vicidial.org

Hello;

vicibox 8.1.2, vicidial 2.14-714a, Build 190628-1511, asterisk 13.24.1-vici, Single Machine

Please, this is a major problem need really help to resolve it because it is effecting a lot in the work. I am not able to do transfer from agent interface using TRANSFER-CONF button to internal extension (by placing the extension in the NUMBER TO CALL field and then pressing the DIAL WITH CUSTOMER button), then once the destination answered the call, then the call is hanged up and the agent see this message: xfer line hangup (although the destination did not hangup at all).

Below is the most important part (once extension 2204 answered the transferred call) of the sip debug and below it the detailed log starting from sending the call for the agent until the end of transfer for the extension 2204:

The summary sip debug (once extension 2204 answered the call and until the left 'simple_bridge' basic-bridge debug):

Code: Select all

[Oct 31 20:13:04]     -- SIP/2204-00000185 answered Local/2204@default-00000192;                                                                                                             2
[Oct 31 20:13:04]     -- Local/2204@default-00000192;1 answered
[Oct 31 20:13:04]     -- Executing [8600061@default:1] MeetMe("Local/2204@default-00000192;1", "8600061,F") in new stack
[Oct 31 20:13:04]     -- Channel SIP/2204-00000185 joined 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>
[Oct 31 20:13:04]     -- Channel Local/2204@default-00000192;2 joined 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>
[Oct 31 20:13:04]        > 0x7fa9f80164d0 -- Strict RTP switching to RTP target address 192.168.1.94:5062 as source
[Oct 31 20:13:05]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 20:13:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 20:13:06]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel:SIP/2204-00000185
[Oct 31 20:13:06]     -- Channel SIP/2204-00000185 left 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>
[Oct 31 20:13:06] Scheduling destruction of SIP dialog '7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060' in 6400 ms (Method: INVITE)
[Oct 31 20:13:06]     -- Channel Local/2204@default-00000192;2 left 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>


The long detailed debug:

