vicidial.org

ViciBox v.9.0.3 200630-2117
VERSION: 2.14-842a
BUILD: 220127-2915
© 2022 ViciDial Group
Asterisk: 13.38.2-vici

Brand new installation, at first asterisk wasn't starting up, fixed with:

Code: Select all

zypper in openSUSE-Leap-15.1-ViciDial:libjansson4

However, now when dialling out, the moment the callee picks up and the call is connected, the Agent screen shows Call Hungup, and the Realtime Report shows the agent in a Dead call.
Call proceeds just fine and the Recording occurs as expected, dispositions also fine.

Just this issue with Dead Call, which will obviously impact reporting (as well as creating FUD with Agents).

Any ideas?

Thanks
IanGP

did you update to leap 15.2? if not try that dirst, heres my article on how to do so:

https://dialer.one/how-to-upgrade-opens ... 2-or-15-3/

hope this helps,

Chris aka carpenox

can you post the CLI Log, with and without sip debug.

type asterisk -vvvvvvr for asterisk cli

type sip set debug on - to enable debug
sip set debug off - to disable debug...

striker wrote:can you post the CLI Log, with and without sip debug.

type asterisk -vvvvvvr for asterisk cli

type sip set debug on - to enable debug
sip set debug off - to disable debug...

SIP-NoDebug

Code: Select all

[Feb 10 11:28:02]     -- Called 6666
[Feb 10 11:28:02]     -- SIP/6666-000003cc is ringing
[Feb 10 11:28:02]        > 0x7f915800e150 -- Strict RTP learning after remote address set to: 000.00.136.133:49353
[Feb 10 11:28:02]     -- SIP/6666-000003cc answered
[Feb 10 11:28:02]     -- Executing [8600060@default:1] MeetMe("SIP/6666-000003cc", "8600060,F") in new stack
[Feb 10 11:28:02]     -- Created MeetMe conference 1023 for conference '8600060'
[Feb 10 11:28:02]     -- <SIP/6666-000003cc> Playing 'conf-onlyperson.gsm' (language 'en')
[Feb 10 11:28:02]        > 0x7f915800e150 -- Strict RTP learning after ICE completion
[Feb 10 11:28:02]        > 0x7f915800e150 -- Strict RTP learning after remote address set to: 000.00.136.133:49353
[Feb 10 11:28:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:03]        > 0x7f915800e150 -- Strict RTP switching to RTP target address 000.00.136.133:49353 as source
[Feb 10 11:28:04]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:06]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:07]        > 0x7f915800e150 -- Strict RTP learning complete - Locking on source address 000.00.136.133:49353
[Feb 10 11:28:08]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:11]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:12]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:16]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:16]     -- Called 8600060@default
[Feb 10 11:28:16]     -- Executing [8600060@default:1] MeetMe("Local/8600060@default-000006db;2", "8600060,F") in new stack
[Feb 10 11:28:16]     -- Local/8600060@default-000006db;1 answered
[Feb 10 11:28:16]     -- Executing [50127835560000@default:1] AGI("Local/8600060@default-000006db;1", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 10 11:28:16]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTING))
[Feb 10 11:28:16]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPDTO=60))
[Feb 10 11:28:16]     -- <Local/8600060@default-000006db;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 10 11:28:16]     -- Executing [50127835560000@default:2] Dial("Local/8600060@default-000006db;1", "SIP/27835560000@CM_1,,tTo") in new stack
[Feb 10 11:28:16]   == Using SIP RTP CoS mark 5
[Feb 10 11:28:16]     -- Called SIP/27835560000@CM_1
[Feb 10 11:28:16]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:16]     -- Called 58600060@default
[Feb 10 11:28:16]     -- Executing [58600060@default:1] MeetMe("Local/58600060@default-000006dc;2", "8600060,Fmq") in new stack
[Feb 10 11:28:16]     -- Local/58600060@default-000006dc;1 answered
[Feb 10 11:28:16]     -- Executing [8309@default:1] Answer("Local/58600060@default-000006dc;1", "") in new stack
[Feb 10 11:28:16]     -- Executing [8309@default:2] Monitor("Local/58600060@default-000006dc;1", "wav,20220210-092816_27835560000") in new stack
[Feb 10 11:28:16]     -- Executing [8309@default:3] Wait("Local/58600060@default-000006dc;1", "3600") in new stack
[Feb 10 11:28:17]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:17]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:17]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:17]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:18]        > 0x7f915405a4b0 -- Strict RTP learning after remote address set to: 000.00.55.54:16394
[Feb 10 11:28:18]     -- SIP/CM_1-000003cd is making progress passing it to Local/8600060@default-000006db;1
[Feb 10 11:28:21]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:22]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:23]        > 0x7f915405a4b0 -- Strict RTP switching to RTP target address 000.00.55.54:16394 as source
[Feb 10 11:28:23]        > 0x7f915405a4b0 -- Strict RTP learning complete - Locking on source address 000.00.55.54:16394
[Feb 10 11:28:28]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:28]     -- SIP/CM_1-000003cd answered Local/8600060@default-000006db;1
[Feb 10 11:28:28]     -- Channel SIP/CM_1-000003cd joined 'simple_bridge' basic-bridge <61a7cb45-ad0e-469d-85f3-6e428c501a01>
[Feb 10 11:28:28]     -- Channel Local/8600060@default-000006db;1 joined 'simple_bridge' basic-bridge <61a7cb45-ad0e-469d-85f3-6e428c501a01>
[Feb 10 11:28:29]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:32]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:33]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:36]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:36]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600060@default-000006db;1
[Feb 10 11:28:36]     -- Channel Local/8600060@default-000006db;1 left 'simple_bridge' basic-bridge <61a7cb45-ad0e-469d-85f3-6e428c501a01>
[Feb 10 11:28:36]     -- Channel SIP/CM_1-000003cd left 'simple_bridge' basic-bridge <61a7cb45-ad0e-469d-85f3-6e428c501a01>
[Feb 10 11:28:36]   == Spawn extension (default, 50127835560000, 2) exited non-zero on 'Local/8600060@default-000006db;1'
[Feb 10 11:28:36]     -- Executing [h@default:1] AGI("Local/8600060@default-000006db;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----20-----SIP 200 OK)") in new stack
[Feb 10 11:28:36]     -- <Local/8600060@default-000006db;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----20-----SIP 200 OK) completed, returning 0
[Feb 10 11:28:36]   == Spawn extension (default, 8600060, 1) exited non-zero on 'Local/8600060@default-000006db;2'
[Feb 10 11:28:36] WARNING[6475][C-00000d78]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Feb 10 11:28:36]     -- Executing [h@default:1] AGI("Local/8600060@default-000006db;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Feb 10 11:28:36]     -- <Local/8600060@default-000006db;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Feb 10 11:28:36]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:36] NOTICE[6513]: manager.c:4471 action_hangup: Request to hangup non-existent channel: Local/8600060@default-000006db;2
[Feb 10 11:28:36]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:36]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/58600060@default-000006dc;2
[Feb 10 11:28:36]   == Spawn extension (default, 58600060, 1) exited non-zero on 'Local/58600060@default-000006dc;2'
[Feb 10 11:28:36] WARNING[6479][C-00000d7a]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Feb 10 11:28:36]     -- Executing [h@default:1] AGI("Local/58600060@default-000006dc;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[Feb 10 11:28:36]     -- <Local/58600060@default-000006dc;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
[Feb 10 11:28:36]   == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600060@default-000006dc;1'
[Feb 10 11:28:36] WARNING[6478][C-00000d7b]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Feb 10 11:28:36]     -- Executing [h@default:1] AGI("Local/58600060@default-000006dc;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[Feb 10 11:28:36]     -- <Local/58600060@default-000006dc;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
[Feb 10 11:28:37]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:37]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:37]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:38]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:39]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:43]   == SRTCP unprotect failed because of authentication failure
[Feb 10 11:28:43]     -- Hungup 'DAHDI/pseudo-477707325'
[Feb 10 11:28:43]   == Spawn extension (default, 8600060, 1) exited non-zero on 'SIP/6666-000003cc'
[Feb 10 11:28:43]     -- Executing [h@default:1] AGI("SIP/6666-000003cc", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
[Feb 10 11:28:43]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:43]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/6666-000003cc
[Feb 10 11:28:43]     -- <SIP/6666-000003cc>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed, returning 0
[Feb 10 11:28:43]   == Manager 'sendcron' logged on from 127.0.0.1
[Feb 10 11:28:43]     -- Called 55558600060@default
[Feb 10 11:28:43]     -- Executing [55558600060@default:1] MeetMeAdmin("Local/55558600060@default-000006dd;2", "8600060,K") in new stack
[Feb 10 11:28:43] WARNING[6529][C-00000d7c]: app_meetme.c:5261 admin_exec: Conference number '8600060' not found!
[Feb 10 11:28:43]     -- Executing [55558600060@default:2] Hangup("Local/55558600060@default-000006dd;2", "") in new stack
[Feb 10 11:28:43]   == Spawn extension (default, 55558600060, 2) exited non-zero on 'Local/55558600060@default-000006dd;2'
[Feb 10 11:28:43] WARNING[6529][C-00000d7c]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Feb 10 11:28:43]     -- Executing [h@default:1] AGI("Local/55558600060@default-000006dd;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Feb 10 11:28:43]     -- <Local/55558600060@default-000006dd;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Feb 10 11:28:44]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:44]   == Manager 'sendcron' logged off from 127.0.0.1
[Feb 10 11:28:46] ERROR[2349]: chan_sip.c:4309 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
innotrend*CLI> exit
[Feb 10 11:28:49] Asterisk cleanly ending (0).
[Feb 10 11:28:49] Executing last minute cleanups


carpenox wrote:did you update to leap 15.2? if not try that dirst, heres my article on how to do so:

https://cyburdial.net/how-to-upgrade-op ... 2-or-15-3/

hope this helps,

Chris aka carpenox

Thanks, I'll give this a try tonight.

