Hello,
I have a problem with outgoing calls VICIDIAL, calls hungs up 2 seconds after the person answers.
I’ve seen lot of people having this issue. HOWEVER, none of Their solutions Worked out for me.
Thank you
The people on these forums aren’t generally familiar with VICIDIAL, so this probably isn’t the best place to come for support. Really, you want to start with the VICIDIAL people. If they can’t solve it, then the people who write it should probably bring it up w/ the Asterisk-focused people.
thank you, it is an asterisk problem, just I mentioned that I use VICIDIAL for more idea . I saw the same type of problems on this forum.
Here are the contents of the log file :
[code][Nov 4 09:22:58] VERBOSE[10092] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:22:58] VERBOSE[10093] pbx.c: – Executing [003397044****@default:1] AGI(“Local/003397044****@default-00000000;2”, “agi://127.0.0.1:4577/call_log”) in new stack
[Nov 4 09:22:58] VERBOSE[10093] res_agi.c: – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Nov 4 09:22:58] VERBOSE[10093] res_agi.c: – <Local/003397044****@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 4 09:22:58] VERBOSE[10093] pbx.c: – Executing [003397044****@default:2] Dial(“Local/003397044****@default-00000000;2”, “SIP/003397211XXXX/003397044****,tTo”) in new stack
[Nov 4 09:22:58] VERBOSE[10093] netsock2.c: == Using SIP RTP CoS mark 5
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Audio is at 17594
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
INVITE sip:003397044****@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK651d6bb0;rport
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Date: Wed, 04 Nov 2015 14:22:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “V104092258000004XXXX” sip:0000000000@192.99.XXX.XXX;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 396446797 396446797 IN IP4 192.99.XXX.XXX
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 192.99.XXX.XXX
t=0 0
m=audio 17594 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[Nov 4 09:22:58] VERBOSE[10093] app_dial.c: – Called SIP/003397211XXXX/003397044****
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 100 Trying
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 102 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK651d6bb0
Content-Length: 0
<------------->
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: — (7 headers 0 lines) —
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 407 authentication required
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:003397044****@10.7.1.60:5060;user=phone
CSeq: 102 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Proxy-Authenticate: Digest realm=“sip.ovh.fr”,nonce=“6979a73f516923e64a083b9a55731158”,opaque=“696cc94274db829”,stale=false,algorithm=MD5
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979adc9-018b59af6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK651d6bb0
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 0
<------------->
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: — (12 headers 0 lines) —
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Transmitting (NAT) to 91.121.129.20:5060:
ACK sip:003397044****@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK651d6bb0;rport
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979adc9-018b59af6
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Content-Length: 0
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Audio is at 17594
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
INVITE sip:003397044****@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK42296f21;rport
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Proxy-Authorization: Digest username=“003397211XXXX”, realm=“sip.ovh.fr”, algorithm=MD5, uri=“sip:003397044****@sip.ovh.fr”, nonce=“6979a73f516923e64a083b9a55731158”, response=“cf84de23fc77767ded3bddd32479d026”, opaque="696cc94274db829"
Date: Wed, 04 Nov 2015 14:22:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “V104092258000004XXXX” sip:0000000000@192.99.XXX.XXX;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 396446797 396446798 IN IP4 192.99.XXX.XXX
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 192.99.XXX.XXX
t=0 0
m=audio 17594 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 100 Trying
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Content-Length: 0
<------------->
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: — (7 headers 0 lines) —
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 180 Ringing
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144664697857 144664697859 IN IP4 10.7.1.149
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 30502 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: — (12 headers 12 lines) —
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found RTP audio format 0
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found RTP audio format 101
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Peer audio RTP is at port 91.121.129.141:30502
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 180 Ringing
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144664697857 144664697859 IN IP4 10.7.1.149
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 30502 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: — (12 headers 12 lines) —
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is ringing
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is making progress passing it to Local/003397044****@default-00000000;2
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is ringing
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is making progress passing it to Local/003397044****@default-00000000;2
[Nov 4 09:23:01] VERBOSE[10129] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:23:01] VERBOSE[10136] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:23:01] VERBOSE[10136] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:02] VERBOSE[10129] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 200 OK
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144664697857 144664697859 IN IP4 10.7.1.149
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 30502 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: — (12 headers 12 lines) —
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: set_destination: Parsing sip:91.121.129.20:5060;transport=udp;lr for address/port to send to
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: set_destination: set destination to 91.121.129.20:5060
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: Transmitting (NAT) to 91.121.129.20:5060:
ACK sip:10.7.1.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK467aa2a0;rport
Route: sip:91.121.129.20:5060;transport=udp;lr
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Content-Length: 0
[Nov 4 09:23:04] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 answered Local/003397044****@default-00000000;2
[Nov 4 09:23:04] VERBOSE[10194] pbx.c: – Executing [8368@default:1] Playback(“Local/003397044****@default-00000000;1”, “sip-silence”) in new stack
[Nov 4 09:23:04] VERBOSE[10194] file.c: – <Local/003397044****@default-00000000;1> Playing ‘sip-silence.gsm’ (language ‘en’)
[Nov 4 09:23:04] VERBOSE[10092] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:04] VERBOSE[10194] pbx.c: – Executing [8368@default:2] AGI(“Local/003397044****@default-00000000;1”, “agi://127.0.0.1:4577/call_log”) in new stack
[Nov 4 09:23:04] VERBOSE[10194] res_agi.c: – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Nov 4 09:23:04] VERBOSE[10194] res_agi.c: – <Local/003397044****@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 4 09:23:04] VERBOSE[10194] pbx.c: – Executing [8368@default:3] AGI(“Local/003397044****@default-00000000;1”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
[Nov 4 09:23:04] VERBOSE[10194] res_agi.c: – Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 4 09:23:05] VERBOSE[10093] pbx.c: – Executing [h@default:1] AGI(“Local/003397044****@default-00000000;2”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16-----ANSWER-----7-----1”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – <SIP/003397211XXXX-00000004>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: – Executing [8368@default:4] AGI(“SIP/003397211XXXX-00000004”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – <SIP/003397211XXXX-00000004>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: – Executing [8368@default:5] Hangup(“SIP/003397211XXXX-00000004”, “”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: == Spawn extension (default, 8368, 5) exited non-zero on ‘SIP/003397211XXXX-00000004’
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: – Executing [h@default:1] AGI(“SIP/003397211XXXX-00000004”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16---------------”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – <SIP/003397211XXXX-00000004>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16--------------- completed, returning 0
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: Scheduling destruction of SIP dialog ‘08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060’ in 32000 ms (Method: INVITE)
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: set_destination: Parsing sip:91.121.129.20:5060;transport=udp;lr for address/port to send to
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: set_destination: set destination to 91.121.129.20:5060
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
BYE sip:10.7.1.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK07c5d11f;rport
Route: sip:91.121.129.20:5060;transport=udp;lr
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Proxy-Authorization: Digest username=“003397211XXXX”, realm=“sip.ovh.fr”, algorithm=MD5, uri=“sip:10.7.1.60:5060”, nonce=“6979a73f516923e64a083b9a55731158”, response=“0700f23634d343b0989ee772b40a9b5a”, opaque="696cc94274db829"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
[Nov 4 09:23:05] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 200 OK
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 104 BYE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK07c5d11f
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 0
<------------->
[Nov 4 09:23:05] VERBOSE[13353] chan_sip.c: — (9 headers 0 lines) —
[Nov 4 09:23:05] VERBOSE[13353] chan_sip.c: Really destroying SIP dialog ‘08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060’ Method: INVITE
[Nov 4 09:23:06] VERBOSE[10093] res_agi.c: – <Local/003397044****@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16-----ANSWER-----7-----1 completed, returning 0
[Nov 4 09:23:06] VERBOSE[10093] pbx.c: == Spawn extension (default, 003397044****, 2) exited non-zero on ‘Local/003397044****@default-00000000;2’
[Nov 4 09:23:06] VERBOSE[10206] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:23:06] VERBOSE[10206] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:11] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:58340 —>[/code]
Thanks
In case it is an Asterisk’s issue, there is not any relevant information, you need to post your cli output, Enable the full log in logger.conf. Do core set verbose 5, core set debug 5, and sip set debug on.