Code: Select all

[Oct 31 20:12:36]     -- Called 58600061@default
[Oct 31 20:12:36]     -- Executing [58600061@default:1] MeetMe("Local/58600061@default-00000190;2", "8600061,Fmq") in new stack
[Oct 31 20:12:36]     -- Local/58600061@default-00000190;1 answered
[Oct 31 20:12:36]     -- Executing [8309@default:1] Answer("Local/58600061@default-00000190;1", "") in new stack
[Oct 31 20:12:36]     -- Executing [8309@default:2] Monitor("Local/58600061@default-00000190;1", "wav,COLLECT_InGroup_22524998_Staff-22524998_20191031-201236_24                                                                                                             781622") in new stack
[Oct 31 20:12:36]     -- Executing [8309@default:3] Wait("Local/58600061@default-00000190;1", "3600") in new stack
[Oct 31 20:12:36]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 20:12:36]     -- Called 192*168*001*020*78600061@default
[Oct 31 20:12:36]     -- Executing [192*168*001*020*78600061@default:1] Goto("Local/192*168*001*020*78600061@default-00000191;2", "default,78600061,1") in new stack
[Oct 31 20:12:36]     -- Goto (default,78600061,1)
[Oct 31 20:12:36]     -- Executing [78600061@default:1] MeetMe("Local/192*168*001*020*78600061@default-00000191;2", "8600061,Fq") in new stack
[Oct 31 20:12:36]     -- Local/192*168*001*020*78600061@default-00000191;1 answered
[Oct 31 20:12:36]     -- Executing [83047777777777@vicidial-auto:1] Answer("Local/192*168*001*020*78600061@default-00000191;1", "") in new stack
[Oct 31 20:12:36]     -- Executing [83047777777777@vicidial-auto:2] Playback("Local/192*168*001*020*78600061@default-00000191;1", "ding") in new stack
[Oct 31 20:12:36]     -- <Local/192*168*001*020*78600061@default-00000191;1> Playing 'ding.gsm' (language 'en')
[Oct 31 20:12:36]     -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/192*168*001*020*78600061@default-00000191;1", "") in new stack
[Oct 31 20:12:36]   == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/192*168*001*020*78600061@default-00000191;1'
[Oct 31 20:12:36] WARNING[23868][C-00000384]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Oct 31 20:12:36]     -- Executing [h@vicidial-auto:1] AGI("Local/192*168*001*020*78600061@default-00000191;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Oct 31 20:12:36]     -- <Local/192*168*001*020*78600061@default-00000191;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Oct 31 20:12:36]   == Spawn extension (default, 78600061, 1) exited non-zero on                                                                                                              'Local/192*168*001*020*78600061@default-00000191;2' [Oct 31 20:12:36] WARNING[23869][C-00000383]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel [Oct 31 20:12:36]     -- Executing [h@default:1] AGI("Local/192*168*001*020*78600061@default-00000191;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODE BUG-----16--------------------)") in new stack
[Oct 31 20:12:36]     -- <Local/192*168*001*020*78600061@default-00000191;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Oct 31 20:12:37]     -- Stopped music on hold on SIP/ooredoo-00000184
[Oct 31 20:12:37]     -- <SIP/ooredoo-00000184> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 20:12:37]     -- <SIP/ooredoo-00000184> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 20:12:37]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 20:12:37]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 20:12:37]     -- <SIP/ooredoo-00000184> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 20:12:38]     -- <SIP/ooredoo-00000184> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 20:12:38]     -- <SIP/ooredoo-00000184> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 20:12:38]     -- <SIP/ooredoo-00000184> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Oct 31 20:12:38]     -- <SIP/ooredoo-00000184>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Oct 31 20:12:38]     -- Executing [192*168*001*020*8600061@default:1] Goto("SIP/ooredoo-00000184", "default,8600061,1") in new stack
[Oct 31 20:12:38]     -- Goto (default,8600061,1)
[Oct 31 20:12:38]     -- Executing [8600061@default:1] MeetMe("SIP/ooredoo-00000184", "8600061,F") in new stack
[Oct 31 20:12:40] Reliably Transmitting (NAT) to 192.168.1.94:5060:
[Oct 31 20:12:40] OPTIONS sip:2204@192.168.1.94:5060 SIP/2.0
[Oct 31 20:12:40] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3fcd2a19;rpor                                                                                                             t
[Oct 31 20:12:40] Max-Forwards: 70
[Oct 31 20:12:40] From: "asterisk" <sip:asterisk@192.168.1.20>;tag=as1b940e92
[Oct 31 20:12:40] To: <sip:2204@192.168.1.94:5060>
[Oct 31 20:12:40] Contact: <sip:asterisk@192.168.1.20:5060>
[Oct 31 20:12:40] Call-ID: 4f81d4af48db9db40f03dbed223efb98@192.168.1.20:5060
[Oct 31 20:12:40] CSeq: 102 OPTIONS
[Oct 31 20:12:40] User-Agent: Asterisk PBX 13.24.1-vici
[Oct 31 20:12:40] Date: Thu, 31 Oct 2019 16:12:40 GMT
[Oct 31 20:12:40] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Oct 31 20:12:40] Supported: replaces, timer
[Oct 31 20:12:40] Content-Length: 0
[Oct 31 20:12:40]
[Oct 31 20:12:40]
[Oct 31 20:12:40] ---
[Oct 31 20:12:40]
[Oct 31 20:12:40] <--- SIP read from UDP:192.168.1.94:5060 --->
[Oct 31 20:12:40] SIP/2.0 200 OK
[Oct 31 20:12:40] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK3fcd2a19;rport=5060
[Oct 31 20:12:40] From: "asterisk" <sip:asterisk@192.168.1.20>;tag=as1b940e92
[Oct 31 20:12:40] To: <sip:2204@192.168.1.94:5060>;tag=8077bb4567fae911a4c6b625910707be
[Oct 31 20:12:40] Call-ID: 4f81d4af48db9db40f03dbed223efb98@192.168.1.20:5060
[Oct 31 20:12:40] CSeq: 102 OPTIONS
[Oct 31 20:12:40] Contact: <sip:2204@192.168.1.94:5060>
[Oct 31 20:12:40] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