Upgraded from 15.1 to 15.2, calls still hangup on callee answer.

SIP-Debug

Code: Select all

SIP Debugging enabled
[11:29:51] Reliably Transmitting (NAT) to 000.00.58.11:5060:
[11:29:51] OPTIONS sip:000.00.58.11 SIP/2.0
[11:29:51] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40292888;rport
[11:29:51] Max-Forwards: 70
[11:29:51] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as53c81925
[11:29:51] To: <sip:000.00.58.11>
[11:29:51] Contact: <sip:asterisk@000.00.236.130:5060>
[11:29:51] Call-ID: 3f921755711d9ad238d524a91bd3e4b6@000.00.236.130:5060
[11:29:51] CSeq: 102 OPTIONS
[11:29:51] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:51] Date: Thu, 10 Feb 2022 09:29:51 GMT
[11:29:51] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:51] Supported: replaces, timer
[11:29:51] Content-Length: 0

[11:29:51] ---
[11:29:51] Reliably Transmitting (NAT) to 000.00.184.100:5060:
[11:29:51] OPTIONS sip:000.00.184.100 SIP/2.0
[11:29:51] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK5fb7092a;rport
[11:29:51] Max-Forwards: 70
[11:29:51] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as59ded067
[11:29:51] To: <sip:000.00.184.100>
[11:29:51] Contact: <sip:asterisk@000.00.236.130:5060>
[11:29:51] Call-ID: 4f99ab86559ffb6c2b7d81383803a35c@000.00.236.130:5060
[11:29:51] CSeq: 102 OPTIONS
[11:29:51] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:51] Date: Thu, 10 Feb 2022 09:29:51 GMT
[11:29:51] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:51] Supported: replaces, timer
[11:29:51] Content-Length: 0

[11:29:51] <--- SIP read from UDP:000.00.184.100:5060 --->
[11:29:51] SIP/2.0 200 OK
[11:29:51] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK5fb7092a;rport=5060;received=000.00.236.130
[11:29:51] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as59ded067
[11:29:51] To: <sip:000.00.184.100>;tag=fbd3937dab4e3807eec3291d529da3a6.8fd0
[11:29:51] Call-ID: 4f99ab86559ffb6c2b7d81383803a35c@000.00.236.130:5060
[11:29:51] CSeq: 102 OPTIONS
[11:29:51] Content-Length: 0
[11:29:51]
[11:29:51] <------------->
[11:29:51] --- (7 headers 0 lines) ---
[11:29:51] Really destroying SIP dialog '4f99ab86559ffb6c2b7d81383803a35c@000.00.236.130:5060' Method: OPTIONS
[11:29:52] Retransmitting #1 (NAT) to 000.00.58.11:5060:
[11:29:52] OPTIONS sip:000.00.58.11 SIP/2.0
[11:29:52] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40292888;rport
[11:29:52] Max-Forwards: 70
[11:29:52] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as53c81925
[11:29:52] To: <sip:000.00.58.11>
[11:29:52] Contact: <sip:asterisk@000.00.236.130:5060>
[11:29:52] Call-ID: 3f921755711d9ad238d524a91bd3e4b6@000.00.236.130:5060
[11:29:52] CSeq: 102 OPTIONS
[11:29:52] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:52] Date: Thu, 10 Feb 2022 09:29:51 GMT
[11:29:52] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:52] Supported: replaces, timer
[11:29:52] Content-Length: 0

[11:29:52] ---
[11:29:53] Retransmitting #2 (NAT) to 000.00.58.11:5060:
[11:29:53] OPTIONS sip:000.00.58.11 SIP/2.0
[11:29:53] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40292888;rport
[11:29:53] Max-Forwards: 70
[11:29:53] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as53c81925
[11:29:53] To: <sip:000.00.58.11>
[11:29:53] Contact: <sip:asterisk@000.00.236.130:5060>
[11:29:53] Call-ID: 3f921755711d9ad238d524a91bd3e4b6@000.00.236.130:5060
[11:29:53] CSeq: 102 OPTIONS
[11:29:53] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:53] Date: Thu, 10 Feb 2022 09:29:51 GMT
[11:29:53] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:53] Supported: replaces, timer
[11:29:53] Content-Length: 0

[11:29:53] ---
[11:29:54] Retransmitting #3 (NAT) to 000.00.58.11:5060:
[11:29:54] OPTIONS sip:000.00.58.11 SIP/2.0
[11:29:54] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40292888;rport
[11:29:54] Max-Forwards: 70
[11:29:54] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as53c81925
[11:29:54] To: <sip:000.00.58.11>
[11:29:54] Contact: <sip:asterisk@000.00.236.130:5060>
[11:29:54] Call-ID: 3f921755711d9ad238d524a91bd3e4b6@000.00.236.130:5060
[11:29:54] CSeq: 102 OPTIONS
[11:29:54] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:54] Date: Thu, 10 Feb 2022 09:29:51 GMT
[11:29:54] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:54] Supported: replaces, timer
[11:29:54] Content-Length: 0

[11:29:54] ---
[11:29:54]   == WebSocket connection from '000.00.136.133:56170' for protocol 'sip' accepted using version '13'
[11:29:54]
[11:29:54] <--- SIP read from WS:000.00.136.133:56170 --->
[11:29:54] REGISTER sip:000.00.236.130 SIP/2.0
[11:29:54] Via: SIP/2.0/WSS 192.0.2.208;branch=z9hG4bK5732767
[11:29:54] Max-Forwards: 70
[11:29:54] To: "6666" <sip:6666@000.00.236.130>
[11:29:54] From: "6666" <sip:6666@000.00.236.130>;tag=btqcf8vvp2
[11:29:54] Call-ID: rbldpetpopihlf3heeujqa
[11:29:54] CSeq: 81 REGISTER
[11:29:54] Contact: <sip:1ujsbm0a@192.0.2.208;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:63c68d06-c12e-4871-9e8f-09f13ee2b357>";expires=600
[11:29:54] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
[11:29:54] Supported: path, gruu, outbound
[11:29:54] User-Agent: VICIphone 1.0-rc1
[11:29:54] Content-Length: 0
[11:29:54]
[11:29:54] <------------->
[11:29:54] --- (12 headers 0 lines) ---
[11:29:54]
[11:29:54] <--- Transmitting (NAT) to 000.00.136.133:56170 --->
[11:29:54] SIP/2.0 401 Unauthorized
[11:29:54] Via: SIP/2.0/WSS 192.0.2.208;branch=z9hG4bK5732767;received=000.00.136.133;rport=56170
[11:29:54] From: "6666" <sip:6666@000.00.236.130>;tag=btqcf8vvp2
[11:29:54] To: "6666" <sip:6666@000.00.236.130>;tag=as294de3fa
[11:29:54] Call-ID: rbldpetpopihlf3heeujqa
[11:29:54] CSeq: 81 REGISTER
[11:29:54] Server: Asterisk PBX 13.38.2-vici
[11:29:54] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:54] Supported: replaces, timer
[11:29:54] WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61221018"
[11:29:54] Content-Length: 0
[11:29:54]
[11:29:54]
[11:29:54] <------------>
[11:29:54] Scheduling destruction of SIP dialog 'rbldpetpopihlf3heeujqa' in 32000 ms (Method: REGISTER)
[11:29:54]
[11:29:54] <--- SIP read from WS:000.00.136.133:56170 --->
[11:29:54] REGISTER sip:000.00.236.130 SIP/2.0
[11:29:54] Via: SIP/2.0/WSS 192.0.2.208;branch=z9hG4bK3302488
[11:29:54] Max-Forwards: 70
[11:29:54] To: "6666" <sip:6666@000.00.236.130>
[11:29:54] From: "6666" <sip:6666@000.00.236.130>;tag=btqcf8vvp2
[11:29:54] Call-ID: rbldpetpopihlf3heeujqa
[11:29:54] CSeq: 82 REGISTER
[11:29:54] Authorization: Digest algorithm=MD5, username="6666", realm="asterisk", nonce="61221018", uri="sip:000.00.236.130", response="30b2b29c13de7deb02a1ac17ba2fb128"
[11:29:54] Contact: <sip:1ujsbm0a@192.0.2.208;transport=wss>;reg-id=1;+sip.instance="<urn:uuid:63c68d06-c12e-4871-9e8f-09f13ee2b357>";expires=600
[11:29:54] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
[11:29:54] Supported: path, gruu, outbound
[11:29:54] User-Agent: VICIphone 1.0-rc1
[11:29:54] Content-Length: 0
[11:29:54]
[11:29:54] <------------->
[11:29:54] --- (13 headers 0 lines) ---
[11:29:54] ERROR[6643]: utils.c:1499 ast_careful_fwrite: fflush() returned error: Bad file descriptor
[11:29:54] ERROR[6643]: tcptls.c:488 tcptls_stream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
[11:29:54]   == WebSocket connection from '000.00.136.133:56168' forcefully closed due to fatal write error
[11:29:54]     -- Registered SIP '6666' at 000.00.136.133:56170
[11:29:54] Reliably Transmitting (NAT) to 000.00.136.133:56170:
[11:29:54] OPTIONS sip:1ujsbm0a@192.0.2.208;transport=wss SIP/2.0
[11:29:54] Via: SIP/2.0/WS 000.00.236.130:0;branch=z9hG4bK49fa3070;rport
[11:29:54] Max-Forwards: 70
[11:29:54] From: "asterisk" <sip:asterisk@000.00.236.130:0>;tag=as436413ea
[11:29:54] To: <sip:1ujsbm0a@192.0.2.208;transport=wss>
[11:29:54] Contact: <sip:asterisk@000.00.236.130:0;transport=ws>
[11:29:54] Call-ID: 3a6dc246566dff6b2c12972e24593751@000.00.236.130:0
[11:29:54] CSeq: 102 OPTIONS
[11:29:54] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:54] Date: Thu, 10 Feb 2022 09:29:54 GMT
[11:29:54] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:54] Supported: replaces, timer
[11:29:54] Content-Length: 0
[11:29:54]