Actually you also need to tell us you are using SIP, and if you are using it for all devices.
this is my cli output :
yes i am using sip , and i test with iax, the same problem !
[Nov 4 09:22:58] VERBOSE[10092] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:22:58] VERBOSE[10093] pbx.c: – Executing [003397044****@default:1] AGI(“Local/003397044****@default-00000000;2”, “agi://127.0.0.1:4577/call_log”) in new stack
[Nov 4 09:22:58] VERBOSE[10093] res_agi.c: – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Nov 4 09:22:58] VERBOSE[10093] res_agi.c: – <Local/003397044****@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 4 09:22:58] VERBOSE[10093] pbx.c: – Executing [003397044****@default:2] Dial(“Local/003397044****@default-00000000;2”, “SIP/003397211XXXX/003397044****,tTo”) in new stack
[Nov 4 09:22:58] VERBOSE[10093] netsock2.c: == Using SIP RTP CoS mark 5
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Audio is at 17594
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 4 09:22:58] VERBOSE[10093] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
INVITE sip:003397044****@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK651d6bb0;rport
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Date: Wed, 04 Nov 2015 14:22:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “V104092258000004XXXX” sip:0000000000@192.99.XXX.XXX;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 396446797 396446797 IN IP4 192.99.XXX.XXX
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 192.99.XXX.XXX
t=0 0
m=audio 17594 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[Nov 4 09:22:58] VERBOSE[10093] app_dial.c: – Called SIP/003397211XXXX/003397044****
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 100 Trying
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 102 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK651d6bb0
Content-Length: 0
<------------->
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: — (7 headers 0 lines) —
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 407 authentication required
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:003397044****@10.7.1.60:5060;user=phone
CSeq: 102 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Proxy-Authenticate: Digest realm=“sip.ovh.fr”,nonce=“6979a73f516923e64a083b9a55731158”,opaque=“696cc94274db829”,stale=false,algorithm=MD5
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979adc9-018b59af6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK651d6bb0
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 0
<------------->
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: — (12 headers 0 lines) —
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Transmitting (NAT) to 91.121.129.20:5060:
ACK sip:003397044****@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK651d6bb0;rport
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979adc9-018b59af6
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Content-Length: 0
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Audio is at 17594
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
INVITE sip:003397044****@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK42296f21;rport
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Proxy-Authorization: Digest username=“003397211XXXX”, realm=“sip.ovh.fr”, algorithm=MD5, uri=“sip:003397044****@sip.ovh.fr”, nonce=“6979a73f516923e64a083b9a55731158”, response=“cf84de23fc77767ded3bddd32479d026”, opaque="696cc94274db829"
Date: Wed, 04 Nov 2015 14:22:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “V104092258000004XXXX” sip:0000000000@192.99.XXX.XXX;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 396446797 396446798 IN IP4 192.99.XXX.XXX
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 192.99.XXX.XXX
t=0 0
m=audio 17594 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 100 Trying
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Content-Length: 0
<------------->
[Nov 4 09:22:58] VERBOSE[13353] chan_sip.c: — (7 headers 0 lines) —
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 180 Ringing
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144664697857 144664697859 IN IP4 10.7.1.149
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 30502 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: — (12 headers 12 lines) —
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found RTP audio format 0
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found RTP audio format 101
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: Peer audio RTP is at port 91.121.129.141:30502
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 180 Ringing
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144664697857 144664697859 IN IP4 10.7.1.149
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 30502 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: — (12 headers 12 lines) —
[Nov 4 09:23:01] VERBOSE[13353] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is ringing
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is making progress passing it to Local/003397044****@default-00000000;2
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is ringing
[Nov 4 09:23:01] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 is making progress passing it to Local/003397044****@default-00000000;2
[Nov 4 09:23:01] VERBOSE[10129] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:23:01] VERBOSE[10136] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:23:01] VERBOSE[10136] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:02] VERBOSE[10129] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 200 OK
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK42296f21
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144664697857 144664697859 IN IP4 10.7.1.149
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 30502 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: — (12 headers 12 lines) —
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: set_destination: Parsing sip:91.121.129.20:5060;transport=udp;lr for address/port to send to
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: set_destination: set destination to 91.121.129.20:5060
[Nov 4 09:23:04] VERBOSE[13353] chan_sip.c: Transmitting (NAT) to 91.121.129.20:5060:
ACK sip:10.7.1.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK467aa2a0;rport
Route: sip:91.121.129.20:5060;transport=udp;lr
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Contact: sip:003397211XXXX@192.99.XXX.XXX:5060
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Content-Length: 0
[Nov 4 09:23:04] VERBOSE[10093] app_dial.c: – SIP/003397211XXXX-00000004 answered Local/003397044****@default-00000000;2
[Nov 4 09:23:04] VERBOSE[10194] pbx.c: – Executing [8368@default:1] Playback(“Local/003397044****@default-00000000;1”, “sip-silence”) in new stack
[Nov 4 09:23:04] VERBOSE[10194] file.c: – <Local/003397044****@default-00000000;1> Playing ‘sip-silence.gsm’ (language ‘en’)
[Nov 4 09:23:04] VERBOSE[10092] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:04] VERBOSE[10194] pbx.c: – Executing [8368@default:2] AGI(“Local/003397044****@default-00000000;1”, “agi://127.0.0.1:4577/call_log”) in new stack
[Nov 4 09:23:04] VERBOSE[10194] res_agi.c: – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Nov 4 09:23:04] VERBOSE[10194] res_agi.c: – <Local/003397044****@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 4 09:23:04] VERBOSE[10194] pbx.c: – Executing [8368@default:3] AGI(“Local/003397044****@default-00000000;1”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
[Nov 4 09:23:04] VERBOSE[10194] res_agi.c: – Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 4 09:23:05] VERBOSE[10093] pbx.c: – Executing [h@default:1] AGI(“Local/003397044****@default-00000000;2”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16-----ANSWER-----7-----1”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – <SIP/003397211XXXX-00000004>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: – Executing [8368@default:4] AGI(“SIP/003397211XXXX-00000004”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – <SIP/003397211XXXX-00000004>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: – Executing [8368@default:5] Hangup(“SIP/003397211XXXX-00000004”, “”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: == Spawn extension (default, 8368, 5) exited non-zero on ‘SIP/003397211XXXX-00000004’
[Nov 4 09:23:05] VERBOSE[10194] pbx.c: – Executing [h@default:1] AGI(“SIP/003397211XXXX-00000004”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16---------------”) in new stack
[Nov 4 09:23:05] VERBOSE[10194] res_agi.c: – <SIP/003397211XXXX-00000004>AGI Script agi://127.0.0.1:4577/call_log–HVcauses … ---------- completed, returning 0
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: Scheduling destruction of SIP dialog '08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060’ in 32000 ms (Method: INVITE)
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: set_destination: Parsing sip:91.121.129.20:5060;transport=udp;lr for address/port to send to
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: set_destination: set destination to 91.121.129.20:5060
[Nov 4 09:23:05] VERBOSE[10194] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
BYE sip:10.7.1.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK07c5d11f;rport
Route: sip:91.121.129.20:5060;transport=udp;lr
Max-Forwards: 70
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Proxy-Authorization: Digest username=“003397211XXXX”, realm=“sip.ovh.fr”, algorithm=MD5, uri=“sip:10.7.1.60:5060”, nonce=“6979a73f516923e64a083b9a55731158”, response=“0700f23634d343b0989ee772b40a9b5a”, opaque="696cc94274db829"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
[Nov 4 09:23:05] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 200 OK
Call-ID: 08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060
CSeq: 104 BYE
From: “V104092258000004XXXX” sip:003397211XXXX@192.99.XXX.XXX;tag=as5f27c7b7
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044****@sip.ovh.fr;tag=00-07854-6979add5-42140b2a6
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK07c5d11f
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 0
<------------->
[Nov 4 09:23:05] VERBOSE[13353] chan_sip.c: — (9 headers 0 lines) —
[Nov 4 09:23:05] VERBOSE[13353] chan_sip.c: Really destroying SIP dialog '08184f77694b55af50f3ff8d02e213c4@192.99.XXX.XXX:5060’ Method: INVITE
[Nov 4 09:23:06] VERBOSE[10093] res_agi.c: – <Local/003397044****@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log–HVcauses … —7-----1 completed, returning 0
[Nov 4 09:23:06] VERBOSE[10093] pbx.c: == Spawn extension (default, 003397044****, 2) exited non-zero on ‘Local/003397044****@default-00000000;2’
[Nov 4 09:23:06] VERBOSE[10206] manager.c: == Manager ‘sendcron’ logged on from 127.0.0.1
[Nov 4 09:23:06] VERBOSE[10206] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 4 09:23:11] VERBOSE[13353] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:58340 —>
thank you
The problem lies in agi-VDAD_ALL_outbound.agi You will need to ask its authors.