[Oct 31 20:12:40] Server: SIPPER for PhonerLite
[Oct 31 20:12:40] Content-Length: 0
[Oct 31 20:12:40]
[Oct 31 20:12:40] <------------->
[Oct 31 20:12:40] --- (10 headers 0 lines) ---
[Oct 31 20:12:40] Really destroying SIP dialog '4f81d4af48db9db40f03dbed223efb98@192.168.1.20:5060' Method: OPTIONS
[Oct 31 20:12:56]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 20:12:56]     -- Called 2204@default
[Oct 31 20:12:56]     -- Executing [2204@default:1] Dial("Local/2204@default-00000192;2", "SIP/2204,60,") in new stack
[Oct 31 20:12:56]   == Using SIP RTP CoS mark 5
[Oct 31 20:12:56] Audio is at 14896
[Oct 31 20:12:56] Adding codec alaw to SDP
[Oct 31 20:12:56] Adding codec ulaw to SDP
[Oct 31 20:12:56] Adding non-codec 0x1 (telephone-event) to SDP
[Oct 31 20:12:56] Reliably Transmitting (NAT) to 192.168.1.94:5060:
[Oct 31 20:12:56] INVITE sip:2204@192.168.1.94:5060 SIP/2.0
[Oct 31 20:12:56] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK1dd6db5d;rport
[Oct 31 20:12:56] Max-Forwards: 70
[Oct 31 20:12:56] From: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;tag=as06ec8756
[Oct 31 20:12:56] To: <sip:2204@192.168.1.94:5060>
[Oct 31 20:12:56] Contact: <sip:0000000000@192.168.1.20:5060>
[Oct 31 20:12:56] Call-ID: 7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060
[Oct 31 20:12:56] CSeq: 102 INVITE
[Oct 31 20:12:56] User-Agent: Asterisk PBX 13.24.1-vici
[Oct 31 20:12:56] Date: Thu, 31 Oct 2019 16:12:56 GMT
[Oct 31 20:12:56] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Oct 31 20:12:56] Supported: replaces, timer
[Oct 31 20:12:56] Remote-Party-ID: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;party=calling;privacy=off;screen=no
[Oct 31 20:12:56] Content-Type: application/sdp
[Oct 31 20:12:56] Content-Length: 279
[Oct 31 20:12:56]
[Oct 31 20:12:56] v=0
[Oct 31 20:12:56] o=root 217924584 217924584 IN IP4 192.168.1.20
[Oct 31 20:12:56] s=Asterisk PBX 13.24.1-vici
[Oct 31 20:12:56] c=IN IP4 192.168.1.20
[Oct 31 20:12:56] t=0 0
[Oct 31 20:12:56] m=audio 14896 RTP/AVP 8 0 101
[Oct 31 20:12:56] a=rtpmap:8 PCMA/8000
[Oct 31 20:12:56] a=rtpmap:0 PCMU/8000
[Oct 31 20:12:56] a=rtpmap:101 telephone-event/8000
[Oct 31 20:12:56] a=fmtp:101 0-16
[Oct 31 20:12:56] a=ptime:20
[Oct 31 20:12:56] a=maxptime:150
[Oct 31 20:12:56] a=sendrecv
[Oct 31 20:12:56]
[Oct 31 20:12:56] ---
[Oct 31 20:12:56]     -- Called SIP/2204
[Oct 31 20:12:56]
[Oct 31 20:12:56] <--- SIP read from UDP:192.168.1.94:5060 --->
[Oct 31 20:12:56] SIP/2.0 100 Trying
[Oct 31 20:12:56] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK1dd6db5d;rport=5060
[Oct 31 20:12:56] From: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;tag=as06ec8756
[Oct 31 20:12:56] To: <sip:2204@192.168.1.94:5060>
[Oct 31 20:12:56] Call-ID: 7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060
[Oct 31 20:12:56] CSeq: 102 INVITE
[Oct 31 20:12:56] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
[Oct 31 20:12:56] Server: SIPPER for PhonerLite
[Oct 31 20:12:56] Content-Length: 0
[Oct 31 20:12:56]
[Oct 31 20:12:56] <------------->
[Oct 31 20:12:56] --- (9 headers 0 lines) ---
[Oct 31 20:12:56]
[Oct 31 20:12:56] <--- SIP read from UDP:192.168.1.94:5060 --->
[Oct 31 20:12:56] SIP/2.0 180 Ringing
[Oct 31 20:12:56] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK1dd6db5d;rport=5060
[Oct 31 20:12:56] From: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;tag=as06ec8756
[Oct 31 20:12:56] To: <sip:2204@192.168.1.94:5060>;tag=0076dd4f67fae911a4c6b625910707be
[Oct 31 20:12:56] Call-ID: 7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060
[Oct 31 20:12:56] CSeq: 102 INVITE
[Oct 31 20:12:56] Contact: <sip:2204@192.168.1.94:5060;gr=8009CB47-5AFA-E911-A4B2-B625910707BE>
[Oct 31 20:12:56] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
[Oct 31 20:12:56] Supported: 100rel, replaces, from-change, gruu
[Oct 31 20:12:56] Server: SIPPER for PhonerLite
[Oct 31 20:12:56] Content-Length: 0
[Oct 31 20:12:56]
[Oct 31 20:12:56] <------------->
[Oct 31 20:12:56] --- (11 headers 0 lines) ---
[Oct 31 20:12:56] sip_route_dump: route/path hop: <sip:2204@192.168.1.94:5060;gr=8009CB47-5AFA-E911-A4B2-B625910707BE>
[Oct 31 20:12:56]     -- SIP/2204-00000185 is ringing
[Oct 31 20:12:56]     -- Local/2204@default-00000192;1 is ringing
[Oct 31 20:13:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 20:13:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 20:13:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 20:13:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 20:13:04]
[Oct 31 20:13:04] <--- SIP read from UDP:192.168.1.94:5060 --->
[Oct 31 20:13:04] SIP/2.0 200 OK
[Oct 31 20:13:04] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK1dd6db5d;rport=5060
[Oct 31 20:13:04] From: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;tag=as06ec8756
[Oct 31 20:13:04] To: <sip:2204@192.168.1.94:5060>;tag=0076dd4f67fae911a4c6b625910707be
[Oct 31 20:13:04] Call-ID: 7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060
[Oct 31 20:13:04] CSeq: 102 INVITE
[Oct 31 20:13:04] Contact: <sip:2204@192.168.1.94:5060;gr=8009CB47-5AFA-E911-A4B2-B625910707BE>
[Oct 31 20:13:04] Content-Type: application/sdp
[Oct 31 20:13:04] Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
[Oct 31 20:13:04] Supported: 100rel, replaces, from-change, gruu
[Oct 31 20:13:04] Server: SIPPER for PhonerLite
[Oct 31 20:13:04] Content-Length: 254
[Oct 31 20:13:04]
[Oct 31 20:13:04] v=0
[Oct 31 20:13:04] o=- 175443441 1 IN IP4 192.168.1.94
[Oct 31 20:13:04] s=SIPPER for PhonerLite
[Oct 31 20:13:04] c=IN IP4 192.168.1.94
[Oct 31 20:13:04] t=0 0
[Oct 31 20:13:04] m=audio 5062 RTP/AVP 8 0 101
[Oct 31 20:13:04] a=rtpmap:8 PCMA/8000
[Oct 31 20:13:04] a=rtpmap:0 PCMU/8000
[Oct 31 20:13:04] a=rtpmap:101 telephone-event/8000
[Oct 31 20:13:04] a=fmtp:101 0-16
[Oct 31 20:13:04] a=ssrc:3780825374