[11:29:54] <--- Transmitting (NAT) to 000.00.136.133:56170 --->
[11:29:54] SIP/2.0 200 OK
[11:29:54] Via: SIP/2.0/WSS 192.0.2.208;branch=z9hG4bK3302488;received=000.00.136.133;rport=56170
[11:29:54] From: "6666" <sip:6666@000.00.236.130>;tag=btqcf8vvp2
[11:29:54] To: "6666" <sip:6666@000.00.236.130>;tag=as294de3fa
[11:29:54] Call-ID: rbldpetpopihlf3heeujqa
[11:29:54] CSeq: 82 REGISTER
[11:29:54] Server: Asterisk PBX 13.38.2-vici
[11:29:54] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:54] Supported: replaces, timer
[11:29:54] Expires: 600
[11:29:54] Contact: <sip:1ujsbm0a@192.0.2.208;transport=wss>;expires=600
[11:29:54] Date: Thu, 10 Feb 2022 09:29:54 GMT
[11:29:54] Content-Length: 0

[11:29:54] <------------>
[11:29:54] Scheduling destruction of SIP dialog 'rbldpetpopihlf3heeujqa' in 32000 ms (Method: REGISTER)
[11:29:54]
[11:29:54] <--- SIP read from WS:000.00.136.133:56170 --->
[11:29:54] SIP/2.0 200 OK
[11:29:54] Via: SIP/2.0/WS 000.00.236.130:0;branch=z9hG4bK49fa3070;rport
[11:29:54] To: <sip:1ujsbm0a@192.0.2.208;transport=wss>;tag=d93fdh3ds9
[11:29:54] From: "asterisk" <sip:asterisk@000.00.236.130:0>;tag=as436413ea
[11:29:54] Call-ID: 3a6dc246566dff6b2c12972e24593751@000.00.236.130:0
[11:29:54] CSeq: 102 OPTIONS
[11:29:54] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
[11:29:54] Accept: application/sdp,application/dtmf-relay
[11:29:54] Supported: outbound
[11:29:54] User-Agent: VICIphone 1.0-rc1
[11:29:54] Content-Length: 0
[11:29:54]
[11:29:54] <------------->
[11:29:54] --- (11 headers 0 lines) ---
[11:29:54] NOTICE[6643]: chan_sip.c:24817 handle_response_peerpoke: Peer '6666' is now Reachable. (58ms / 2000ms)
[11:29:55] Really destroying SIP dialog '417739f9097fc88a35c1049d329119a8@000.00.236.130:5060' Method: NOTIFY
[11:29:55] Retransmitting #4 (NAT) to 000.00.58.11:5060:
[11:29:55] OPTIONS sip:000.00.58.11 SIP/2.0
[11:29:55] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40292888;rport
[11:29:55] Max-Forwards: 70
[11:29:55] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as53c81925
[11:29:55] To: <sip:000.00.58.11>
[11:29:55] Contact: <sip:asterisk@000.00.236.130:5060>
[11:29:55] Call-ID: 3f921755711d9ad238d524a91bd3e4b6@000.00.236.130:5060
[11:29:55] CSeq: 102 OPTIONS
[11:29:55] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:55] Date: Thu, 10 Feb 2022 09:29:51 GMT
[11:29:55] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:55] Supported: replaces, timer
[11:29:55] Content-Length: 0

[11:29:55] ---
[11:29:55] Really destroying SIP dialog '3f921755711d9ad238d524a91bd3e4b6@000.00.236.130:5060' Method: OPTIONS
[11:29:55] Really destroying SIP dialog '3a6dc246566dff6b2c12972e24593751@000.00.236.130:0' Method: OPTIONS
[11:29:56]   == Manager 'sendcron' logged on from 127.0.0.1
[11:29:56]   == Using SIP RTP CoS mark 5
[11:29:56] Audio is at 17948
[11:29:56] Adding codec ulaw to SDP
[11:29:56] Adding codec gsm to SDP
[11:29:56] Adding non-codec 0x1 (telephone-event) to SDP
[11:29:56] Reliably Transmitting (NAT) to 000.00.136.133:56170:
[11:29:56] INVITE sip:1ujsbm0a@192.0.2.208;transport=wss SIP/2.0
[11:29:56] Via: SIP/2.0/WS 000.00.236.130:0;branch=z9hG4bK4da75b61;rport
[11:29:56] Max-Forwards: 70
[11:29:56] From: "ACagcW16444853946666666666666666" <sip:27108240682@000.00.236.130:0>;tag=as7a133fdb
[11:29:56] To: <sip:1ujsbm0a@192.0.2.208;transport=wss>
[11:29:56] Contact: <sip:27108240682@000.00.236.130:0;transport=ws>
[11:29:56] Call-ID: 5f678ec451204cd4462efc5f1db9c452@000.00.236.130:0
[11:29:56] CSeq: 102 INVITE
[11:29:56] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:56] Date: Thu, 10 Feb 2022 09:29:56 GMT
[11:29:56] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:29:56] Supported: replaces, timer
[11:29:56] Remote-Party-ID: "ACagcW16444853946666666666666666" <sip:27108240682@000.00.236.130>;party=calling;privacy=off;screen=no
[11:29:56] Content-Type: application/sdp
[11:29:56] Content-Length: 670
[11:29:56]
[11:29:56] v=0
[11:29:56] o=root 1736140266 1736140266 IN IP4 000.00.236.130
[11:29:56] s=Asterisk PBX 13.38.2-vici
[11:29:56] c=IN IP4 000.00.236.130
[11:29:56] t=0 0
[11:29:56] m=audio 17948 RTP/SAVPF 0 3 101
[11:29:56] a=rtpmap:0 PCMU/8000
[11:29:56] a=rtpmap:3 GSM/8000
[11:29:56] a=rtpmap:101 telephone-event/8000
[11:29:56] a=fmtp:101 0-16
[11:29:56] a=ptime:20
[11:29:56] a=maxptime:150
[11:29:56] a=ice-ufrag:0b85003020a787951102123f3a000971
[11:29:56] a=ice-pwd:6ee5a6df6713f56173e213fa7e8de840
[11:29:56] a=candidate:H9c26ec82 1 UDP 2130706431 000.00.236.130 17948 typ host
[11:29:56] a=candidate:H9c26ec82 2 UDP 2130706430 000.00.236.130 17949 typ host
[11:29:56] a=connection:new
[11:29:56] a=setup:actpass
[11:29:56] a=fingerprint:SHA-256 D5:F9:FE:C5:96:CB:05:33:EB:1E:02:4F:17:C5:44:97:A8:A7:90:33:77:1E:37:21:EA:A6:3F:CB:B0:B4:FF:6A
[11:29:56] a=sendrecv

