DEBUG[1294] threadpool.c: Increasing threadpool stasis/pool's size by 1
[Jun 20 21:15:34] DEBUG[1311] devicestate.c: No provider found, checking channel drivers for SIP - 1*7403
[Jun 20 21:15:34] DEBUG[1311] chan_sip.c: Checking device state for peer 1*7403
[Jun 20 21:15:34] DEBUG[1311] devicestate.c: Changing state for SIP/1*7403 - state 1 (Not in use)
[Jun 20 21:15:34] ERROR[1436] cdr.c: Unable to find CDR for channel SIP/1*7403-0000001d
[Jun 20 21:15:34] ERROR[1436] cdr.c: Unable to find CDR for channel SIP/1*7403-0000001d
[Jun 20 21:15:34] ERROR[1436] cdr.c: Unable to find CDR for channel SIP/1*7403-0000001d
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] channel.c: Channel 0x55b4e91f3a60 'SIP/1*7403-0000001d' allocated
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] channel_internal_api.c: SIP/1*7403-0000001d: Formats: (alaw)
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] channel_internal_api.c: New topology set
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: *** Our native formats are (alaw)
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: *** Joint capabilities are (alaw|ulaw)
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: *** Our capabilities are (alaw|h261|h263|h263p|h264|ulaw)
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e91d52a0) DTLS stop
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e91d52a0) DTLS srtp - stopped timeout timer'
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] res_rtp_asterisk.c: (0x55b4e91d52a0) DTLS srtp - stopped timeout timer'
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] rtp_engine.c: Destroyed RTP instance '0x55b4e91d52a0'
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: This channel will not be able to handle video.
[Jun 20 21:15:34] VERBOSE[1436] cdr.c: 0x7f58c8075890 - Created CDR for channel SIP/1*7403-0000001d
[Jun 20 21:15:34] VERBOSE[1436] cdr.c: 0x7f58c8075890 - Transitioning CDR for SIP/1*7403-0000001d from state NONE to Single
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: SIP/1*7403-0000001d: New call is still down.... Trying...
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.10.7.84:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Allocating new SIP dialog for 110b57ab70587ede727cdd68028a3b6d@10.10.7.31:5060 - OPTIONS (No RTP)
[Jun 20 21:15:34] DEBUG[1462] acl.c: For destination '10.10.7.84', our source address is '10.10.7.31'.
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.10.7.31:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: SIP call-id changed from '110b57ab70587ede727cdd68028a3b6d@10.10.7.31:5060' to '63921f2b54cb7aa33b9a62f968fe2225@10.10.7.31:5060'
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Initializing initreq for method OPTIONS - callid 63921f2b54cb7aa33b9a62f968fe2225@10.10.7.31:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.10.7.84:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: = Looking for Call ID: 66814a9578fab3d5100e465e613d4661@10.10.7.31:5060 (Checking To) --From tag as6c723ad0 --To-tag 848819789
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Stopping retransmission on '66814a9578fab3d5100e465e613d4661@10.10.7.31:5060' of Request 102: Match Found
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Destroying SIP dialog 66814a9578fab3d5100e465e613d4661@10.10.7.31:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c:
---------- SIP HISTORY for '66814a9578fab3d5100e465e613d4661@10.10.7.31:5060'
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: * SIP Call
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: 001. NeedDestroy Setting needdestroy because got OPTIONS response
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c:
---------- END SIP HISTORY for '66814a9578fab3d5100e465e613d4661@10.10.7.31:5060'
[Jun 20 21:15:34] DEBUG[1311] devicestate.c: No provider found, checking channel drivers for SIP - 1*7403
[Jun 20 21:15:34] DEBUG[1311] chan_sip.c: Checking device state for peer 1*7403
[Jun 20 21:15:34] DEBUG[1311] devicestate.c: Changing state for SIP/1*7403 - state 1 (Not in use)
[Jun 20 21:15:34] DEBUG[1294] threadpool.c: Increasing threadpool stasis/pool's size by 1
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] pbx.c: Launching 'Answer'
[Jun 20 21:15:34] VERBOSE[507737][C-0000001e] pbx.c: Executing [7405@default:1] Answer("SIP/1*7403-0000001d", "") in new stack
[Jun 20 21:15:34] DEBUG[1294] threadpool.c: Increasing threadpool stasis/pool's size by 1
[Jun 20 21:15:34] VERBOSE[1436] cdr.c: 0x7f58c8075890 - Set answered time to 1718903734.568450