this is my cli output after activate debuging, thanks
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Outgoing Call for 003397044XXXX
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Updating call counter for outgoing call
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: ** Our prefcodec: 0x40 (slin)
[Nov 6 11:27:33] VERBOSE[15388] chan_sip.c: Audio is at 16830
[Nov 6 11:27:33] VERBOSE[15388] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Nov 6 11:27:33] VERBOSE[15388] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: – Done with adding codecs to SDP
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Initializing initreq for method INVITE - callid 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 0 [ 43]: INVITE sip:003397044XXXX@sip.ovh.fr SIP/2.0
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 1 [ 63]: Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK3cbd5f80;rport
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 2 [ 16]: Max-Forwards: 70
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 3 [ 76]: From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 4 [ 34]: To: sip:003397044XXXX@sip.ovh.fr
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 5 [ 46]: Contact: sip:0033972XXXXXX@192.99.XXX.XXX:5060
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 6 [ 59]: Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 7 [ 16]: CSeq: 102 INVITE
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 8 [ 75]: User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 9 [ 35]: Date: Fri, 06 Nov 2015 16:27:33 GMT
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 11 [ 26]: Supported: replaces, timer
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 12 [105]: Remote-Party-ID: “V1061127320000047779” sip:0000000000@192.99.XXX.XXX;party=calling;privacy=off;screen=no
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Header 13 [ 29]: Content-Type: application/sdp
[Nov 6 11:27:33] VERBOSE[15388] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
INVITE sip:003397044XXXX@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK3cbd5f80;rport
Max-Forwards: 70
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
To: sip:003397044XXXX@sip.ovh.fr
Contact: sip:0033972XXXXXX@192.99.XXX.XXX:5060
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Date: Fri, 06 Nov 2015 16:27:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “V1061127320000047779” sip:0000000000@192.99.XXX.XXX;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 782817187 782817187 IN IP4 192.99.XXX.XXX
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 192.99.XXX.XXX
t=0 0
m=audio 16830 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #17
[Nov 6 11:27:33] DEBUG[15388] chan_sip.c: Trying to put ‘INVITE sip:’ onto UDP socket destined for 91.121.129.20:5060
[Nov 6 11:27:33] DEBUG[15120] manager.c: Examining event:
Event: Dial
Privilege: call,all
SubEvent: Begin
Channel: Local/003397044XXXX@default-00000000;2
Destination: SIP/0033972XXXXXX-00000000
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
UniqueID: 1446827253.3
DestUniqueID: 1446827253.4
Dialstring: 0033972XXXXXX/003397044XXXX
[Nov 6 11:27:33] VERBOSE[15388] app_dial.c: – Called SIP/0033972XXXXXX/003397044XXXX
[Nov 6 11:27:33] DEBUG[15388] channel.c: Set channel SIP/0033972XXXXXX-00000000 to read format slin
[Nov 6 11:27:33] DEBUG[15388] channel.c: Set channel SIP/0033972XXXXXX-00000000 to write format slin
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 100 Trying
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 102 INVITE
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
To: sip:003397044XXXX@sip.ovh.fr
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK3cbd5f80
Content-Length: 0
<------------->
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 1 [ 59]: Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 2 [ 16]: CSeq: 102 INVITE
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 3 [ 76]: From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 4 [ 34]: To: sip:003397044XXXX@sip.ovh.fr
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 5 [ 90]: Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK3cbd5f80
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 6 [ 17]: Content-Length: 0
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: — (7 headers 0 lines) —
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: = Looking for Call ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060 (Checking To) --From tag as2778b1fb --To-tag
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: *** SIP TIMER: Cancelling retransmission #17 - INVITE (got response)
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ Request 102: Found
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: SIP response 100 to standard invite
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 407 authentication required
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
Contact: sip:003397044XXXX@10.7.1.60:5060;user=phone
CSeq: 102 INVITE
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
Proxy-Authenticate: Digest realm=“sip.ovh.fr”,nonce=“6b5949554e9b261e26ae9a6448c13d0f”,opaque=“6b58d3cc4aa6c67”,stale=false,algorithm=MD5
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044XXXX@sip.ovh.fr;tag=00-08057-6b594a03-757ee1b64
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK3cbd5f80
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 0
<------------->
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 0 [ 35]: SIP/2.0 407 authentication required
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 1 [ 59]: Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 2 [ 54]: Contact: sip:003397044XXXX@10.7.1.60:5060;user=phone
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 3 [ 16]: CSeq: 102 INVITE
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 4 [ 76]: From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 5 [137]: Proxy-Authenticate: Digest realm=“sip.ovh.fr”,nonce=“6b5949554e9b261e26ae9a6448c13d0f”,opaque=“6b58d3cc4aa6c67”,stale=false,algorithm=MD5
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 6 [ 55]: Record-Route: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 7 [ 66]: To: sip:003397044XXXX@sip.ovh.fr;tag=00-08057-6b594a03-757ee1b64
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 8 [ 90]: Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK3cbd5f80
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 9 [ 88]: Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 10 [ 30]: Server: Cirpack/v4.56 (gw_sip)
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 11 [ 17]: Content-Length: 0
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: — (12 headers 0 lines) —
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: = Looking for Call ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060 (Checking To) --From tag as2778b1fb --To-tag 00-08057-6b594a03-757ee1b64
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Acked pending invite 102
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Stopping retransmission on '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ of Request 102: Match Found
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: SIP response 407 to standard invite
[Nov 6 11:27:33] DEBUG[15120] manager.c: Examining event:
Event: SIP-Hangup-Cause
Privilege: system,all
ChannelDriver: SIP
Channel: SIP/0033972XXXXXX-00000000
CallerIDName: V1061127320000047779
Uniqueid: 1446827253.4
Result: 407|authentication required
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: Transmitting (NAT) to 91.121.129.20:5060:
ACK sip:003397044XXXX@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK3cbd5f80;rport
Max-Forwards: 70
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
To: sip:003397044XXXX@sip.ovh.fr;tag=00-08057-6b594a03-757ee1b64
Contact: sip:0033972XXXXXX@192.99.XXX.XXX:5060
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Content-Length: 0
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Trying to put ‘ACK sip:003’ onto UDP socket destined for 91.121.129.20:5060