[11:29:56]     -- Called 6666
[11:29:56]
[11:29:56] <--- SIP read from WS:000.00.136.133:56170 --->
[11:29:56] SIP/2.0 100 Trying
[11:29:56] Via: SIP/2.0/WS 000.00.236.130:0;branch=z9hG4bK4da75b61;rport
[11:29:56] To: <sip:1ujsbm0a@192.0.2.208;transport=wss>
[11:29:56] From: "ACagcW16444853946666666666666666" <sip:27108240682@000.00.236.130:0>;tag=as7a133fdb
[11:29:56] Call-ID: 5f678ec451204cd4462efc5f1db9c452@000.00.236.130:0
[11:29:56] CSeq: 102 INVITE
[11:29:56] Supported: outbound
[11:29:56] User-Agent: VICIphone 1.0-rc1
[11:29:56] Content-Length: 0
[11:29:56]
[11:29:56] <------------->
[11:29:56] --- (9 headers 0 lines) ---
[11:29:56]
[11:29:56] <--- SIP read from WS:000.00.136.133:56170 --->
[11:29:56] SIP/2.0 180 Ringing
[11:29:56] Via: SIP/2.0/WS 000.00.236.130:0;branch=z9hG4bK4da75b61;rport
[11:29:56] To: <sip:1ujsbm0a@192.0.2.208;transport=wss>;tag=plgtohq9to
[11:29:56] From: "ACagcW16444853946666666666666666" <sip:27108240682@000.00.236.130:0>;tag=as7a133fdb
[11:29:56] Call-ID: 5f678ec451204cd4462efc5f1db9c452@000.00.236.130:0
[11:29:56] CSeq: 102 INVITE
[11:29:56] Contact: <sip:1ujsbm0a@192.0.2.208;transport=wss>
[11:29:56] Supported: outbound
[11:29:56] User-Agent: VICIphone 1.0-rc1
[11:29:56] Content-Length: 0
[11:29:56]
[11:29:56] <------------->
[11:29:56] --- (10 headers 0 lines) ---
[11:29:56] sip_route_dump: route/path hop: <sip:1ujsbm0a@192.0.2.208;transport=wss>
[11:29:56]     -- SIP/6666-000003ce is ringing
[11:29:56]        > 0x7f91c8032e50 -- Strict RTP learning after remote address set to: 000.00.136.133:37113
[11:29:56]
[11:29:56] <--- SIP read from WS:000.00.136.133:56170 --->
[11:29:56] SIP/2.0 200 OK
[11:29:56] Via: SIP/2.0/WS 000.00.236.130:0;branch=z9hG4bK4da75b61;rport
[11:29:56] To: <sip:1ujsbm0a@192.0.2.208;transport=wss>;tag=plgtohq9to
[11:29:56] From: "ACagcW16444853946666666666666666" <sip:27108240682@000.00.236.130:0>;tag=as7a133fdb
[11:29:56] Call-ID: 5f678ec451204cd4462efc5f1db9c452@000.00.236.130:0
[11:29:56] CSeq: 102 INVITE
[11:29:56] Contact: <sip:1ujsbm0a@192.0.2.208;transport=wss>
[11:29:56] Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
[11:29:56] Supported: outbound
[11:29:56] User-Agent: VICIphone 1.0-rc1
[11:29:56] Content-Type: application/sdp
[11:29:56] Content-Length: 1252
[11:29:56]
[11:29:56] v=0
[11:29:56] o=- 7434380689229622682 2 IN IP4 127.0.0.1
[11:29:56] s=-
[11:29:56] t=0 0
[11:29:56] a=msid-semantic: WMS 8f0BYp75AC7ssMNw4CR5gtxPmd8uQpG6cKDH
[11:29:56] m=audio 37113 UDP/TLS/RTP/SAVPF 0 101
[11:29:56] c=IN IP4 10.8.0.79
[11:29:56] a=rtcp:35824 IN IP4 10.8.0.79
[11:29:56] a=candidate:3377426696 1 udp 2122260223 10.8.0.79 37113 typ host generation 0 network-id 1
[11:29:56] a=candidate:3790925857 1 udp 2122194687 192.168.8.125 60523 typ host generation 0 network-id 2 network-cost 10
[11:29:56] a=candidate:3377426696 2 udp 2122260222 10.8.0.79 35824 typ host generation 0 network-id 1
[11:29:56] a=candidate:3790925857 2 udp 2122194686 192.168.8.125 45643 typ host generation 0 network-id 2 network-cost 10
[11:29:56] a=ice-ufrag:C8CH
[11:29:56] a=ice-pwd:PDSt8Ua+jRx8qs81FE1dFnMg
[11:29:56] a=ice-options:trickle
[11:29:56] a=fingerprint:sha-256 A9:04:33:92:B7:A9:C0:89:18:80:3A:24:93:F1:39:67:6E:91:35:BA:6F:B2:1D:F5:13:10:EF:27:BF:7E:E4:15
[11:29:56] a=setup:active
[11:29:56] a=mid:0
[11:29:56] a=sendrecv
[11:29:56] a=msid:8f0BYp75AC7ssMNw4CR5gtxPmd8uQpG6cKDH 08abde0b-f29c-423d-97ca-d59c4f1bcde3
[11:29:56] a=rtpmap:0 PCMU/8000
[11:29:56] a=rtpmap:101 telephone-event/8000
[11:29:56] a=ssrc:415776355 cname:7QU7bTYYt2G/vMK1
[11:29:56] a=ssrc:415776355 msid:8f0BYp75AC7ssMNw4CR5gtxPmd8uQpG6cKDH 08abde0b-f29c-423d-97ca-d59c4f1bcde3
[11:29:56] a=ssrc:415776355 mslabel:8f0BYp75AC7ssMNw4CR5gtxPmd8uQpG6cKDH
[11:29:56] a=ssrc:415776355 label:08abde0b-f29c-423d-97ca-d59c4f1bcde3
[11:29:56] <------------->
[11:29:56] --- (12 headers 26 lines) ---
[11:29:56] Got SDP version 2 and unique parts [- 7434380689229622682 IN IP4 127.0.0.1]
[11:29:56] Found RTP audio format 0
[11:29:56] Found RTP audio format 101
[11:29:56] Found audio description format PCMU for ID 0
[11:29:56] Found audio description format telephone-event for ID 101
[11:29:56] Capabilities: us - (ulaw|gsm), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[11:29:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[11:29:56] Peer audio RTP is at port 10.8.0.79:37113
[11:29:56] sip_route_dump: route/path hop: <sip:1ujsbm0a@192.0.2.208;transport=wss>
[11:29:56] Transmitting (NAT) to 000.00.136.133:56170:
[11:29:56] ACK sip:1ujsbm0a@192.0.2.208;transport=wss SIP/2.0
[11:29:56] Via: SIP/2.0/WS 000.00.236.130:0;branch=z9hG4bK2fdcc0b5;rport
[11:29:56] Max-Forwards: 70
[11:29:56] From: "ACagcW16444853946666666666666666" <sip:27108240682@000.00.236.130:0>;tag=as7a133fdb
[11:29:56] To: <sip:1ujsbm0a@192.0.2.208;transport=wss>;tag=plgtohq9to
[11:29:56] Contact: <sip:27108240682@000.00.236.130:0;transport=ws>
[11:29:56] Call-ID: 5f678ec451204cd4462efc5f1db9c452@000.00.236.130:0
[11:29:56] CSeq: 102 ACK
[11:29:56] User-Agent: Asterisk PBX 13.38.2-vici
[11:29:56] Content-Length: 0

[11:29:56] ---
[11:29:56]     -- SIP/6666-000003ce answered
[11:29:56]     -- Executing [8600060@default:1] MeetMe("SIP/6666-000003ce", "8600060,F") in new stack
[11:29:56]     -- Created MeetMe conference 1023 for conference '8600060'
[11:29:56]     -- <SIP/6666-000003ce> Playing 'conf-onlyperson.gsm' (language 'en')
[11:29:56]        > 0x7f91c8032e50 -- Strict RTP learning after ICE completion
[11:29:56]        > 0x7f91c8032e50 -- Strict RTP learning after remote address set to: 000.00.136.133:37113
[11:29:57]   == Manager 'sendcron' logged off from 127.0.0.1
[11:29:57]        > 0x7f91c8032e50 -- Strict RTP switching to RTP target address 000.00.136.133:37113 as source
[11:29:58]   == SRTCP unprotect failed because of authentication failure
[11:30:00]   == SRTCP unprotect failed because of authentication failure
[11:30:00]
[11:30:00] <--- SIP read from UDP:000.00.185.110:5060 --->
[11:30:00] OPTIONS sip:000.00.236.130:5060 SIP/2.0
[11:30:00] Via: SIP/2.0/UDP 000.00.185.110;branch=z9hG4bK80c6.6a52ca87000000000000000000000000.0
[11:30:00] To: <sip:000.00.236.130:5060>
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=dacba587fa51057588cbc4c43ac2ecaf-a391
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Call-ID: 2bf2a3eb4eea42cd-28800@10.35.144.15
[11:30:00] Max-Forwards: 70
[11:30:00] Content-Length: 0
[11:30:00]
[11:30:00] <------------->
[11:30:00] --- (8 headers 0 lines) ---
[11:30:00] Sending to 000.00.185.110:5060 (NAT)
[11:30:00] Looking for s in trunkinbound (domain 000.00.236.130)
[11:30:00]
[11:30:00] <--- Transmitting (NAT) to 000.00.185.110:5060 --->
[11:30:00] SIP/2.0 200 OK
[11:30:00] Via: SIP/2.0/UDP 000.00.185.110;branch=z9hG4bK80c6.6a52ca87000000000000000000000000.0;received=000.00.185.110;rport=5060
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=dacba587fa51057588cbc4c43ac2ecaf-a391
[11:30:00] To: <sip:000.00.236.130:5060>;tag=as5a989e5d
[11:30:00] Call-ID: 2bf2a3eb4eea42cd-28800@10.35.144.15
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Server: Asterisk PBX 13.38.2-vici
[11:30:00] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:00] Supported: replaces, timer
[11:30:00] Contact: <sip:000.00.236.130:5060>
[11:30:00] Accept: application/sdp
[11:30:00] Content-Length: 0
[11:30:00]