[Jun 20 21:15:34] DEBUG[1311] devicestate.c: No provider found, checking channel drivers for SIP - 1*7403
[Jun 20 21:15:34] DEBUG[1311] chan_sip.c: Checking device state for peer 1*7403
[Jun 20 21:15:34] DEBUG[1311] devicestate.c: Changing state for SIP/1*7403 - state 1 (Not in use)
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] chan_sip.c: SIP answering channel: SIP/1*7403-0000001d
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: (0x55b4e86a0620) RTP setting the marker bit due to a source update
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: (0x55b4e86a0620) DTLS - ast_rtp_activate rtp=0x55b4e91e9290 - setup and perform DTLS'
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: (0x55b4e91e9290) DTLS perform handshake - ssl = (nil), setup = 0
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: (0x55b4e91e9290) DTLS perform handshake - ssl = (nil), setup = 0
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] chan_sip.c: ** Our capability: (alaw|ulaw) Video flag: True Text flag: True
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] chan_sip.c: ** Our prefcodec: (nothing)
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] chan_sip.c: -- Done with adding codecs to SDP
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] chan_sip.c: Setting framing on incoming call: 0
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] chan_sip.c: Done building SDP. Settling with this capability: (alaw|ulaw)
[Jun 20 21:15:34] DEBUG[507737][C-0000001e] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.10.7.84:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: = Looking for Call ID: 63921f2b54cb7aa33b9a62f968fe2225@10.10.7.31:5060 (Checking To) --From tag as1bd396c3 --To-tag 148576609
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Stopping retransmission on '63921f2b54cb7aa33b9a62f968fe2225@10.10.7.31:5060' of Request 102: Match Found
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Destroying SIP dialog 63921f2b54cb7aa33b9a62f968fe2225@10.10.7.31:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c:
---------- SIP HISTORY for '63921f2b54cb7aa33b9a62f968fe2225@10.10.7.31:5060'
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: * SIP Call
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: 001. NeedDestroy Setting needdestroy because got OPTIONS response
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c:
---------- END SIP HISTORY for '63921f2b54cb7aa33b9a62f968fe2225@10.10.7.31:5060'
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: = Looking for Call ID: 1565120909987-58651465025737@10.10.7.84 (Checking From) --From tag 2528113059 --To-tag as12f77c76
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Jun 20 21:15:34] DEBUG[1462][C-0000001e] chan_sip.c: Stopping retransmission on '1565120909987-58651465025737@10.10.7.84' of Response 2: Match Found
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 10.10.10.191:5060
[Jun 20 21:15:34] DEBUG[1462] chan_sip.c: Trying to put 'OPTIONS sip' onto UDP socket destined for 213.40.29.7:5060
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: ehtestvm-1718903734.31: Seed ts: 920793019 current time: 1718903735.052473
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] pbx.c: Launching 'AGI'
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] pbx.c: Executing [7405@default:2] AGI("SIP/1*7403-0000001d", "agitest.php") in new stack
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/agitest.php
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_request: agitest.php
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_channel: SIP/1*7403-0000001d
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_language: en
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_type: SIP
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_uniqueid: ehtestvm-1718903734.31
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_version: 18.22.0
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_callerid: 7403
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_calleridname: 7403
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_callingpres: 0
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_callingani2: 0
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_callington: 0
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_callingtns: 0
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_dnid: 7405
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_rdnis: unknown
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_context: default
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_extension: 7405