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Auth attempt 1 on INVITE
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: False
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: ** Our prefcodec: 0x40 (slin)
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: Audio is at 16830
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: – Done with adding codecs to SDP
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
INVITE sip:003397044XXXX@sip.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK1a2cf7bf;rport
Max-Forwards: 70
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
To: sip:003397044XXXX@sip.ovh.fr
Contact: sip:0033972XXXXXX@192.99.XXX.XXX:5060
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Proxy-Authorization: Digest username=“0033972XXXXXX”, realm=“sip.ovh.fr”, algorithm=MD5, uri="sip:003397044XXXX@sip.ovh.fr", nonce=“6b5949554e9b261e26ae9a6448c13d0f”, response=“e7bb6e885c890fe6e41d66f1abbd8511”, opaque="6b58d3cc4aa6c67"
Date: Fri, 06 Nov 2015 16:27:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “V1061127320000047779” sip:0000000000@192.99.XXX.XXX;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 277
v=0
o=root 782817187 782817188 IN IP4 192.99.XXX.XXX
s=Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
c=IN IP4 192.99.XXX.XXX
t=0 0
m=audio 16830 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #19
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Trying to put ‘INVITE sip:’ onto UDP socket destined for 91.121.129.20:5060
[Nov 6 11:27:33] DEBUG[15120] manager.c: Running action ‘Command’
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 100 Trying
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 103 INVITE
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
To: sip:003397044XXXX@sip.ovh.fr
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK1a2cf7bf
Content-Length: 0
<------------->
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 0 [ 18]: SIP/2.0 100 Trying
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 1 [ 59]: Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 2 [ 16]: CSeq: 103 INVITE
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 3 [ 76]: From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 4 [ 34]: To: sip:003397044XXXX@sip.ovh.fr
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 5 [ 90]: Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK1a2cf7bf
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 6 [ 17]: Content-Length: 0
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c: — (7 headers 0 lines) —
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: = Looking for Call ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060 (Checking To) --From tag as2778b1fb --To-tag
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: *** SIP TIMER: Cancelling retransmission #19 - INVITE (got response)
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ Request 103: Found
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: SIP response 100 to standard invite
[Nov 6 11:27:33] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:33] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:33] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:XXX.XXX.XXX.XXX:64638 —>
<------------->
[Nov 6 11:27:33] DEBUG[14933] chan_sip.c: Header 0 [ 0]:
[Nov 6 11:27:34] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:34] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:35] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 180 Ringing
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK1a2cf7bf
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144682725397 144682725398 IN IP4 10.7.1.154
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 33678 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 0 [ 19]: SIP/2.0 180 Ringing
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 1 [ 59]: Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 2 [ 29]: Contact: sip:10.7.1.60:5060
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 3 [ 29]: Content-Type: application/sdp
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 5 [ 76]: From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 6 [ 55]: Record-Route: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 7 [ 66]: To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 8 [ 90]: Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK1a2cf7bf
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 9 [ 88]: Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 10 [ 30]: Server: Cirpack/v4.56 (gw_sip)
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 11 [ 19]: Content-Length: 239
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Header 12 [ 0]:
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 0 [ 3]: v=0
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 1 [ 50]: o=cp10 144682725397 144682725398 IN IP4 10.7.1.154
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 3 [ 23]: c=IN IP4 91.121.129.141
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 5 [ 27]: m=audio 33678 RTP/AVP 0 101
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 6 [ 7]: b=AS:82
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 7 [ 22]: a=rtpmap:0 PCMU/8000/1
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 10 [ 10]: a=ptime:20
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Body 11 [ 10]: a=sendrecv
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: — (12 headers 12 lines) —
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: = Looking for Call ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060 (Checking To) --From tag as2778b1fb --To-tag 00-08092-6b594a0f-39fedb527
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ Request 103: Found
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: SIP response 180 to standard invite
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: build_route: Record-Route hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Nov 6 11:27:35] DEBUG[15120] manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: SIP/0033972XXXXXX-00000000
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Uniqueid: 1446827253.4
[Nov 6 11:27:35] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for SIP - 0033972XXXXXX
[Nov 6 11:27:35] DEBUG[14925] chan_sip.c: Checking device state for peer 0033972XXXXXX
[Nov 6 11:27:35] DEBUG[14925] devicestate.c: Changing state for SIP/0033972XXXXXX - state 1 (Not in use)
[Nov 6 11:27:35] DEBUG[14925] devicestate.c: device ‘SIP/0033972XXXXXX’ state ‘1’
[Nov 6 11:27:35] DEBUG[15000] app_queue.c: Device ‘SIP/0033972XXXXXX’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing session-level SDP o=cp10 144682725397 144682725398 IN IP4 10.7.1.154… OK.
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing session-level SDP s=SIP Call… UNSUPPORTED OR FAILED.
[Nov 6 11:27:35] DEBUG[14933] netsock2.c: Splitting ‘91.121.129.141’ into…
[Nov 6 11:27:35] DEBUG[14933] netsock2.c: …host ‘91.121.129.141’ and port ‘’.
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing session-level SDP c=IN IP4 91.121.129.141… OK.
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing session-level SDP t=0 0… UNSUPPORTED OR FAILED.
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: Found RTP audio format 0
[Nov 6 11:27:35] DEBUG[14933] rtp_engine.c: Setting payload 0 based on m type on 0x2b0b91319da0
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: Found RTP audio format 101
[Nov 6 11:27:35] DEBUG[14933] rtp_engine.c: Setting payload 101 based on m type on 0x2b0b91319da0
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing media-level (audio) SDP b=AS:82… UNSUPPORTED OR FAILED.
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: Found audio description format PCMU for ID 0
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000/1… OK.
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: Found audio description format telephone-event for ID 101
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15… UNSUPPORTED OR FAILED.
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing media-level (audio) SDP a=ptime:20… OK.