[11:30:00] <------------>
[11:30:00] Scheduling destruction of SIP dialog '2bf2a3eb4eea42cd-28800@10.35.144.15' in 32000 ms (Method: OPTIONS)
[11:30:00]
[11:30:00] <--- SIP read from UDP:000.00.184.98:5060 --->
[11:30:00] OPTIONS sip:000.00.236.130:5060 SIP/2.0
[11:30:00] Via: SIP/2.0/UDP 000.00.184.98;branch=z9hG4bK3dd2.17f112b5000000000000000000000000.0
[11:30:00] To: <sip:000.00.236.130:5060>
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=2604b1054517227070401e3a989680d0-8b69
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Call-ID: 2af58330579b1417-22882@10.25.144.11
[11:30:00] Max-Forwards: 70
[11:30:00] Content-Length: 0
[11:30:00]
[11:30:00] <------------->
[11:30:00] --- (8 headers 0 lines) ---
[11:30:00] Sending to 000.00.184.98:5060 (NAT)
[11:30:00] Looking for s in trunkinbound (domain 000.00.236.130)
[11:30:00]
[11:30:00] <--- Transmitting (NAT) to 000.00.184.98:5060 --->
[11:30:00] SIP/2.0 200 OK
[11:30:00] Via: SIP/2.0/UDP 000.00.184.98;branch=z9hG4bK3dd2.17f112b5000000000000000000000000.0;received=000.00.184.98;rport=5060
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=2604b1054517227070401e3a989680d0-8b69
[11:30:00] To: <sip:000.00.236.130:5060>;tag=as617c10d5
[11:30:00] Call-ID: 2af58330579b1417-22882@10.25.144.11
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Server: Asterisk PBX 13.38.2-vici
[11:30:00] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:00] Supported: replaces, timer
[11:30:00] Contact: <sip:000.00.236.130:5060>
[11:30:00] Accept: application/sdp
[11:30:00] Content-Length: 0
[11:30:00]
[11:30:00] <------------>
[11:30:00] Scheduling destruction of SIP dialog '2af58330579b1417-22882@10.25.144.11' in 32000 ms (Method: OPTIONS)
[11:30:00]
[11:30:00] <--- SIP read from UDP:000.00.185.99:5060 --->
[11:30:00] OPTIONS sip:000.00.236.130:5060 SIP/2.0
[11:30:00] Via: SIP/2.0/UDP 000.00.185.99;branch=z9hG4bKb448.fdacce95000000000000000000000000.0
[11:30:00] To: <sip:000.00.236.130:5060>
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=59df438a99a213dfc6bd8f28502f7283-f826
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Call-ID: 5a225bff80abe5ab-11451@10.35.144.14
[11:30:00] Max-Forwards: 70
[11:30:00] Content-Length: 0
[11:30:00]
[11:30:00] <------------->
[11:30:00] --- (8 headers 0 lines) ---
[11:30:00] Sending to 000.00.185.99:5060 (NAT)
[11:30:00] Looking for s in trunkinbound (domain 000.00.236.130)
[11:30:00]
[11:30:00] <--- Transmitting (NAT) to 000.00.185.99:5060 --->
[11:30:00] SIP/2.0 200 OK
[11:30:00] Via: SIP/2.0/UDP 000.00.185.99;branch=z9hG4bKb448.fdacce95000000000000000000000000.0;received=000.00.185.99;rport=5060
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=59df438a99a213dfc6bd8f28502f7283-f826
[11:30:00] To: <sip:000.00.236.130:5060>;tag=as7e0340e4
[11:30:00] Call-ID: 5a225bff80abe5ab-11451@10.35.144.14
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Server: Asterisk PBX 13.38.2-vici
[11:30:00] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:00] Supported: replaces, timer
[11:30:00] Contact: <sip:000.00.236.130:5060>
[11:30:00] Accept: application/sdp
[11:30:00] Content-Length: 0
[11:30:00]
[11:30:00] <------------>
[11:30:00] Scheduling destruction of SIP dialog '5a225bff80abe5ab-11451@10.35.144.14' in 32000 ms (Method: OPTIONS)
[11:30:00]
[11:30:00] <--- SIP read from UDP:000.00.185.98:5060 --->
[11:30:00] OPTIONS sip:000.00.236.130:5060 SIP/2.0
[11:30:00] Via: SIP/2.0/UDP 000.00.185.98;branch=z9hG4bK421.04ee75c3000000000000000000000000.0
[11:30:00] To: <sip:000.00.236.130:5060>
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=ff48732cce851cc79e5c1e7ee06651cc-e92c2433
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Call-ID: 20c45c634a987f2b-14679@10.35.144.11
[11:30:00] Max-Forwards: 70
[11:30:00] Content-Length: 0
[11:30:00]
[11:30:00] <------------->
[11:30:00] --- (8 headers 0 lines) ---
[11:30:00] Sending to 000.00.185.98:5060 (NAT)
[11:30:00] Looking for s in trunkinbound (domain 000.00.236.130)
[11:30:00]
[11:30:00] <--- Transmitting (NAT) to 000.00.185.98:5060 --->
[11:30:00] SIP/2.0 200 OK
[11:30:00] Via: SIP/2.0/UDP 000.00.185.98;branch=z9hG4bK421.04ee75c3000000000000000000000000.0;received=000.00.185.98;rport=5060
[11:30:00] From: <sip:keepalive@sip.cm.nl>;tag=ff48732cce851cc79e5c1e7ee06651cc-e92c2433
[11:30:00] To: <sip:000.00.236.130:5060>;tag=as2c2b2a65
[11:30:00] Call-ID: 20c45c634a987f2b-14679@10.35.144.11
[11:30:00] CSeq: 10 OPTIONS
[11:30:00] Server: Asterisk PBX 13.38.2-vici
[11:30:00] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:00] Supported: replaces, timer
[11:30:00] Contact: <sip:000.00.236.130:5060>
[11:30:00] Accept: application/sdp
[11:30:00] Content-Length: 0
[11:30:00]
[11:30:00] <------------>
[11:30:00] Scheduling destruction of SIP dialog '20c45c634a987f2b-14679@10.35.144.11' in 32000 ms (Method: OPTIONS)
[11:30:01]        > 0x7f91c8032e50 -- Strict RTP learning complete - Locking on source address 000.00.136.133:37113
[11:30:01]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:01]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:01]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:02]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:03]   == SRTCP unprotect failed because of authentication failure
[11:30:03]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:03]     -- Called 8600060@default
[11:30:03]     -- Executing [8600060@default:1] MeetMe("Local/8600060@default-000006de;2", "8600060,F") in new stack
[11:30:03]     -- Local/8600060@default-000006de;1 answered
[11:30:03]     -- Executing [50127835560000@default:1] AGI("Local/8600060@default-000006de;1", "agi://127.0.0.1:4577/call_log") in new stack
[11:30:03]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTING))
[11:30:03]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPDTO=60))
[11:30:03]     -- <Local/8600060@default-000006de;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[11:30:03]     -- Executing [50127835560000@default:2] Dial("Local/8600060@default-000006de;1", "SIP/27835560000@CM_1,,tTo") in new stack
[11:30:03]   == Using SIP RTP CoS mark 5
[11:30:03] Audio is at 10384
[11:30:03] Adding codec g729 to SDP
[11:30:03] Adding codec ulaw to SDP
[11:30:03] Adding non-codec 0x1 (telephone-event) to SDP
[11:30:03] Reliably Transmitting (NAT) to 000.00.55.52:5060:
[11:30:03] INVITE sip:27835560000@000.00.55.52 SIP/2.0
[11:30:03] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK00712313;rport
[11:30:03] Max-Forwards: 70
[11:30:03] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:03] To: <sip:27835560000@000.00.55.52>
[11:30:03] Contact: <sip:27108240683@000.00.236.130:5060>
[11:30:03] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:03] CSeq: 102 INVITE
[11:30:03] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:03] Date: Thu, 10 Feb 2022 09:30:03 GMT
[11:30:03] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:03] Supported: replaces, timer
[11:30:03] Remote-Party-ID: "M2100930030000000008" <sip:27108240683@000.00.236.130>;party=calling;privacy=off;screen=no
[11:30:03] Content-Type: application/sdp
[11:30:03] Content-Length: 308
[11:30:03]
[11:30:03] v=0
[11:30:03] o=root 1874938296 1874938296 IN IP4 000.00.236.130
[11:30:03] s=Asterisk PBX 13.38.2-vici
[11:30:03] c=IN IP4 000.00.236.130
[11:30:03] t=0 0
[11:30:03] m=audio 10384 RTP/AVP 18 0 101
[11:30:03] a=rtpmap:18 G729/8000
[11:30:03] a=fmtp:18 annexb=no
[11:30:03] a=rtpmap:0 PCMU/8000
[11:30:03] a=rtpmap:101 telephone-event/8000
[11:30:03] a=fmtp:101 0-16
[11:30:03] a=ptime:20
[11:30:03] a=maxptime:150
[11:30:03] a=sendrecv
[11:30:03]
[11:30:03] ---
[11:30:03]     -- Called SIP/27835560000@CM_1
[11:30:03]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:03]     -- Called 58600060@default
[11:30:03]     -- Executing [58600060@default:1] MeetMe("Local/58600060@default-000006df;2", "8600060,Fmq") in new stack
[11:30:03]     -- Local/58600060@default-000006df;1 answered
[11:30:03]     -- Executing [8309@default:1] Answer("Local/58600060@default-000006df;1", "") in new stack
[11:30:03]     -- Executing [8309@default:2] Monitor("Local/58600060@default-000006df;1", "wav,20220210-093003_27835560000") in new stack
[11:30:03]     -- Executing [8309@default:3] Wait("Local/58600060@default-000006df;1", "3600") in new stack
[11:30:03]
[11:30:03] <--- SIP read from UDP:000.00.55.52:5060 --->
[11:30:03] SIP/2.0 407 Proxy Authentication Required
[11:30:03] Via: SIP/2.0/UDP 000.00.236.130:5060;rport=5060;received=000.00.236.130;branch=z9hG4bK00712313
[11:30:03] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:03] To: <sip:27835560000@000.00.55.52>;tag=3c0b8cba1117db0aa30ef691314fe7e2.61ec
[11:30:03] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:03] CSeq: 102 INVITE
[11:30:03] Proxy-Authenticate: Digest realm="sip.cm.nl", nonce="YgTbV2IE2xtswIOjaJzd41m/tXJAELXvjmKDe6MeMXNdd3gNlH2Frrlp9uiA", algorithm=MD5
[11:30:03] Content-Length: 0
[11:30:03]
[11:30:03] <------------->
[11:30:03] --- (8 headers 0 lines) ---
[11:30:03] Transmitting (NAT) to 000.00.55.52:5060:
[11:30:03] ACK sip:27835560000@000.00.55.52 SIP/2.0
[11:30:03] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK00712313;rport
[11:30:03] Max-Forwards: 70
[11:30:03] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:03] To: <sip:27835560000@000.00.55.52>;tag=3c0b8cba1117db0aa30ef691314fe7e2.61ec
[11:30:03] Contact: <sip:27108240683@000.00.236.130:5060>
[11:30:03] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:03] CSeq: 102 ACK
[11:30:03] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:03] Content-Length: 0
[11:30:03]
[11:30:03] ---
[11:30:03] Audio is at 10384
[11:30:03] Adding codec g729 to SDP
[11:30:03] Adding codec ulaw to SDP
[11:30:03] Adding non-codec 0x1 (telephone-event) to SDP
[11:30:03] Reliably Transmitting (NAT) to 000.00.55.52:5060:
[11:30:03] INVITE sip:27835560000@000.00.55.52 SIP/2.0
[11:30:03] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK57c426e3;rport
[11:30:03] Max-Forwards: 70
[11:30:03] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:03] To: <sip:27835560000@000.00.55.52>
[11:30:03] Contact: <sip:27108240683@000.00.236.130:5060>
[11:30:03] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:03] CSeq: 103 INVITE
[11:30:03] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:03] Proxy-Authorization: Digest username="CrispSol282", realm="sip.cm.nl", algorithm=MD5, uri="sip:27835560000@000.00.55.52", nonce="YgTbV2IE2xtswIOjaJzd41m/tXJAELXvjmKDe6MeMXNdd3gNlH2Frrlp9uiA", response="a97f368939cccd84046850c302aed84e"
[11:30:03] Date: Thu, 10 Feb 2022 09:30:03 GMT
[11:30:03] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:03] Supported: replaces, timer
[11:30:03] Remote-Party-ID: "M2100930030000000008" <sip:27108240683@000.00.236.130>;party=calling;privacy=off;screen=no
[11:30:03] Content-Type: application/sdp
[11:30:03] Content-Length: 308
[11:30:03]
[11:30:03] v=0
[11:30:03] o=root 1874938296 1874938297 IN IP4 000.00.236.130
[11:30:03] s=Asterisk PBX 13.38.2-vici
[11:30:03] c=IN IP4 000.00.236.130
[11:30:03] t=0 0
[11:30:03] m=audio 10384 RTP/AVP 18 0 101
[11:30:03] a=rtpmap:18 G729/8000
[11:30:03] a=fmtp:18 annexb=no
[11:30:03] a=rtpmap:0 PCMU/8000
[11:30:03] a=rtpmap:101 telephone-event/8000
[11:30:03] a=fmtp:101 0-16
[11:30:03] a=ptime:20
[11:30:03] a=maxptime:150
[11:30:03] a=sendrecv