[Oct 31 20:13:04] a=sendrecv
[Oct 31 20:13:04] <------------->
[Oct 31 20:13:04] --- (12 headers 12 lines) ---
[Oct 31 20:13:04] Found RTP audio format 8
[Oct 31 20:13:04] Found RTP audio format 0
[Oct 31 20:13:04] Found RTP audio format 101
[Oct 31 20:13:04] Found audio description format PCMA for ID 8
[Oct 31 20:13:04] Found audio description format PCMU for ID 0
[Oct 31 20:13:04] Found audio description format telephone-event for ID 101
[Oct 31 20:13:04] Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
[Oct 31 20:13:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Oct 31 20:13:04]        > 0x7fa9f80164d0 -- Strict RTP learning after remote address set to: 192.168.1.94:5062
[Oct 31 20:13:04] Peer audio RTP is at port 192.168.1.94:5062
[Oct 31 20:13:04] sip_route_dump: route/path hop: <sip:2204@192.168.1.94:5060;gr=8009CB47-5AFA-E911-A4B2-B625910707BE>
[Oct 31 20:13:04] Transmitting (NAT) to 192.168.1.94:5060:
[Oct 31 20:13:04] ACK sip:2204@192.168.1.94:5060;gr=8009CB47-5AFA-E911-A4B2-B625910707BE SIP/2.0
[Oct 31 20:13:04] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK1e6def48;rport
[Oct 31 20:13:04] Max-Forwards: 70
[Oct 31 20:13:04] From: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;tag=as06ec8756
[Oct 31 20:13:04] To: <sip:2204@192.168.1.94:5060>;tag=0076dd4f67fae911a4c6b625910707be
[Oct 31 20:13:04] Contact: <sip:0000000000@192.168.1.20:5060>
[Oct 31 20:13:04] Call-ID: 7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060
[Oct 31 20:13:04] CSeq: 102 ACK
[Oct 31 20:13:04] User-Agent: Asterisk PBX 13.24.1-vici
[Oct 31 20:13:04] Content-Length: 0
[Oct 31 20:13:04]
[Oct 31 20:13:04]
[Oct 31 20:13:04] ---
[Oct 31 20:13:04]     -- SIP/2204-00000185 answered Local/2204@default-00000192;                                                                                                             2
[Oct 31 20:13:04]     -- Local/2204@default-00000192;1 answered
[Oct 31 20:13:04]     -- Executing [8600061@default:1] MeetMe("Local/2204@default-00000192;1", "8600061,F") in new stack
[Oct 31 20:13:04]     -- Channel SIP/2204-00000185 joined 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>
[Oct 31 20:13:04]     -- Channel Local/2204@default-00000192;2 joined 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>
[Oct 31 20:13:04]        > 0x7fa9f80164d0 -- Strict RTP switching to RTP target address 192.168.1.94:5062 as source
[Oct 31 20:13:05]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 31 20:13:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 31 20:13:06]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel:SIP/2204-00000185
[Oct 31 20:13:06]     -- Channel SIP/2204-00000185 left 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>
[Oct 31 20:13:06] Scheduling destruction of SIP dialog '7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060' in 6400 ms (Method: INVITE)
[Oct 31 20:13:06]     -- Channel Local/2204@default-00000192;2 left 'simple_bridge' basic-bridge <7834dcfd-6550-4f65-97a2-0b26b85b3561>
[Oct 31 20:13:06]   == Spawn extension (default, 2204, 1) exited non-zero on 'Local/2204@default-00000192;2'
[Oct 31 20:13:06] Reliably Transmitting (NAT) to 192.168.1.94:5060:
[Oct 31 20:13:06] BYE sip:2204@192.168.1.94:5060;gr=8009CB47-5AFA-E911-A4B2-B625910707BE SIP/2.0
[Oct 31 20:13:06] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK2ff578bf;rport
[Oct 31 20:13:06] Max-Forwards: 70
[Oct 31 20:13:06] From: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;tag=as06ec8756
[Oct 31 20:13:06] To: <sip:2204@192.168.1.94:5060>;tag=0076dd4f67fae911a4c6b625910707be
[Oct 31 20:13:06] Call-ID: 7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060
[Oct 31 20:13:06] CSeq: 103 BYE
[Oct 31 20:13:06] User-Agent: Asterisk PBX 13.24.1-vici
[Oct 31 20:13:06] X-Asterisk-HangupCause: Normal Clearing
[Oct 31 20:13:06] X-Asterisk-HangupCauseCode: 16
[Oct 31 20:13:06] Content-Length: 0
[Oct 31 20:13:06]
[Oct 31 20:13:06]
[Oct 31 20:13:06] ---
[Oct 31 20:13:06]     -- Executing [h@default:1] AGI("Local/2204@default-00000192;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----1-----SIP 200 OK)") in new stack
[Oct 31 20:13:06]
[Oct 31 20:13:06] <--- SIP read from UDP:192.168.1.94:5060 --->
[Oct 31 20:13:06] SIP/2.0 200 OK
[Oct 31 20:13:06] Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK2ff578bf;rport=5060
[Oct 31 20:13:06] From: "DC538163W0000000062W" <sip:0000000000@192.168.1.20>;tag=as06ec8756
[Oct 31 20:13:06] To: <sip:2204@192.168.1.94:5060>;tag=0076dd4f67fae911a4c6b625910707be
[Oct 31 20:13:06] Call-ID: 7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060
[Oct 31 20:13:06] CSeq: 103 BYE
[Oct 31 20:13:06] Contact: <sip:2204@192.168.1.94:5060;gr=8009CB47-5AFA-E911-A4B2-B625910707BE>
[Oct 31 20:13:06] Server: SIPPER for PhonerLite
[Oct 31 20:13:06] Content-Length: 0
[Oct 31 20:13:06]
[Oct 31 20:13:06] <------------->
[Oct 31 20:13:06] --- (9 headers 0 lines) ---
[Oct 31 20:13:06] Really destroying SIP dialog '7350d58a0b8d3d611903c14d23b14a13@192.168.1.20:5060' Method: INVITE
[Oct 31 20:13:06]     -- <Local/2204@default-00000192;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----1-----SIP 200 OK) completed, returning 0
[Oct 31 20:13:06]   == Spawn extension (default, 8600061, 1) exited non-zero on 'Local/2204@default-00000192;1'
[Oct 31 20:13:06] WARNING[23898][C-00000386]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Oct 31 20:13:06]     -- Executing [h@default:1] AGI("Local/2204@default-00000192;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Oct 31 20:13:06]     -- <Local/2204@default-00000192;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) complete                                                                                                             d, returning 0
[Oct 31 20:13:07]   == Manager 'sendcron' logged off from 127.0.0.1