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[Nov 6 11:27:35] DEBUG[14933] rtp_engine.c: Incorporating payload 0 on 0x2b0b91319da0
[Nov 6 11:27:35] DEBUG[14933] rtp_engine.c: Incorporating payload 101 on 0x2b0b91319da0
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 6 11:27:35] DEBUG[14933] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0x1d52f7d8’
[Nov 6 11:27:35] VERBOSE[14933] chan_sip.c: Peer audio RTP is at port 91.121.129.141:33678
[Nov 6 11:27:35] DEBUG[14933] rtp_engine.c: Copying payload 0 from 0x2b0b91319da0 to 0x1d52f9a0
[Nov 6 11:27:35] DEBUG[14933] rtp_engine.c: Copying payload 101 from 0x2b0b91319da0 to 0x1d52f9a0
[Nov 6 11:27:35] DEBUG[14933] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance ‘0x1d52f7d8’
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: We’re settling with these formats: 0x4 (ulaw)
[Nov 6 11:27:35] DEBUG[14933] chan_sip.c: We have an owner, now see if we need to change this call
[Nov 6 11:27:35] VERBOSE[15388] app_dial.c: – SIP/0033972XXXXXX-00000000 is ringing
[Nov 6 11:27:35] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for Local - 003397044XXXX@default
[Nov 6 11:27:35] DEBUG[14925] chan_local.c: Checking if extension 003397044XXXX@default exists (devicestate)
[Nov 6 11:27:35] DEBUG[15120] manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: Local/003397044XXXX@default-00000000;1
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Uniqueid: 1446827253.2
[Nov 6 11:27:35] DEBUG[14925] devicestate.c: Changing state for Local/003397044XXXX@default - state 2 (In use)
[Nov 6 11:27:35] DEBUG[14925] devicestate.c: device ‘Local/003397044XXXX@default’ state ‘2’
[Nov 6 11:27:35] VERBOSE[15388] app_dial.c: – SIP/0033972XXXXXX-00000000 is making progress passing it to Local/003397044XXXX@default-00000000;2
[Nov 6 11:27:35] DEBUG[15000] app_queue.c: Device ‘Local/003397044XXXX@default’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:35] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:35] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:36] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:36] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:37] DEBUG[14938] chan_iax2.c: ip callno count decremented to 9 for 127.0.0.1
[Nov 6 11:27:37] DEBUG[14938] chan_iax2.c: ip callno count decremented to 8 for 127.0.0.1
[Nov 6 11:27:37] DEBUG[14938] chan_iax2.c: ip callno count decremented to 7 for 127.0.0.1
[Nov 6 11:27:37] DEBUG[14938] chan_iax2.c: ip callno count decremented to 6 for 127.0.0.1
[Nov 6 11:27:37] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:37] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:38] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:38] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:38] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 200 OK
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
Contact: sip:10.7.1.60:5060
Content-Type: application/sdp
CSeq: 103 INVITE
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK1a2cf7bf
Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 239
v=0
o=cp10 144682725397 144682725398 IN IP4 10.7.1.154
s=SIP Call
c=IN IP4 91.121.129.141
t=0 0
m=audio 33678 RTP/AVP 0 101
b=AS:82
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 1 [ 59]: Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 2 [ 29]: Contact: sip:10.7.1.60:5060
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 3 [ 29]: Content-Type: application/sdp
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 4 [ 16]: CSeq: 103 INVITE
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 5 [ 76]: From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 6 [ 55]: Record-Route: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 7 [ 66]: To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 8 [ 90]: Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK1a2cf7bf
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 9 [ 88]: Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 10 [ 30]: Server: Cirpack/v4.56 (gw_sip)
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 11 [ 19]: Content-Length: 239
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Header 12 [ 0]:
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 0 [ 3]: v=0
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 1 [ 50]: o=cp10 144682725397 144682725398 IN IP4 10.7.1.154
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 2 [ 10]: s=SIP Call
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 3 [ 23]: c=IN IP4 91.121.129.141
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 4 [ 5]: t=0 0
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 5 [ 27]: m=audio 33678 RTP/AVP 0 101
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 6 [ 7]: b=AS:82
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 7 [ 22]: a=rtpmap:0 PCMU/8000/1
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 8 [ 33]: a=rtpmap:101 telephone-event/8000
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 9 [ 15]: a=fmtp:101 0-15
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 10 [ 10]: a=ptime:20
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Body 11 [ 10]: a=sendrecv
[Nov 6 11:27:38] VERBOSE[14933] chan_sip.c: — (12 headers 12 lines) —
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: = Looking for Call ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060 (Checking To) --From tag as2778b1fb --To-tag 00-08092-6b594a0f-39fedb527
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Acked pending invite 103
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Stopping retransmission on '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ of Request 103: Match Found
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: SIP response 200 to standard invite
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED OR FAILED.
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Call 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060 responded to our reinvite without changing SDP version; ignoring SDP.
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Updating call counter for outgoing call
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: build_route: Record-Route hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:38] VERBOSE[14933] chan_sip.c: list_route: hop: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:38] VERBOSE[14933] chan_sip.c: set_destination: Parsing sip:91.121.129.20:5060;transport=udp;lr for address/port to send to
[Nov 6 11:27:38] DEBUG[14933] netsock2.c: Splitting ‘91.121.129.20:5060’ into…
[Nov 6 11:27:38] DEBUG[15120] manager.c: Examining event:
Event: ChannelUpdate
Privilege: system,all
Channel: SIP/0033972XXXXXX-00000000
Channeltype: SIP
Uniqueid: 1446827253.4
SIPcallid: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
SIPfullcontact: sip:10.7.1.60:5060
Peername: 0033972XXXXXX
[Nov 6 11:27:38] DEBUG[14933] netsock2.c: …host ‘91.121.129.20’ and port ‘5060’.
[Nov 6 11:27:38] VERBOSE[14933] chan_sip.c: set_destination: set destination to 91.121.129.20:5060
[Nov 6 11:27:38] VERBOSE[14933] chan_sip.c: Transmitting (NAT) to 91.121.129.20:5060:
ACK sip:10.7.1.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK44e35026;rport
Route: sip:91.121.129.20:5060;transport=udp;lr
Max-Forwards: 70
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
Contact: sip:0033972XXXXXX@192.99.XXX.XXX:5060
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Content-Length: 0
[Nov 6 11:27:38] DEBUG[14933] chan_sip.c: Trying to put ‘ACK sip:10.’ onto UDP socket destined for 91.121.129.20:5060
[Nov 6 11:27:38] VERBOSE[15388] app_dial.c: – SIP/0033972XXXXXX-00000000 answered Local/003397044XXXX@default-00000000;2
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for SIP - 0033972XXXXXX
[Nov 6 11:27:38] DEBUG[14925] chan_sip.c: Checking device state for peer 0033972XXXXXX
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: Changing state for SIP/0033972XXXXXX - state 1 (Not in use)
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: device ‘SIP/0033972XXXXXX’ state ‘1’
[Nov 6 11:27:38] DEBUG[15388] chan_local.c: Blocked indication -1
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for Local - 003397044XXXX@default
[Nov 6 11:27:38] DEBUG[14925] chan_local.c: Checking if extension 003397044XXXX@default exists (devicestate)
[Nov 6 11:27:38] DEBUG[15388] features.c: bridge answer set, chan answer set
[Nov 6 11:27:38] DEBUG[15388] features.c: Removing dialed interfaces datastore on SIP/0033972XXXXXX-00000000 since we’re bridging
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: Changing state for Local/003397044XXXX@default - state 2 (In use)
[Nov 6 11:27:38] DEBUG[15388] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: device ‘Local/003397044XXXX@default’ state ‘2’
[Nov 6 11:27:38] DEBUG[15120] manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: SIP/0033972XXXXXX-00000000
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Uniqueid: 1446827253.4
[Nov 6 11:27:38] DEBUG[15120] manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: Local/003397044XXXX@default-00000000;2
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Uniqueid: 1446827253.3
[Nov 6 11:27:38] DEBUG[15120] manager.c: Examining event:
Event: NewAccountCode
Privilege: call,all
Channel: SIP/0033972XXXXXX-00000000
Uniqueid: 1446827253.4
AccountCode:
OldAccountCode:
[Nov 6 11:27:38] DEBUG[15120] manager.c: Examining event:
Event: Bridge
Privilege: call,all
Bridgestate: Link
Bridgetype: core
Channel1: Local/003397044XXXX@default-00000000;2
Channel2: SIP/0033972XXXXXX-00000000
Uniqueid1: 1446827253.3
Uniqueid2: 1446827253.4
CallerID1: 0000000000
CallerID2: 0000000000
[Nov 6 11:27:38] VERBOSE[15387] pbx.c: > Channel Local/003397044XXXX@default-00000000;1 was answered.