[11:30:03]
[11:30:03] <--- SIP read from UDP:000.00.55.52:5060 --->
[11:30:03] SIP/2.0 100 Trying
[11:30:03] Via: SIP/2.0/UDP 000.00.236.130:5060;rport=5060;received=000.00.236.130;branch=z9hG4bK57c426e3
[11:30:03] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:03] To: <sip:27835560000@000.00.55.52>
[11:30:03] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:03] CSeq: 103 INVITE
[11:30:03] Content-Length: 0
[11:30:03]
[11:30:03] <------------->
[11:30:03] --- (7 headers 0 lines) ---
[11:30:04]
[11:30:04] <--- SIP read from UDP:000.00.184.99:5060 --->
[11:30:04] OPTIONS sip:000.00.236.130:5060 SIP/2.0
[11:30:04] Via: SIP/2.0/UDP 000.00.184.99;branch=z9hG4bK275d.34db54f1000000000000000000000000.0
[11:30:04] To: <sip:000.00.236.130:5060>
[11:30:04] From: <sip:keepalive@sip.cm.nl>;tag=4e57940b9af36d8200fe0f97d696f8eb-6793
[11:30:04] CSeq: 10 OPTIONS
[11:30:04] Call-ID: 731ed203582256a5-29659@10.25.144.12
[11:30:04] Max-Forwards: 70
[11:30:04] Content-Length: 0
[11:30:04]
[11:30:04] <------------->
[11:30:04] --- (8 headers 0 lines) ---
[11:30:04] Sending to 000.00.184.99:5060 (NAT)
[11:30:04] Looking for s in trunkinbound (domain 000.00.236.130)
[11:30:04]
[11:30:04] <--- Transmitting (NAT) to 000.00.184.99:5060 --->
[11:30:04] SIP/2.0 200 OK
[11:30:04] Via: SIP/2.0/UDP 000.00.184.99;branch=z9hG4bK275d.34db54f1000000000000000000000000.0;received=000.00.184.99;rport=5060
[11:30:04] From: <sip:keepalive@sip.cm.nl>;tag=4e57940b9af36d8200fe0f97d696f8eb-6793
[11:30:04] To: <sip:000.00.236.130:5060>;tag=as2ec81eec
[11:30:04] Call-ID: 731ed203582256a5-29659@10.25.144.12
[11:30:04] CSeq: 10 OPTIONS
[11:30:04] Server: Asterisk PBX 13.38.2-vici
[11:30:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:04] Supported: replaces, timer
[11:30:04] Contact: <sip:000.00.236.130:5060>
[11:30:04] Accept: application/sdp
[11:30:04] Content-Length: 0
[11:30:04] <------------>
[11:30:04] Scheduling destruction of SIP dialog '731ed203582256a5-29659@10.25.144.12' in 32000 ms (Method: OPTIONS)
[11:30:04]
[11:30:04] <--- SIP read from UDP:000.00.184.100:5060 --->
[11:30:04] OPTIONS sip:000.00.236.130:5060 SIP/2.0
[11:30:04] Via: SIP/2.0/UDP 000.00.184.100;branch=z9hG4bKb838.069c7e66000000000000000000000000.0
[11:30:04] To: <sip:000.00.236.130:5060>
[11:30:04] From: <sip:keepalive@sip.cm.nl>;tag=f7e997928c13343c407524a70e269f78-01fd
[11:30:04] CSeq: 10 OPTIONS
[11:30:04] Call-ID: 1194b6dc733000e1-30930@10.25.144.13
[11:30:04] Max-Forwards: 70
[11:30:04] Content-Length: 0
[11:30:04]
[11:30:04] <------------->
[11:30:04] --- (8 headers 0 lines) ---
[11:30:04] Sending to 000.00.184.100:5060 (NAT)
[11:30:04] Looking for s in trunkinbound (domain 000.00.236.130)
[11:30:04]
[11:30:04] <--- Transmitting (NAT) to 000.00.184.100:5060 --->
[11:30:04] SIP/2.0 200 OK
[11:30:04] Via: SIP/2.0/UDP 000.00.184.100;branch=z9hG4bKb838.069c7e66000000000000000000000000.0;received=000.00.184.100;rport=5060
[11:30:04] From: <sip:keepalive@sip.cm.nl>;tag=f7e997928c13343c407524a70e269f78-01fd
[11:30:04] To: <sip:000.00.236.130:5060>;tag=as5c49391d
[11:30:04] Call-ID: 1194b6dc733000e1-30930@10.25.144.13
[11:30:04] CSeq: 10 OPTIONS
[11:30:04] Server: Asterisk PBX 13.38.2-vici
[11:30:04] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:04] Supported: replaces, timer
[11:30:04] Contact: <sip:000.00.236.130:5060>
[11:30:04] Accept: application/sdp
[11:30:04] Content-Length: 0

[11:30:04] <------------>
[11:30:04] Scheduling destruction of SIP dialog '1194b6dc733000e1-30930@10.25.144.13' in 32000 ms (Method: OPTIONS)
[11:30:04]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:04]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:05] Reliably Transmitting (NAT) to 000.00.58.11:5060:
[11:30:05] OPTIONS sip:000.00.58.11 SIP/2.0
[11:30:05] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40e0b9a6;rport
[11:30:05] Max-Forwards: 70
[11:30:05] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as25d22c8d
[11:30:05] To: <sip:000.00.58.11>
[11:30:05] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:05] Call-ID: 639c68807aa14df626d84d795a8bbb90@000.00.236.130:5060
[11:30:05] CSeq: 102 OPTIONS
[11:30:05] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:05] Date: Thu, 10 Feb 2022 09:30:05 GMT
[11:30:05] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:05] Supported: replaces, timer
[11:30:05] Content-Length: 0
[11:30:05]
[11:30:05] <--- SIP read from UDP:000.00.55.52:5060 --->
[11:30:05] SIP/2.0 183 Session Progress
[11:30:05] Via: SIP/2.0/UDP 000.00.236.130:5060;rport=5060;received=000.00.236.130;branch=z9hG4bK57c426e3
[11:30:05] Record-Route: <sip:000.00.55.62:8635;lr;did=9ad.d6b>
[11:30:05] Record-Route: <sip:000.00.55.52;lr>
[11:30:05] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:05] To: <sip:27835560000@000.00.55.52>;tag=Q7ggc3HHSBtHN
[11:30:05] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:05] CSeq: 103 INVITE
[11:30:05] Contact: <sip:27835560000@000.00.55.49:10941;transport=udp>
[11:30:05] User-Agent: CMSBC
[11:30:05] Accept: application/sdp
[11:30:05] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
[11:30:05] Supported: timer, path, replaces
[11:30:05] Allow-Events: talk, hold, conference, refer
[11:30:05] Content-Disposition: session
[11:30:05] Content-Type: application/sdp
[11:30:05] Content-Length: 258
[11:30:05]
[11:30:05] v=0
[11:30:05] o=PortaSIP 4126630277 3798185784 IN IP4 000.00.55.54
[11:30:05] s=-
[11:30:05] t=0 0
[11:30:05] m=audio 16950 RTP/AVP 18 101
[11:30:05] c=IN IP4 000.00.55.54
[11:30:05] a=rtpmap:18 G729/8000
[11:30:05] a=fmtp:18 annexb=no
[11:30:05] a=rtpmap:101 telephone-event/8000
[11:30:05] a=fmtp:101 0-15
[11:30:05] a=ptime:20
[11:30:05] a=sendrecv
[11:30:05] a=rtcp:16951
[11:30:05] <------------->
[11:30:05] --- (17 headers 13 lines) ---
[11:30:05] sip_route_dump: route/path hop: <sip:000.00.55.52;lr>
[11:30:05] sip_route_dump: route/path hop: <sip:000.00.55.62:8635;lr;did=9ad.d6b>
[11:30:05] Got SDP version 3798185784 and unique parts [PortaSIP 4126630277 IN IP4 000.00.55.54]
[11:30:05] Found RTP audio format 18
[11:30:05] Found RTP audio format 101
[11:30:05] Found audio description format G729 for ID 18
[11:30:05] Found audio description format telephone-event for ID 101
[11:30:05] Capabilities: us - (g729|ulaw), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
[11:30:05] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[11:30:05]        > 0x7f91c4010700 -- Strict RTP learning after remote address set to: 000.00.55.54:16950
[11:30:05] Peer audio RTP is at port 000.00.55.54:16950
[11:30:05]     -- SIP/CM_1-000003cf is making progress passing it to Local/8600060@default-000006de;1
[11:30:05]        > 0x7f91c4010700 -- Strict RTP switching to RTP target address 000.00.55.54:16950 as source
[11:30:05]   == SRTCP unprotect failed because of authentication failure
[11:30:06] Retransmitting #1 (NAT) to 000.00.58.11:5060:
[11:30:06] OPTIONS sip:000.00.58.11 SIP/2.0
[11:30:06] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40e0b9a6;rport
[11:30:06] Max-Forwards: 70
[11:30:06] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as25d22c8d
[11:30:06] To: <sip:000.00.58.11>
[11:30:06] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:06] Call-ID: 639c68807aa14df626d84d795a8bbb90@000.00.236.130:5060
[11:30:06] CSeq: 102 OPTIONS
[11:30:06] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:06] Date: Thu, 10 Feb 2022 09:30:05 GMT
[11:30:06] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:06] Supported: replaces, timer
[11:30:06] Content-Length: 0
[11:30:06] ---
[11:30:06]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:06]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:07] Retransmitting #2 (NAT) to 000.00.58.11:5060:
[11:30:07] OPTIONS sip:000.00.58.11 SIP/2.0
[11:30:07] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40e0b9a6;rport
[11:30:07] Max-Forwards: 70
[11:30:07] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as25d22c8d
[11:30:07] To: <sip:000.00.58.11>
[11:30:07] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:07] Call-ID: 639c68807aa14df626d84d795a8bbb90@000.00.236.130:5060
[11:30:07] CSeq: 102 OPTIONS
[11:30:07] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:07] Date: Thu, 10 Feb 2022 09:30:05 GMT
[11:30:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:07] Supported: replaces, timer
[11:30:07] Content-Length: 0
[11:30:07] ---
[11:30:07] Reliably Transmitting (NAT) to 000.00.184.98:5060:
[11:30:07] OPTIONS sip:000.00.184.98 SIP/2.0
[11:30:07] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK4ff09458;rport
[11:30:07] Max-Forwards: 70
[11:30:07] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as676cde49
[11:30:07] To: <sip:000.00.184.98>
[11:30:07] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:07] Call-ID: 346911ce4c8c4ce76c49553666cb1b78@000.00.236.130:5060
[11:30:07] CSeq: 102 OPTIONS
[11:30:07] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:07] Date: Thu, 10 Feb 2022 09:30:07 GMT
[11:30:07] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:07] Supported: replaces, timer
[11:30:07] Content-Length: 0