Please note that this problem was not existed in previous versions (like vicibox version 7).

Also below is the link of a post for the same problem:

viewtopic.php?f=4&t=37714

Also there is one more note I need to mention it, that vicibox version 8.1.2 is using asterisk version 13.21.1-vici, but at certain time, it was installing asterisk version 13.24.1-vici which I placed the above log for it, and both asterisk versions are having the same problem, but in asterisk version 13.21.1-vici, the call is hanged up automatically after 1st ring at the destination and before the destination extension answered. Again, it is a problem when doing transfer for extension.

Regards
Bilal

Hello ALL;

Please, how can we resolve this?
I need help.

Regards
Bilal

I am currently having this same issue, i guess no one made any siggestions?

Haven't heard of this issue from any of our clients, but have you tried using a new dial prefix for these calls to send them through the loopback IAX trunk?

ok so i have the transfer going to local closer, with an inbound group created for closers.....then when i hit "local closer" it sends the call to the closer, but immediately disconnects the call on the closer side, here is the cli output:

Code: Select all

[Aug 18 14:07:14]     -- Registered SIP '1002' at 76.110.127.205:62943
[Aug 18 14:07:14]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:07:14]   == Using SIP RTP CoS mark 5
[Aug 18 14:07:14]     -- Called 1002
[Aug 18 14:07:14]     -- SIP/1002-00000006 is ringing
[Aug 18 14:07:15]        > 0x7efd10006b80 -- Strict RTP learning after remote address set to: 76.110.127.205:55012
[Aug 18 14:07:15]     -- SIP/1002-00000006 answered
[Aug 18 14:07:15]     -- Executing [8600052@meetme-enter-login:1] MeetMe("SIP/1002-00000006", "8600052,FG(what-are-you-wearing)") in new stack
[Aug 18 14:07:15]     -- Created MeetMe conference 1021 for conference '8600052'
[Aug 18 14:07:15]     -- <SIP/1002-00000006> Playing 'what-are-you-wearing.slin' (language 'en')
[Aug 18 14:07:15]        > 0x7efd10006b80 -- Strict RTP learning after ICE completion
[Aug 18 14:07:15]        > 0x7efd10006b80 -- Strict RTP switching to RTP target address 76.110.127.205:55012 as source
[Aug 18 14:07:16]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:07:16]     -- <SIP/1002-00000006> Playing 'conf-onlyperson.gsm' (language 'en')
[Aug 18 14:07:18]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:07:18]   == Using SIP RTP CoS mark 5
[Aug 18 14:07:18]     -- Called 1002
[Aug 18 14:07:18]     -- SIP/1002-00000007 is ringing
[Aug 18 14:07:18]     -- Got SIP response 486 "Busy Here" back from 76.110.127.205:62943
[Aug 18 14:07:18]     -- SIP/1002-00000007 is busy
[Aug 18 14:07:19]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:07:20]        > 0x7efd10006b80 -- Strict RTP learning complete - Locking on source address 76.110.127.205:55012
[Aug 18 14:07:45]   == WebSocket connection from '76.110.127.205:62959' for protocol 'sip' accepted using version '13'
[Aug 18 14:07:46]     -- Registered SIP '9000' at 76.xxx.127.205:62959
[Aug 18 14:07:46]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:07:46]   == Using SIP RTP CoS mark 5
[Aug 18 14:07:46]     -- Called 9000
[Aug 18 14:07:46]     -- SIP/9000-00000008 is ringing
[Aug 18 14:07:47]     -- SIP/9000-00000008 answered
[Aug 18 14:07:47]     -- Executing [8600054@meetme-enter-login:1] MeetMe("SIP/9000-00000008", "8600054,FG(what-are-you-wearing)") in new stack
[Aug 18 14:07:47]     -- Created MeetMe conference 1020 for conference '8600054'
[Aug 18 14:07:47]     -- <SIP/9000-00000008> Playing 'what-are-you-wearing.slin' (language 'en')
[Aug 18 14:07:47]        > 0x7efd30006e00 -- Strict RTP learning after ICE completion
[Aug 18 14:07:47]        > 0x7efd30006e00 -- Strict RTP switching to RTP target address 76.xxx.127.205:49237 as source
[Aug 18 14:07:48]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:07:48]     -- <SIP/9000-00000008> Playing 'conf-onlyperson.gsm' (language 'en')
[Aug 18 14:07:50]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:07:50]   == Using SIP RTP CoS mark 5
[Aug 18 14:07:50]     -- Called 9000
[Aug 18 14:07:50]     -- SIP/9000-00000009 is ringing
[Aug 18 14:07:50]     -- Got SIP response 486 "Busy Here" back from 76.xxx.127.205:62959
[Aug 18 14:07:50]     -- SIP/9000-00000009 is busy
[Aug 18 14:07:51]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:07:52]        > 0x7efd30006e00 -- Strict RTP learning complete - Locking on source address 76.xxx.127.205:49237
[Aug 18 14:07:56]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:07:56]     -- Called 8600053@default
[Aug 18 14:07:56]     -- Executing [8600053@default:1] MeetMe("Local/8600053@default-00000001;2", "8600053,F") in new stack
[Aug 18 14:07:56]     -- Local/8600053@default-00000001;1 answered
[Aug 18 14:07:56]     -- Executing [519549477572@default:1] AGI("Local/8600053@default-00000001;1", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 18 14:07:56]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=MAIN))