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for Local - 003397044XXXX@default
[Nov 6 11:27:38] DEBUG[14925] chan_local.c: Checking if extension 003397044XXXX@default exists (devicestate)
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: Changing state for Local/003397044XXXX@default - state 2 (In use)
[Nov 6 11:27:38] DEBUG[14925] devicestate.c: device ‘Local/003397044XXXX@default’ state ‘2’
[Nov 6 11:27:38] DEBUG[15120] manager.c: Examining event:
Event: Newstate
Privilege: call,all
Channel: Local/003397044XXXX@default-00000000;1
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Uniqueid: 1446827253.2
[Nov 6 11:27:38] DEBUG[15000] app_queue.c: Device ‘SIP/0033972XXXXXX’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:38] DEBUG[15000] app_queue.c: Device ‘Local/003397044XXXX@default’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:38] DEBUG[15000] app_queue.c: Device ‘Local/003397044XXXX@default’ changed to state ‘2’ (In use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:38] DEBUG[15404] pbx.c: Launching ‘Playback’
[Nov 6 11:27:38] VERBOSE[15404] pbx.c: – Executing [8368@default:1] Playback(“Local/003397044XXXX@default-00000000;1”, “sip-silence”) in new stack
[Nov 6 11:27:38] DEBUG[15404] channel.c: Set channel Local/003397044XXXX@default-00000000;1 to write format gsm
[Nov 6 11:27:38] DEBUG[15388] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[Nov 6 11:27:38] DEBUG[15388] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[Nov 6 11:27:38] DEBUG[15388] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance ‘0x1d52f7d8’
[Nov 6 11:27:38] DEBUG[15404] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second
[Nov 6 11:27:38] VERBOSE[15404] file.c: – <Local/003397044XXXX@default-00000000;1> Playing ‘sip-silence.gsm’ (language ‘en’)
[Nov 6 11:27:38] DEBUG[15387] manager.c: Running action ‘Logoff’
[Nov 6 11:27:38] VERBOSE[15387] manager.c: == Manager ‘sendcron’ logged off from 127.0.0.1
[Nov 6 11:27:38] DEBUG[15404] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Nov 6 11:27:38] DEBUG[15404] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Nov 6 11:27:38] DEBUG[15404] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second
[Nov 6 11:27:38] DEBUG[15404] channel.c: Set channel Local/003397044XXXX@default-00000000;1 to write format slin
[Nov 6 11:27:38] DEBUG[15404] pbx.c: Launching ‘AGI’
[Nov 6 11:27:38] VERBOSE[15404] pbx.c: – Executing [8368@default:2] AGI(“Local/003397044XXXX@default-00000000;1”, “agi://127.0.0.1:4577/call_log”) in new stack
[Nov 6 11:27:38] DEBUG[15404] res_agi.c: Wow, connected!
[Nov 6 11:27:38] VERBOSE[15404] res_agi.c: – AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TESTCAMP))
[Nov 6 11:27:38] VERBOSE[15404] res_agi.c: – <Local/003397044XXXX@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 6 11:27:38] DEBUG[15404] pbx.c: Launching ‘AGI’
[Nov 6 11:27:38] VERBOSE[15404] pbx.c: – Executing [8368@default:3] AGI(“Local/003397044XXXX@default-00000000;1”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
[Nov 6 11:27:38] VERBOSE[15404] res_agi.c: – Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 6 11:27:39] DEBUG[15388] channel.c: Planning to masquerade channel SIP/0033972XXXXXX-00000000 into the structure of Local/003397044XXXX@default-00000000;1
[Nov 6 11:27:39] DEBUG[15388] channel.c: Done planning to masquerade channel SIP/0033972XXXXXX-00000000 into the structure of Local/003397044XXXX@default-00000000;1
[Nov 6 11:27:39] DEBUG[15388] chan_local.c: Masquerading Local/003397044XXXX@default-00000000;1 <- SIP/0033972XXXXXX-00000000
[Nov 6 11:27:39] DEBUG[15404] chan_local.c: Blocked indication -1
[Nov 6 11:27:39] DEBUG[15404] channel.c: Actually Masquerading SIP/0033972XXXXXX-00000000(6) into the structure of Local/003397044XXXX@default-00000000;1(6)
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: Masquerade
Privilege: call,all
Clone: SIP/0033972XXXXXX-00000000
CloneState: Up
Original: Local/003397044XXXX@default-00000000;1
OriginalState: Up
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: Rename
Privilege: call,all
Channel: SIP/0033972XXXXXX-00000000
Newname: SIP/0033972XXXXXX-00000000
Uniqueid: 1446827253.4
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: Rename
Privilege: call,all
Channel: Local/003397044XXXX@default-00000000;1
Newname: SIP/0033972XXXXXX-00000000
Uniqueid: 1446827253.2
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: Rename
Privilege: call,all
Channel: SIP/0033972XXXXXX-00000000
Newname: Local/003397044XXXX@default-00000000;1
Uniqueid: 1446827253.4
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: NewCallerid
Privilege: call,all
Channel: SIP/0033972XXXXXX-00000000
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
Uniqueid: 1446827253.2
CID-CallingPres: 0 (Presentation Allowed, Not Screened)
[Nov 6 11:27:39] DEBUG[15404] channel.c: Set channel SIP/0033972XXXXXX-00000000 to write format slin
[Nov 6 11:27:39] DEBUG[15404] channel.c: Set channel SIP/0033972XXXXXX-00000000 to read format slin
[Nov 6 11:27:39] DEBUG[15404] channel.c: Putting channel SIP/0033972XXXXXX-00000000 in slin/slin formats
[Nov 6 11:27:39] DEBUG[15404] chan_sip.c: SIP Fixup: New owner for dialogue 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060: SIP/0033972XXXXXX-00000000 (Old parent: Local/003397044XXXX@default-00000000;1)
[Nov 6 11:27:39] DEBUG[15404] channel.c: Done Masquerading SIP/0033972XXXXXX-00000000 (6)
[Nov 6 11:27:39] DEBUG[15404] res_rtp_asterisk.c: Changing ssrc from 609914259 to 321984035 due to a source change
[Nov 6 11:27:39] DEBUG[15388] chan_local.c: Not posting to ‘Local/003397044XXXX@default-00000000;2’ queue since already masqueraded out
[Nov 6 11:27:39] DEBUG[15388] channel.c: Bridge stops because we’re zombie or need a soft hangup: c0=Local/003397044XXXX@default-00000000;2, c1=Local/003397044XXXX@default-00000000;1, flags: No,Yes,Yes,Yes
[Nov 6 11:27:39] DEBUG[15388] channel.c: Bridge stops bridging channels Local/003397044XXXX@default-00000000;2 and Local/003397044XXXX@default-00000000;1
[Nov 6 11:27:39] DEBUG[15388] channel.c: Soft-Hanging up channel ‘Local/003397044XXXX@default-00000000;2’
[Nov 6 11:27:39] DEBUG[15388] pbx.c: Result of ‘HANGUPCAUSE’ is ‘16’
[Nov 6 11:27:39] DEBUG[15388] pbx.c: Result of ‘DIALSTATUS’ is ‘ANSWER’
[Nov 6 11:27:39] DEBUG[15388] pbx.c: Result of ‘DIALEDTIME’ is ‘6’
[Nov 6 11:27:39] DEBUG[15388] pbx.c: Result of ‘ANSWEREDTIME’ is ‘1’
[Nov 6 11:27:39] DEBUG[15388] pbx.c: Launching ‘AGI’
[Nov 6 11:27:39] VERBOSE[15388] pbx.c: – Executing [h@default:1] AGI(“Local/003397044XXXX@default-00000000;2”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16-----ANSWER-----6-----1”) in new stack
[Nov 6 11:27:39] DEBUG[15388] res_agi.c: Hungup channel detected, running agi in dead mode.
[Nov 6 11:27:39] DEBUG[15388] res_agi.c: Wow, connected!