[11:30:07] ---
[11:30:08]
[11:30:08] <--- SIP read from UDP:000.00.184.98:5060 --->
[11:30:08] SIP/2.0 200 OK
[11:30:08] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK4ff09458;rport=5060;received=000.00.236.130
[11:30:08] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as676cde49
[11:30:08] To: <sip:000.00.184.98>;tag=476c01069c86642d787608d1585f32a7.7682
[11:30:08] Call-ID: 346911ce4c8c4ce76c49553666cb1b78@000.00.236.130:5060
[11:30:08] CSeq: 102 OPTIONS
[11:30:08] Content-Length: 0
[11:30:08]
[11:30:08] <------------->
[11:30:08] --- (7 headers 0 lines) ---
[11:30:08] Really destroying SIP dialog '346911ce4c8c4ce76c49553666cb1b78@000.00.236.130:5060' Method: OPTIONS
[11:30:08]   == SRTCP unprotect failed because of authentication failure
[11:30:08] Retransmitting #3 (NAT) to 000.00.58.11:5060:
[11:30:08] OPTIONS sip:000.00.58.11 SIP/2.0
[11:30:08] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40e0b9a6;rport
[11:30:08] Max-Forwards: 70
[11:30:08] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as25d22c8d
[11:30:08] To: <sip:000.00.58.11>
[11:30:08] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:08] Call-ID: 639c68807aa14df626d84d795a8bbb90@000.00.236.130:5060
[11:30:08] CSeq: 102 OPTIONS
[11:30:08] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:08] Date: Thu, 10 Feb 2022 09:30:05 GMT
[11:30:08] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:08] Supported: replaces, timer
[11:30:08] Content-Length: 0
[11:30:08] ---
[11:30:08]
[11:30:08] <--- SIP read from UDP:000.00.55.52:5060 --->
[11:30:08] SIP/2.0 200 OK
[11:30:08] Via: SIP/2.0/UDP 000.00.236.130:5060;rport=5060;received=000.00.236.130;branch=z9hG4bK57c426e3
[11:30:08] Record-Route: <sip:000.00.55.62:8635;lr;did=9ad.d6b>
[11:30:08] Record-Route: <sip:000.00.55.52;lr>
[11:30:08] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:08] To: <sip:27835560000@000.00.55.52>;tag=Q7ggc3HHSBtHN
[11:30:08] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:08] CSeq: 103 INVITE
[11:30:08] Contact: <sip:27835560000@000.00.55.49:10941;transport=udp>
[11:30:08] User-Agent: CMSBC
[11:30:08] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
[11:30:08] Require: timer
[11:30:08] Supported: timer, path, replaces
[11:30:08] Allow-Events: talk, hold, conference, refer
[11:30:08] Session-Expires: 1800;refresher=uac
[11:30:08] Content-Disposition: session
[11:30:08] Content-Type: application/sdp
[11:30:08] Content-Length: 258
[11:30:08]
[11:30:08] v=0
[11:30:08] o=PortaSIP 4126630277 3798185784 IN IP4 000.00.55.54
[11:30:08] s=-
[11:30:08] t=0 0
[11:30:08] m=audio 16950 RTP/AVP 18 101
[11:30:08] c=IN IP4 000.00.55.54
[11:30:08] a=rtpmap:18 G729/8000
[11:30:08] a=fmtp:18 annexb=no
[11:30:08] a=rtpmap:101 telephone-event/8000
[11:30:08] a=fmtp:101 0-15
[11:30:08] a=ptime:20
[11:30:08] a=sendrecv
[11:30:08] a=rtcp:16951
[11:30:08] <------------->
[11:30:08] --- (18 headers 13 lines) ---
[11:30:08] Comparing SDP version 3798185784 -> 3798185784 and unique parts [PortaSIP 4126630277 IN IP4 000.00.55.54] -> [PortaSIP 4126630277 IN IP4 000.00.55.54]
[11:30:08] sip_route_dump: route/path hop: <sip:000.00.55.52;lr>
[11:30:08] sip_route_dump: route/path hop: <sip:000.00.55.62:8635;lr;did=9ad.d6b>
[11:30:08] Transmitting (NAT) to 000.00.55.52:5060:
[11:30:08] ACK sip:27835560000@000.00.55.49:10941;transport=udp SIP/2.0
[11:30:08] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK559db329;rport
[11:30:08] Route: <sip:000.00.55.52;lr>,<sip:000.00.55.62:8635;lr;did=9ad.d6b>
[11:30:08] Max-Forwards: 70
[11:30:08] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:08] To: <sip:27835560000@000.00.55.52>;tag=Q7ggc3HHSBtHN
[11:30:08] Contact: <sip:27108240683@000.00.236.130:5060>
[11:30:08] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:08] CSeq: 103 ACK
[11:30:08] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:08] Content-Length: 0
[11:30:08] ---
[11:30:08]     -- SIP/CM_1-000003cf answered Local/8600060@default-000006de;1
[11:30:08]     -- Channel SIP/CM_1-000003cf joined 'simple_bridge' basic-bridge <e918d934-7fb7-410c-b497-7a993baba3f1>
[11:30:08]     -- Channel Local/8600060@default-000006de;1 joined 'simple_bridge' basic-bridge <e918d934-7fb7-410c-b497-7a993baba3f1>
[11:30:09] Retransmitting #4 (NAT) to 000.00.58.11:5060:
[11:30:09] OPTIONS sip:000.00.58.11 SIP/2.0
[11:30:09] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK40e0b9a6;rport
[11:30:09] Max-Forwards: 70
[11:30:09] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as25d22c8d
[11:30:09] To: <sip:000.00.58.11>
[11:30:09] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:09] Call-ID: 639c68807aa14df626d84d795a8bbb90@000.00.236.130:5060
[11:30:09] CSeq: 102 OPTIONS
[11:30:09] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:09] Date: Thu, 10 Feb 2022 09:30:05 GMT
[11:30:09] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:09] Supported: replaces, timer
[11:30:09] Content-Length: 0
[11:30:09] ---
[11:30:09] Really destroying SIP dialog '639c68807aa14df626d84d795a8bbb90@000.00.236.130:5060' Method: OPTIONS
[11:30:10] Reliably Transmitting (NAT) to 000.00.185.110:5060:
[11:30:10] OPTIONS sip:000.00.185.110 SIP/2.0
[11:30:10] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK2463355d;rport
[11:30:10] Max-Forwards: 70
[11:30:10] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as65efc386
[11:30:10] To: <sip:000.00.185.110>
[11:30:10] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:10] Call-ID: 3effd61c417863e81e45b8630ae193b1@000.00.236.130:5060
[11:30:10] CSeq: 102 OPTIONS
[11:30:10] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:10] Date: Thu, 10 Feb 2022 09:30:10 GMT
[11:30:10] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:10] Supported: replaces, timer
[11:30:10] Content-Length: 0
[11:30:10] ---
[11:30:10]
[11:30:10] <--- SIP read from UDP:000.00.185.110:5060 --->
[11:30:10] SIP/2.0 200 OK
[11:30:10] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK2463355d;rport=5060;received=000.00.236.130
[11:30:10] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as65efc386
[11:30:10] To: <sip:000.00.185.110>;tag=448a51c14318c7ca50b4a1a4caa752e4.4933
[11:30:10] Call-ID: 3effd61c417863e81e45b8630ae193b1@000.00.236.130:5060
[11:30:10] CSeq: 102 OPTIONS
[11:30:10] Content-Length: 0
[11:30:10]
[11:30:10] <------------->
[11:30:10] --- (7 headers 0 lines) ---
[11:30:10] Really destroying SIP dialog '3effd61c417863e81e45b8630ae193b1@000.00.236.130:5060' Method: OPTIONS
[11:30:10]        > 0x7f91c4010700 -- Strict RTP learning complete - Locking on source address 000.00.55.54:16950
[11:30:11] Reliably Transmitting (NAT) to 000.00.185.98:5060:
[11:30:11] OPTIONS sip:000.00.185.98 SIP/2.0
[11:30:11] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK7b366f11;rport
[11:30:11] Max-Forwards: 70
[11:30:11] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as230ec008
[11:30:11] To: <sip:000.00.185.98>
[11:30:11] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:11] Call-ID: 438b12e36716c93a144c8d905a8a07c4@000.00.236.130:5060
[11:30:11] CSeq: 102 OPTIONS
[11:30:11] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:11] Date: Thu, 10 Feb 2022 09:30:11 GMT
[11:30:11] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:11] Supported: replaces, timer
[11:30:11] Content-Length: 0
[11:30:11] ---
[11:30:11]
[11:30:11] <--- SIP read from UDP:000.00.185.98:5060 --->
[11:30:11] SIP/2.0 200 OK
[11:30:11] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK7b366f11;rport=5060;received=000.00.236.130
[11:30:11] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as230ec008
[11:30:11] To: <sip:000.00.185.98>;tag=b05a6910a0b28c7dce61d7f74575ee68.cbcd0a91