[Aug 18 14:07:56]     -- <Local/8600053@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 18 14:07:56]     -- Executing [519549477572@default:2] Dial("Local/8600053@default-00000001;1", "SIP/AlcazarNetDialer/19549477572,,tTor") in new stack
[Aug 18 14:07:56]   == Using SIP RTP CoS mark 5
[Aug 18 14:07:56]     -- Called SIP/AlcazarNetDialer/19549477572
[Aug 18 14:07:57]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:07:58]        > 0x7efd5400c0b0 -- Strict RTP learning after remote address set to: 162.212.218.145:37926
[Aug 18 14:07:58]     -- SIP/AlcazarNetDialer-0000000a is making progress passing it to Local/8600053@default-00000001;1
[Aug 18 14:08:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:08:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:08:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:08:02]     -- SIP/AlcazarNetDialer-0000000a answered Local/8600053@default-00000001;1
[Aug 18 14:08:02]     -- Channel SIP/AlcazarNetDialer-0000000a joined 'simple_bridge' basic-bridge <815f15e1-b190-471f-b443-bc0f56328803>
[Aug 18 14:08:02]     -- Channel Local/8600053@default-00000001;1 joined 'simple_bridge' basic-bridge <815f15e1-b190-471f-b443-bc0f56328803>
[Aug 18 14:08:02]        > 0x7efd5400c0b0 -- Strict RTP switching to RTP target address 162.212.218.145:37926 as source
[Aug 18 14:08:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:08:03]        > 0x7efd5400c0b0 -- Strict RTP learning complete - Locking on source address 162.212.218.145:37926
[Aug 18 14:08:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:08:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:08:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:08:06]   == Spawn extension (default, 990009*Closers**8155**9549477572*1001**, 1) exited non-zero on 'Local/8600053@default-00000001;2'
[Aug 18 14:08:06]     -- Executing [990009*Closers**8155**9549477572*1001**@default:1] Answer("Local/8600053@default-00000001;2", "") in new stack
[Aug 18 14:08:06]     -- Executing [990009*Closers**8155**9549477572*1001**@default:2] AGI("Local/8600053@default-00000001;2", "agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1") in new stack
[Aug 18 14:08:06]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Aug 18 14:08:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:08:06]     -- Channel SIP/AlcazarNetDialer-0000000a left 'simple_bridge' basic-bridge <815f15e1-b190-471f-b443-bc0f56328803>
[Aug 18 14:08:06]     -- Channel Local/8600053@default-00000001;1 left 'simple_bridge' basic-bridge <815f15e1-b190-471f-b443-bc0f56328803>
[Aug 18 14:08:06]     -- Executing [990009*Closers**8155**9549477572*1001**@default:1] Answer("SIP/AlcazarNetDialer-0000000a", "") in new stack
[Aug 18 14:08:06]     -- Executing [990009*Closers**8155**9549477572*1001**@default:2] AGI("SIP/AlcazarNetDialer-0000000a", "agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1") in new stack
[Aug 18 14:08:06]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Aug 18 14:08:06]   == Spawn extension (default, 519549477572, 2) exited non-zero on 'Local/8600053@default-00000001;1'
[Aug 18 14:08:06]     -- Executing [h@default:1] AGI("Local/8600053@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----10-----SIP 200 OK)") in new stack
[Aug 18 14:08:06]     -- <Local/8600053@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----10-----10-----SIP 200 OK) completed, returning 0
[Aug 18 14:08:06]     -- <Local/8600053@default-00000001;2>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 4
[Aug 18 14:08:06]   == Spawn extension (default, 990009*Closers**8155**9549477572*1001**, 2) exited non-zero on 'Local/8600053@default-00000001;2'
[Aug 18 14:08:06] WARNING[2709][C-00000007]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Aug 18 14:08:06]     -- Executing [h@default:1] AGI("Local/8600053@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Aug 18 14:08:06]     -- <Local/8600053@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Aug 18 14:08:07]     -- <SIP/AlcazarNetDialer-0000000a> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 18 14:08:07]     -- <SIP/AlcazarNetDialer-0000000a> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 18 14:08:07]     -- <SIP/AlcazarNetDialer-0000000a>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Aug 18 14:08:07]     -- Executing [990009*Closers**8155**9549477572*1001**@default:3] Hangup("SIP/AlcazarNetDialer-0000000a", "") in new stack
[Aug 18 14:08:07]   == Spawn extension (default, 990009*Closers**8155**9549477572*1001**, 3) exited non-zero on 'SIP/AlcazarNetDialer-0000000a'
[Aug 18 14:08:07]     -- Executing [h@default:1] AGI("SIP/AlcazarNetDialer-0000000a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK)") in new stack
[Aug 18 14:08:07]     -- <SIP/AlcazarNetDialer-0000000a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed, returning 0
[Aug 18 14:08:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:08:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:08:09]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:08:16] Asterisk cleanly ending (0).
[Aug 18 14:08:16] Executing last minute cleanups