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: Unlink
Privilege: call,all
Channel1: Local/003397044XXXX@default-00000000;2
Channel2: Local/003397044XXXX@default-00000000;1
Uniqueid1: 1446827253.3
Uniqueid2: 1446827253.4
CallerID1: 0000000000
CallerID2: 0000000000
[Nov 6 11:27:39] VERBOSE[15404] res_agi.c: – <SIP/0033972XXXXXX-00000000>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Launching ‘AGI’
[Nov 6 11:27:39] VERBOSE[15404] pbx.c: – Executing [8368@default:4] AGI(“SIP/0033972XXXXXX-00000000”, “agi-VDAD_ALL_outbound.agi,NORMAL-----LB”) in new stack
[Nov 6 11:27:39] VERBOSE[15404] res_agi.c: – Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Nov 6 11:27:39] DEBUG[14941] chan_iax2.c: JB STATS:IAX2/4002-8270 ping=144 ljitterms=59 ljbdelayms=120 ltotlost=5 lrecentlosspct=0 ldropped=3 looo=0 lrecvd=3085 rjitterms=9 rjbdelayms=82 rtotlost=27 rrecentlosspct=0 rdropped=3 rooo=0 rrecvd=3052
[Nov 6 11:27:39] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:39] VERBOSE[15404] res_agi.c: – <SIP/0033972XXXXXX-00000000>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Launching ‘Hangup’
[Nov 6 11:27:39] VERBOSE[15404] pbx.c: – Executing [8368@default:5] Hangup(“SIP/0033972XXXXXX-00000000”, “”) in new stack
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Spawn extension (default,8368,5) exited non-zero on ‘SIP/0033972XXXXXX-00000000’
[Nov 6 11:27:39] VERBOSE[15404] pbx.c: == Spawn extension (default, 8368, 5) exited non-zero on ‘SIP/0033972XXXXXX-00000000’
[Nov 6 11:27:39] DEBUG[15404] channel.c: Soft-Hanging up channel ‘SIP/0033972XXXXXX-00000000’
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Result of ‘HANGUPCAUSE’ is ‘16’
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Result of ‘DIALSTATUS’ is NULL
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Result of ‘DIALEDTIME’ is NULL
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Result of ‘ANSWEREDTIME’ is NULL
[Nov 6 11:27:39] DEBUG[15404] pbx.c: Launching ‘AGI’
[Nov 6 11:27:39] VERBOSE[15404] pbx.c: – Executing [h@default:1] AGI(“SIP/0033972XXXXXX-00000000”, “agi://127.0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----16---------------”) in new stack
[Nov 6 11:27:39] DEBUG[15404] res_agi.c: Hungup channel detected, running agi in dead mode.
[Nov 6 11:27:39] DEBUG[15404] res_agi.c: Wow, connected!
[Nov 6 11:27:39] VERBOSE[15404] res_agi.c: – <SIP/0033972XXXXXX-00000000>AGI Script agi://127.0.0.1:4577/call_log–HVcauses … ---------- completed, returning 0
[Nov 6 11:27:39] DEBUG[15404] channel.c: Hanging up channel ‘SIP/0033972XXXXXX-00000000’
[Nov 6 11:27:39] DEBUG[15404] chan_sip.c: Hangup call SIP/0033972XXXXXX-00000000, SIP callid 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:39] DEBUG[15404] res_rtp_asterisk.c: Setting RTCP address on RTP instance ‘0x1d52f7d8’
[Nov 6 11:27:39] VERBOSE[15404] chan_sip.c: Scheduling destruction of SIP dialog '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ in 32000 ms (Method: INVITE)
[Nov 6 11:27:39] VERBOSE[15404] chan_sip.c: set_destination: Parsing sip:91.121.129.20:5060;transport=udp;lr for address/port to send to
[Nov 6 11:27:39] DEBUG[15404] netsock2.c: Splitting ‘91.121.129.20:5060’ into…
[Nov 6 11:27:39] DEBUG[15404] netsock2.c: …host ‘91.121.129.20’ and port ‘5060’.
[Nov 6 11:27:39] VERBOSE[15404] chan_sip.c: set_destination: set destination to 91.121.129.20:5060
[Nov 6 11:27:39] VERBOSE[15404] chan_sip.c: Reliably Transmitting (NAT) to 91.121.129.20:5060:
BYE sip:10.7.1.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;branch=z9hG4bK2f443bce;rport
Route: sip:91.121.129.20:5060;transport=udp;lr
Max-Forwards: 70
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Proxy-Authorization: Digest username=“0033972XXXXXX”, realm=“sip.ovh.fr”, algorithm=MD5, uri=“sip:10.7.1.60:5060”, nonce=“6b5949554e9b261e26ae9a6448c13d0f”, response=“be1f7239d93e76b7a81f0ee30fd93044”, opaque="6b58d3cc4aa6c67"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
[Nov 6 11:27:39] DEBUG[15404] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id #22
[Nov 6 11:27:39] DEBUG[15404] chan_sip.c: Trying to put ‘BYE sip:10.’ onto UDP socket destined for 91.121.129.20:5060
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: SIP/0033972XXXXXX-00000000
Uniqueid: 1446827253.2
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Cause: 16
Cause-txt: Normal Clearing
[Nov 6 11:27:39] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for SIP - 0033972XXXXXX
[Nov 6 11:27:39] DEBUG[14925] chan_sip.c: Checking device state for peer 0033972XXXXXX
[Nov 6 11:27:39] DEBUG[14925] devicestate.c: Changing state for SIP/0033972XXXXXX - state 1 (Not in use)
[Nov 6 11:27:39] DEBUG[14925] devicestate.c: device ‘SIP/0033972XXXXXX’ state ‘1’
[Nov 6 11:27:39] DEBUG[15000] app_queue.c: Device ‘SIP/0033972XXXXXX’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:39] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
SIP/2.0 200 OK
Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
CSeq: 104 BYE
From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
Record-Route: sip:91.121.129.20:5060;transport=udp;lr
To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK2f443bce
Server: Cirpack/v4.56 (gw_sip)
Content-Length: 0
<------------->
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 1 [ 59]: Call-ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 2 [ 13]: CSeq: 104 BYE
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 3 [ 76]: From: “V1061127320000047779” sip:0033972XXXXXX@192.99.XXX.XXX;tag=as2778b1fb
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 4 [ 55]: Record-Route: sip:91.121.129.20:5060;transport=udp;lr
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 5 [ 66]: To: sip:003397044XXXX@sip.ovh.fr;tag=00-08092-6b594a0f-39fedb527
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 6 [ 90]: Via: SIP/2.0/UDP 192.99.XXX.XXX:5060;received=192.99.XXX.XXX;rport=5060;branch=z9hG4bK2f443bce
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 7 [ 30]: Server: Cirpack/v4.56 (gw_sip)
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Header 8 [ 17]: Content-Length: 0
[Nov 6 11:27:39] VERBOSE[14933] chan_sip.c: — (9 headers 0 lines) —
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: = Looking for Call ID: 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060 (Checking To) --From tag as2778b1fb --To-tag 00-08092-6b594a0f-39fedb527
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #22
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Stopping retransmission on '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ of Request 104: Match Found
[Nov 6 11:27:39] DEBUG[14933] chan_sip.c: Destroying SIP dialog 7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060
[Nov 6 11:27:39] VERBOSE[14933] chan_sip.c: Really destroying SIP dialog '7a5722ef4f5a3e86317a356e3436d9bf@192.99.XXX.XXX:5060’ Method: INVITE
[Nov 6 11:27:39] DEBUG[14933] rtp_engine.c: Destroyed RTP instance ‘0x1d52f7d8’
[Nov 6 11:27:39] DEBUG[15120] manager.c: Examining event:
Event: SIP-Response
Privilege: system,all
ChannelDriver: SIP
Method: BYE
Result: 200|OK
[Nov 6 11:27:39] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:40] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:40] VERBOSE[15388] res_agi.c: – <Local/003397044XXXX@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log–HVcauses … —6-----1 completed, returning 0