[11:30:11] Call-ID: 438b12e36716c93a144c8d905a8a07c4@000.00.236.130:5060
[11:30:11] CSeq: 102 OPTIONS
[11:30:11] Content-Length: 0
[11:30:11]
[11:30:11] <------------->
[11:30:11] --- (7 headers 0 lines) ---
[11:30:11] Really destroying SIP dialog '438b12e36716c93a144c8d905a8a07c4@000.00.236.130:5060' Method: OPTIONS
[11:30:11]   == SRTCP unprotect failed because of authentication failure
[11:30:13]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:13]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600060@default-000006de;1
[11:30:13]     -- Channel Local/8600060@default-000006de;1 left 'simple_bridge' basic-bridge <e918d934-7fb7-410c-b497-7a993baba3f1>
[11:30:13]     -- Channel SIP/CM_1-000003cf left 'simple_bridge' basic-bridge <e918d934-7fb7-410c-b497-7a993baba3f1>
[11:30:13]   == Spawn extension (default, 50127835560000, 2) exited non-zero on 'Local/8600060@default-000006de;1'
[11:30:13]     -- Executing [h@default:1] AGI("Local/8600060@default-000006de;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----10-----SIP 200 OK)") in new stack
[11:30:13] Scheduling destruction of SIP dialog '085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060' in 6400 ms (Method: INVITE)
[11:30:13] Reliably Transmitting (NAT) to 000.00.55.52:5060:
[11:30:13] BYE sip:27835560000@000.00.55.49:10941;transport=udp SIP/2.0
[11:30:13] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK05011df6;rport
[11:30:13] Route: <sip:000.00.55.52;lr>,<sip:000.00.55.62:8635;lr;did=9ad.d6b>
[11:30:13] Max-Forwards: 70
[11:30:13] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:13] To: <sip:27835560000@000.00.55.52>;tag=Q7ggc3HHSBtHN
[11:30:13] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:13] CSeq: 104 BYE
[11:30:13] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:13] Proxy-Authorization: Digest username="CrispSol282", realm="sip.cm.nl", algorithm=MD5, uri="sip:27835560000@000.00.55.49:10941", nonce="YgTbV2IE2xtswIOjaJzd41m/tXJAELXvjmKDe6MeMXNdd3gNlH2Frrlp9uiA", response="11c1bea15b38c4a05b4dc01b1c3e50fe"
[11:30:13] X-Asterisk-HangupCause: Normal Clearing
[11:30:13] X-Asterisk-HangupCauseCode: 16
[11:30:13] Content-Length: 0
[11:30:13] ---
[11:30:13]     -- <Local/8600060@default-000006de;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----10-----SIP 200 OK) completed, returning 0
[11:30:13]   == Spawn extension (default, 8600060, 1) exited non-zero on 'Local/8600060@default-000006de;2'
[11:30:13] WARNING[6718][C-00000d7e]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[11:30:13]     -- Executing [h@default:1] AGI("Local/8600060@default-000006de;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[11:30:13]     -- <Local/8600060@default-000006de;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[11:30:13]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:13] NOTICE[6745]: manager.c:4471 action_hangup: Request to hangup non-existent channel: Local/8600060@default-000006de;2
[11:30:13]
[11:30:13] <--- SIP read from UDP:000.00.55.52:5060 --->
[11:30:13] SIP/2.0 200 OK
[11:30:13] Via: SIP/2.0/UDP 000.00.236.130:5060;rport=5060;received=000.00.236.130;branch=z9hG4bK05011df6
[11:30:13] From: "M2100930030000000008" <sip:27108240683@000.00.236.130>;tag=as6a010f79
[11:30:13] To: <sip:27835560000@000.00.55.52>;tag=Q7ggc3HHSBtHN
[11:30:13] Call-ID: 085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060
[11:30:13] CSeq: 104 BYE
[11:30:13] User-Agent: CMSBC
[11:30:13] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
[11:30:13] Supported: timer, path, replaces
[11:30:13] Content-Length: 0
[11:30:13]
[11:30:13] <------------->
[11:30:13] --- (10 headers 0 lines) ---
[11:30:13] Really destroying SIP dialog '085a0ca31c16b4fe5e99b3f8552033b0@000.00.236.130:5060' Method: INVITE
[11:30:13]   == Manager 'sendcron' logged on from 127.0.0.1
[11:30:13]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/58600060@default-000006df;2
[11:30:13]   == Spawn extension (default, 58600060, 1) exited non-zero on 'Local/58600060@default-000006df;2'
[11:30:13] WARNING[6722][C-00000d80]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[11:30:13]     -- Executing [h@default:1] AGI("Local/58600060@default-000006df;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[11:30:13]     -- <Local/58600060@default-000006df;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
[11:30:13]   == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600060@default-000006df;1'
[11:30:13] WARNING[6721][C-00000d81]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[11:30:13]     -- Executing [h@default:1] AGI("Local/58600060@default-000006df;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[11:30:13]     -- <Local/58600060@default-000006df;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
[11:30:14]   == SRTCP unprotect failed because of authentication failure
[11:30:14]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:14]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:14]   == Manager 'sendcron' logged off from 127.0.0.1
[11:30:15] Reliably Transmitting (NAT) to 000.00.55.52:5060:
[11:30:15] OPTIONS sip:000.00.55.52 SIP/2.0
[11:30:15] Via: SIP/2.0/UDP 000.00.236.130:5060;branch=z9hG4bK58f2985f;rport
[11:30:15] Max-Forwards: 70
[11:30:15] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as59755881
[11:30:15] To: <sip:000.00.55.52>
[11:30:15] Contact: <sip:asterisk@000.00.236.130:5060>
[11:30:15] Call-ID: 2ddaf9115d046f8c0e1a1b8e4ca1b498@000.00.236.130:5060
[11:30:15] CSeq: 102 OPTIONS
[11:30:15] User-Agent: Asterisk PBX 13.38.2-vici
[11:30:15] Date: Thu, 10 Feb 2022 09:30:15 GMT
[11:30:15] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[11:30:15] Supported: replaces, timer
[11:30:15] Content-Length: 0
[11:30:15] <--- SIP read from UDP:000.00.55.52:5060 --->
[11:30:15] SIP/2.0 200 OK
[11:30:15] Via: SIP/2.0/UDP 000.00.236.130:5060;rport=5060;received=000.00.236.130;branch=z9hG4bK58f2985f
[11:30:15] From: "asterisk" <sip:asterisk@000.00.236.130>;tag=as59755881
[11:30:15] To: <sip:000.00.55.52>;tag=3c0b8cba1117db0aa30ef691314fe7e2.6e55
[11:30:15] Call-ID: 2ddaf9115d046f8c0e1a1b8e4ca1b498@000.00.236.130:5060
[11:30:15] CSeq: 102 OPTIONS
[11:30:15] Content-Length: 0
[11:30:15]
[11:30:15] <------------->
[11:30:15] --- (7 headers 0 lines) ---
[11:30:15] Really destroying SIP dialog '2ddaf9115d046f8c0e1a1b8e4ca1b498@000.00.236.130:5060' Method: OPTIONS
[11:30:16]   == SRTCP unprotect failed because of authentication failure
innotrend*CLI> exit
[11:30:18] Asterisk cleanly ending (0).
[11:30:18] Executing last minute cleanups



try changing your context in your dialplan for the carrier to context=default or context=outbound

Code: Select all

[Feb 10 11:28:46] ERROR[2349]: chan_sip.c:4309 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
innotrend*CLI> exit
[Feb 10 11:28:49] Asterisk cleanly ending (0).
[Feb 10 11:28:49] Executing last minute cleanups

Code: Select all

Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/8600060@default-000006de;1

It's hanging up a local channel. Channels shouldn't be local when audio has begun. I didn't see the handshake for choice of codec in the SIP debug. Sometimes they don't have a viable protocol available shared between the two sides of the call. Sometimes the vicidial system is using the wrong configuration files and never fully activates audio (resulting in a "local" channel which vicidial interprets as not really answered, no audio, and kills the call.

So: What version of Asterisk is listed in /etc/astguiclient.conf? What version in admin->servers for this server, and what version is actually installed?