another try with iaxloop

Code: Select all

 Connected to Asterisk 13.34.0-vici currently running on cyburity (pid = 31903)
[Aug 18 14:18:51]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:18:51]     -- Called 90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default
[Aug 18 14:18:51]     -- Executing [90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default:1] Answer("Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2", "") in new stack
[Aug 18 14:18:51]     -- Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;1 answered
[Aug 18 14:18:51]     -- Executing [8600053@default:1] MeetMe("Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;1", "8600053,F") in new stack
[Aug 18 14:18:51]     -- Executing [90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default:2] Dial("Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2", "IAX2/ASTloop:H54XnbVM3ctp14M@127.0.0.1:40569/990009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*,,to") in new stack
[Aug 18 14:18:51]     -- Called IAX2/ASTloop:H54XnbVM3ctp14M@127.0.0.1:40569/990009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*
[Aug 18 14:18:51]     -- Accepting AUTHENTICATED call from 127.0.0.1:37627:
[Aug 18 14:18:51]     --        > requested format = ulaw,
[Aug 18 14:18:51]     --        > requested prefs = (ulaw|gsm),
[Aug 18 14:18:51]     --        > actual format = ulaw,
[Aug 18 14:18:51]     --        > host prefs = (ulaw),
[Aug 18 14:18:51]     --        > priority = mine
[Aug 18 14:18:51]     -- Call accepted by 127.0.0.1:40569 (format ulaw)
[Aug 18 14:18:51]     -- Format for call is (ulaw)
[Aug 18 14:18:51]     -- Executing [990009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default:1] Answer("IAX2/ASTloop-9444", "") in new stack
[Aug 18 14:18:51]     -- IAX2/127.0.0.1:40569-14138 answered Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2
[Aug 18 14:18:51]     -- Channel IAX2/127.0.0.1:40569-14138 joined 'simple_bridge' basic-bridge <6cc37530-f6f1-4004-985c-83c2f9d3eab2>
[Aug 18 14:18:51]     -- Channel Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2 joined 'simple_bridge' basic-bridge <6cc37530-f6f1-4004-985c-83c2f9d3eab2>
[Aug 18 14:18:51]     -- Executing [990009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default:2] AGI("IAX2/ASTloop-9444", "agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1") in new stack
[Aug 18 14:18:51]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Aug 18 14:18:52]     -- <IAX2/ASTloop-9444> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 18 14:18:52]     -- <IAX2/ASTloop-9444> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Aug 18 14:18:52]     -- <IAX2/ASTloop-9444>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Aug 18 14:18:52]     -- Executing [990009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default:3] Hangup("IAX2/ASTloop-9444", "") in new stack
[Aug 18 14:18:52]   == Spawn extension (default, 990009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*, 3) exited non-zero on 'IAX2/ASTloop-9444'
[Aug 18 14:18:52] WARNING[4195][C-0000001f]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Aug 18 14:18:52]     -- Executing [h@default:1] AGI("IAX2/ASTloop-9444", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Aug 18 14:18:52]     -- <IAX2/ASTloop-9444>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Aug 18 14:18:52]     -- Hungup 'IAX2/ASTloop-9444'
[Aug 18 14:18:52]     -- Channel IAX2/127.0.0.1:40569-14138 left 'simple_bridge' basic-bridge <6cc37530-f6f1-4004-985c-83c2f9d3eab2>
[Aug 18 14:18:52]     -- Hungup 'IAX2/127.0.0.1:40569-14138'
[Aug 18 14:18:52]     -- Channel Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2 left 'simple_bridge' basic-bridge <6cc37530-f6f1-4004-985c-83c2f9d3eab2>
[Aug 18 14:18:52]   == Spawn extension (default, 90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*, 2) exited non-zero on 'Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2'
[Aug 18 14:18:52]     -- Executing [h@default:1] AGI("Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0-----IAX2 HANGUP (16))") in new stack
[Aug 18 14:18:52]     -- <Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0-----IAX2 HANGUP (16)) completed, returning 0
[Aug 18 14:18:52]   == Spawn extension (default, 8600053, 1) exited non-zero on 'Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;1'
[Aug 18 14:18:52] WARNING[4193][C-0000001e]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Aug 18 14:18:52]     -- Executing [h@default:1] AGI("Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Aug 18 14:18:52]     -- <Local/90009*AGENTDIRECT*CXFER*8155**9549477572*1001*4001*40*@default-0000000b;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Aug 18 14:18:52]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:18:53]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:18:53] NOTICE[4206]: manager.c:4463 action_hangup: Request to hangup non-existent channel: IAX2/127.0.0.1:40569-14138
[Aug 18 14:18:54]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:19:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:19:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:19:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Aug 18 14:19:02]     -- Recording automatically stopped after a silence of 10 seconds
[Aug 18 14:19:02]     -- <Local/4001@default-0000000a;2> Playing 'auth-thankyou.gsm' (language 'en')
[Aug 18 14:19:03]   == Manager 'sendcron' logged off from 127.0.0.1
[Aug 18 14:19:03]     -- Executing [850266666666664001@default:3] Hangup("Local/4001@default-0000000a;2", "") in new stack
[Aug 18 14:19:03]   == Spawn extension (default, 850266666666664001, 3) exited non-zero on 'Local/4001@default-0000000a;2'
[Aug 18 14:19:03]     -- Executing [h@default:1] AGI("Local/4001@default-0000000a;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY---------------SIP 486 Busy Here)") in new stack
[Aug 18 14:19:03]     -- <Local/4001@default-0000000a;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY---------------SIP 486 Busy Here) completed, returning 0
[Aug 18 14:19:03]   == Spawn extension (default, 8600053, 1) exited non-zero on 'Local/4001@default-0000000a;1'
[Aug 18 14:19:03] WARNING[4148][C-0000001c]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Aug 18 14:19:03]     -- Executing [h@default:1] AGI("Local/4001@default-0000000a;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17--------------------)") in new stack
[Aug 18 14:19:03]     -- <Local/4001@default-0000000a;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17--------------------) completed, returning 0
[Aug 18 14:19:07]   == Manager 'sendcron' logged on from 127.0.0.1