[Nov 6 11:27:40] DEBUG[15388] channel.c: Hanging up zombie ‘Local/003397044XXXX@default-00000000;1’
[Nov 6 11:27:40] DEBUG[15120] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: Local/003397044XXXX@default-00000000;1
Uniqueid: 1446827253.4
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Cause: 16
Cause-txt: Normal Clearing
[Nov 6 11:27:40] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for Local - 003397044XXXX@default
[Nov 6 11:27:40] DEBUG[14925] chan_local.c: Checking if extension 003397044XXXX@default exists (devicestate)
[Nov 6 11:27:40] DEBUG[15120] manager.c: Examining event:
Event: Dial
Privilege: call,all
SubEvent: End
Channel: Local/003397044XXXX@default-00000000;2
UniqueID: 1446827253.3
DialStatus: ANSWER
[Nov 6 11:27:40] DEBUG[14925] devicestate.c: Changing state for Local/003397044XXXX@default - state 1 (Not in use)
[Nov 6 11:27:40] DEBUG[14925] devicestate.c: device ‘Local/003397044XXXX@default’ state ‘1’
[Nov 6 11:27:40] DEBUG[15388] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Nov 6 11:27:40] DEBUG[15000] app_queue.c: Device ‘Local/003397044XXXX@default’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:40] DEBUG[15388] pbx.c: Spawn extension (default,003397044XXXX,2) exited non-zero on ‘Local/003397044XXXX@default-00000000;2’
[Nov 6 11:27:40] VERBOSE[15388] pbx.c: == Spawn extension (default, 003397044XXXX, 2) exited non-zero on ‘Local/003397044XXXX@default-00000000;2’
[Nov 6 11:27:40] DEBUG[15388] channel.c: Soft-Hanging up channel ‘Local/003397044XXXX@default-00000000;2’
[Nov 6 11:27:40] DEBUG[15388] channel.c: Hanging up channel ‘Local/003397044XXXX@default-00000000;2’
[Nov 6 11:27:40] DEBUG[15120] manager.c: Examining event:
Event: Hangup
Privilege: call,all
Channel: Local/003397044XXXX@default-00000000;2
Uniqueid: 1446827253.3
CallerIDNum: 0000000000
CallerIDName: V1061127320000047779
ConnectedLineNum: 0000000000
ConnectedLineName: V1061127320000047779
Cause: 16
Cause-txt: Normal Clearing
[Nov 6 11:27:40] DEBUG[14925] devicestate.c: No provider found, checking channel drivers for Local - 003397044XXXX@default
[Nov 6 11:27:40] DEBUG[14925] chan_local.c: Checking if extension 003397044XXXX@default exists (devicestate)
[Nov 6 11:27:40] DEBUG[14925] devicestate.c: Changing state for Local/003397044XXXX@default - state 1 (Not in use)
[Nov 6 11:27:40] DEBUG[14925] devicestate.c: device ‘Local/003397044XXXX@default’ state ‘1’
[Nov 6 11:27:40] DEBUG[15000] app_queue.c: Device ‘Local/003397044XXXX@default’ changed to state ‘1’ (Not in use) but we don’t care because they’re not a member of any queue.
[Nov 6 11:27:40] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:40] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:41] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:41] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:42] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:42] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:43] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:43] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:44] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:44] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:44] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:45] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:45] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:46] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:46] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:47] DEBUG[15110] manager.c: Running action ‘Command’
[Nov 6 11:27:47] DEBUG[14938] chan_iax2.c: ip callno count decremented to 5 for 127.0.0.1
[Nov 6 11:27:47] DEBUG[14938] chan_iax2.c: ip callno count decremented to 4 for 127.0.0.1
[Nov 6 11:27:47] DEBUG[14970] chan_iax2.c: ip callno count incremented to 5 for 127.0.0.1
[Nov 6 11:27:47] DEBUG[14969] chan_iax2.c: ip callno count incremented to 6 for 127.0.0.1
[Nov 6 11:27:47] DEBUG[14973] chan_iax2.c: schedule decrement of callno used for 127.0.0.1 in 60 seconds
[Nov 6 11:27:47] DEBUG[14973] chan_iax2.c: Peer ASTblind: got pong, lastms 3, historicms 3, maxms 2000
[Nov 6 11:27:47] DEBUG[14974] chan_iax2.c: schedule decrement of callno used for 127.0.0.1 in 60 seconds
[Nov 6 11:27:47] DEBUG[14974] chan_iax2.c: Peer ASTloop: got pong, lastms 3, historicms 3, maxms 2000
[Nov 6 11:27:47] VERBOSE[14933] chan_sip.c:
<— SIP read from UDP:91.121.129.20:5060 —>
OPTIONS sip:0033972XXXXXX@192.99.XXX.XXX:5060 SIP/2.0
Call-ID: 00-07236-06cfefe6-6a375db91@91.121.129.20
Contact: sip:91.121.129.20:5060
CSeq: 1 OPTIONS
From: sip:keepalive@91.121.129.20:5060;tag=00-07236-06cfefe5-47ad4afa4
Max-Forwards: 70
To: sip:0033972XXXXXX@sip.ovh.fr
Via: SIP/2.0/UDP 91.121.129.20:5060;rport;branch=z9hG4bK-JYTB-03efaf44-186a76d9
Content-Length: 0
<------------->
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 0 [ 51]: OPTIONS sip:0033972XXXXXX@192.99.XXX.XXX:5060 SIP/2.0
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 1 [ 50]: Call-ID: 00-07236-06cfefe6-6a375db91@91.121.129.20
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 2 [ 33]: Contact: sip:91.121.129.20:5060
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 3 [ 15]: CSeq: 1 OPTIONS
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 4 [ 72]: From: sip:keepalive@91.121.129.20:5060;tag=00-07236-06cfefe5-47ad4afa4
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 5 [ 16]: Max-Forwards: 70
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 6 [ 34]: To: sip:0033972XXXXXX@sip.ovh.fr
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 7 [ 79]: Via: SIP/2.0/UDP 91.121.129.20:5060;rport;branch=z9hG4bK-JYTB-03efaf44-186a76d9
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Header 8 [ 17]: Content-Length: 0
[Nov 6 11:27:47] VERBOSE[14933] chan_sip.c: — (9 headers 0 lines) —
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: = Looking for Call ID: 00-07236-06cfefe6-6a375db91@91.121.129.20 (Checking From) --From tag 00-07236-06cfefe5-47ad4afa4 --To-tag
[Nov 6 11:27:47] DEBUG[14933] acl.c: For destination ‘91.121.129.20’, our source address is ‘192.99.XXX.XXX’.
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Target address 91.121.129.20:5060 is not local, substituting externaddr
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.99.XXX.XXX:5060
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: Allocating new SIP dialog for 00-07236-06cfefe6-6a375db91@91.121.129.20 - OPTIONS (No RTP)
[Nov 6 11:27:47] DEBUG[14933] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[Nov 6 11:27:47] DEBUG[14933] netsock2.c: Splitting ‘192.99.XXX.XXX:5060’ into…
[Nov 6 11:27:47] DEBUG[14933] netsock2.c: …host ‘192.99.XXX.XXX’ and port ‘’.
[Nov 6 11:27:47] DEBUG[14933] netsock2.c: Splitting ‘91.121.129.20:5060’ into…
[Nov 6 11:27:47] DEBUG[14933] netsock2.c: …host ‘91.121.129.20’ and port ‘’.
[Nov 6 11:27:47] VERBOSE[14933] chan_sip.c: Looking for 0033972XXXXXX in trunkinbound (domain 192.99.XXX.XXX)
[Nov 6 11:27:47] VERBOSE[14933] chan_sip.c:
<— Transmitting (NAT) to 91.121.129.20:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-JYTB-03efaf44-186a76d9;received=91.121.129.20;rport=5060
From: sip:keepalive@91.121.129.20:5060;tag=00-07236-06cfefe5-47ad4afa4
To: sip:0033972XXXXXX@sip.ovh.fr;tag=as16be378c
Call-ID: 00-07236-06cfefe6-6a375db91@91.121.129.20
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:192.99.XXX.XXX:5060
Accept: application/sdp
Content-Length: 0