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_priority: 2
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_enhanced: 0.0
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_accountcode: 5577864113
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> agi_threadid: 140018926560960
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >>
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: ehtestvm-1718903734.31: Packet 2 < 15. Ignoring
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Rx <<
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Tx >> 510 Invalid or unknown command
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] res_agi.c: <SIP/1*7403-0000001d>AGI Script agitest.php completed, returning 0
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] pbx.c: Launching 'Hangup'
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] pbx.c: Executing [7405@default:3] Hangup("SIP/1*7403-0000001d", "") in new stack
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] channel.c: Soft-Hanging (0x20) up channel 'SIP/1*7403-0000001d'
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] pbx.c: Spawn extension (default,7405,3) exited non-zero on 'SIP/1*7403-0000001d'
[Jun 20 21:15:35] VERBOSE[507737][C-0000001e] pbx.c: Spawn extension (default, 7405, 3) exited non-zero on 'SIP/1*7403-0000001d'
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] channel.c: Soft-Hanging (0x10) up channel 'SIP/1*7403-0000001d'
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] channel.c: Soft-Hanging (0x80) up channel 'SIP/1*7403-0000001d'
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] chan_sip.c: Updating call counter for incoming call
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: (0x55b4e86a0620) DTLS srtp - stopped timeout timer'
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: (0x55b4e86a0620) DTLS srtp - stopped timeout timer'
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] res_rtp_asterisk.c: (ehtestvm-1718903734.31) RTP Stop
[Jun 20 21:15:35] DEBUG[1311] devicestate.c: No provider found, checking channel drivers for SIP - 1*7403
[Jun 20 21:15:35] DEBUG[1311] chan_sip.c: Checking device state for peer 1*7403
[Jun 20 21:15:35] DEBUG[1311] devicestate.c: Changing state for SIP/1*7403 - state 1 (Not in use)
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] chan_sip.c: Trying to put 'BYE sip:1*7' onto UDP socket destined for 10.10.7.84:5060
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] channel.c: Channel 0x55b4e91f3a60 'SIP/1*7403-0000001d' destroying
[Jun 20 21:15:35] VERBOSE[1436] cdr.c: 0x7f58c8075890 - Beginning finalize/dispatch for SIP/1*7403-0000001d
[Jun 20 21:15:35] VERBOSE[1436] cdr.c: 0x7f58c8075890 - Dispatching CDR for Party A SIP/1*7403-0000001d, Party B <none>
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] stasis.c: Destroying topic. name: channel:ehtestvm-1718903734.31, detail:
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] stasis.c: Topic 'channel:ehtestvm-1718903734.31': 0x55b4e91903e0 destroyed
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] channel_internal_api.c: <initializing>: Formats: (nothing)
[Jun 20 21:15:35] DEBUG[507737][C-0000001e] channel_internal_api.c: Channel is being initialized or destroyed
[Jun 20 21:15:35] DEBUG[1311] devicestate.c: No provider found, checking channel drivers for SIP - 1*7403
[Jun 20 21:15:35] DEBUG[1311] chan_sip.c: Checking device state for peer 1*7403
[Jun 20 21:15:35] DEBUG[1311] devicestate.c: Changing state for SIP/1*7403 - state 1 (Not in use)
[Jun 20 21:15:35] DEBUG[1436] cdr_radius.c: Unable to create RADIUS record. CDR not recorded!
[Jun 20 21:15:35] DEBUG[1436] stasis.c: Creating topic. name: channel:ehtestvm-1718903735.32, detail:
[Jun 20 21:15:35] DEBUG[1436] stasis.c: Topic 'channel:ehtestvm-1718903735.32': 0x7f58c80272d0 created
[Jun 20 21:15:35] DEBUG[1436] stasis.c: Destroying topic. name: channel:ehtestvm-1718903735.32, detail:
[Jun 20 21:15:35] DEBUG[1436] stasis.c: Topic 'channel:ehtestvm-1718903735.32': 0x7f58c80272d0 destroyed
[Jun 20 21:15:35] DEBUG[1462] chan_sip.c: = Looking for Call ID: 1565120909987-58651465025737@10.10.7.84 (Checking To) --From tag as12f77c76 --To-tag 2528113059
[Jun 20 21:15:35] DEBUG[1462][C-0000001e] chan_sip.c: Stopping retransmission on '1565120909987-58651465025737@10.10.7.84' of Request 102: Match Found
[Jun 20 21:15:35] DEBUG[1462] chan_sip.c: Destroying SIP dialog 1565120909987-58651465025737@10.10.7.84
[Jun 20 21:15:35] DEBUG[1462] chan_sip.c:
Not an asterisk expert so any degree of help will be highly appreciated 
Best,
Hisham