could this be a bug in 13.34.0?

We do have clients using that version and they haven't reported any issues like that. As for me, my most recent version of Asterisk in my testing environment is 13.29.2-vici, so I'm going to try to get a new version installed in my dev environment if I have time so I can test this.

heh, youll never guess what it was. The vTiger integration being turned on was doing it. As soon as I turned it off, it started working.

Thanks for the post-back!

carpenox wrote:heh, youll never guess what it was. The vTiger integration being turned on was doing it. As soon as I turned it off, it started working.

Having the same issue. In my case vtiger was not integrated.

that is what it was in my instance, and if i turn it back on, it happens again. So it def had something to do with whatever vtiger integration does to change the settings. Perhaps you ahve some custom api or custom function doing something similar?

Hello everyone,

I’m revisiting this post because I’m currently facing the same issue.
Internal transfers are not working.
When we press DIAL WITH CUSTOMER after the destination has answered, the call drops, and the agent sees the message XFER LINE HUNG UP.

Configuration details:

• Dedicated OVH Cloud Server
• Vicibox 9.0.3
• Version 2.14.930a
• Build 241021-2138
• Asterisk 13.29.2
• SVN Version: 3882

I also tested with the latest version of Vicibox 11, but the issue persists.

For your information, I have two Vicibox 9.0.3 servers running Asterisk 11, SVN Version 3260 and it works very well. However, Asterisk 13 and above do not work.

I tried installing Asterisk 11 using the vicibox-ast11 script, but it no longer works because the repositories for Asterisk 11 are no longer available.

If anyone can help, I’d appreciate it.
Screenshots are attached.
https://ibb.co/ZLQfdnY


Thank you in advance!

YOu really should update to leap 15.6

Fares MEHENI wrote:I also tested with the latest version of Vicibox 11, but the issue persists.

I have some doubts here.

Did you clean install Vicibox 11.X.X and test without loading your own data set?

Manually add your carrier, perform a single test call with only the necessary users/campaigns/ingroups (whatever) manually added. But NO data importing.

It's common that there is a configuration issue that causes an issue such as this (wrong asterisk version in the Modify Server module, for instance). If you import that data set into the new server (before/without running tests ...) you can bring the problem along for the ride and convince yourself that there is a software issue.

Happens a lot. LOL

Hello,

I tested it on Vicibox 11 with a fresh installation, without any changes or backup restoration. I simply created the minimum setup to test ratio dialing, and I’m still encountering the same issue. After several tests, I’ve confirmed that the problem seems to come either from the Asterisk version or the SVN version. On a Vicibox 9 server with Asterisk 11 and SVN 3260, it works perfectly.

I tried downgrading to Asterisk 11, but I can no longer find the repository.

Best regards,

In that case, you should probably post a Successful and Failed sample of the asterisk -R output so we can compare. It may be a configuration error, but whatever it is may be visible either in the standard asterisk -R output or the debug output.

I think we worked this out already on whatsapp, what was the issue again meheni?