vicidial.org

VICIDIAL astGUIclient discussion forum

Skip to content

Vicidial.org Home      Vicidial Forum      Vicidial Wiki      Vicidial Issue Tracker      astGUIclient Project Page


autodial not working some calls are not transfared to agent

Support forum for the ViciBox ISO Server Install and ISO LiveCD Demo

Moderators: enjay, williamconley, Staydog, mflorell, MJCoate, mcargile, Kumba

Post a reply

autodial not working some calls are not transfared to agent

Postby Michael_N » Thu Apr 22, 2010 5:54 pm

System vicibox redux

Agent web-client version: 2.4-268 BUILD: 100420-1009
VERSION: 2.4-250
BUILD: 100420-1010

The call is not transfered to agent.
But sometimes it is.

SIP DEBUG on 2 calls

[Apr 23 01:48:01] VERBOSE[20436] logger.c: [Apr 23 01:48:01] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:48:01] VERBOSE[20436] logger.c: [Apr 23 01:48:01] Found
[Apr 23 01:48:01] VERBOSE[20436] logger.c: [Apr 23 01:48:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:48:01] VERBOSE[20442] logger.c: [Apr 23 01:48:01] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:48:01] VERBOSE[20442] logger.c: [Apr 23 01:48:01] Found
[Apr 23 01:48:01] VERBOSE[20442] logger.c: [Apr 23 01:48:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:48:01] ERROR[20442] utils.c: write() returned error: Connection reset by peer
[Apr 23 01:48:01] ERROR[20442] utils.c: write() returned error: Broken pipe
[Apr 23 01:48:01] VERBOSE[20442] logger.c: [Apr 23 01:48:01] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:48:04] VERBOSE[20451] logger.c: [Apr 23 01:48:04] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:48:04] VERBOSE[20451] logger.c: [Apr 23 01:48:04] Found
[Apr 23 01:48:04] VERBOSE[20451] logger.c: [Apr 23 01:48:04] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] -- Executing [0418332828@default:1] AGI("Local/0418332828@default-6dfc,2", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] -- Executing [0418332828@default:2] Dial("Local/0418332828@default-6dfc,2", "SIP/cellip/0418332828|55|tTo") in new stack
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] Audio is at 10.27.98.3 port 16138
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] Adding codec 0x8 (alaw) to SDP
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] Adding codec 0x4 (ulaw) to SDP
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] Reliably Transmitting (NAT) to 62.80.200.53:5060:
INVITE sip:0418332828@sip.mysecretary.net;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK78d0e691;rport
From: "V0423014804000022533" <sip:46855118000@10.27.98.3>;tag=as3c8c579a
To: <sip:0418332828@sip.mysecretary.net;cpd=on>
Contact: <sip:46855118000@10.27.98.3>
Call-ID: 3b2b62a0627e203c488ec39924361cef@10.27.98.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V0423014804000022533" <sip:0000000000@10.27.98.3>;privacy=off;screen=no
Date: Thu, 22 Apr 2010 23:48:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 4252 4252 IN IP4 10.27.98.3
s=session
c=IN IP4 10.27.98.3
t=0 0
m=audio 16138 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Apr 23 01:48:04] VERBOSE[20452] logger.c: [Apr 23 01:48:04] -- Called cellip/0418332828
[Apr 23 01:48:04] VERBOSE[4261] logger.c: [Apr 23 01:48:04]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK78d0e691;rport=5060;received=94.254.112.33
From: "V0423014804000022533" <sip:46855118000@10.27.98.3>;tag=as3c8c579a
To: <sip:0418332828@sip.mysecretary.net;cpd=on>
Call-ID: 3b2b62a0627e203c488ec39924361cef@10.27.98.3
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


<------------->
[Apr 23 01:48:04] VERBOSE[4261] logger.c: [Apr 23 01:48:04] --- (8 headers 0 lines) ---
[Apr 23 01:48:05] VERBOSE[4261] logger.c: [Apr 23 01:48:05]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK78d0e691;rport=5060
Record-Route: <sip:62.80.200.53;ftag=as3c8c579a;lr>
From: V0423014804000022533 <sip:46855118000@10.27.98.3>;tag=as3c8c579a
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=b5f11cf086fe5b0b69d9925f888e988fi
Call-ID: 3b2b62a0627e203c488ec39924361cef@10.27.98.3
CSeq: 102 INVITE
Server: Sippy


<------------->
[Apr 23 01:48:05] VERBOSE[4261] logger.c: [Apr 23 01:48:05] --- (8 headers 0 lines) ---
[Apr 23 01:48:05] VERBOSE[20452] logger.c: [Apr 23 01:48:05] -- SIP/cellip-0000008d is ringing
[Apr 23 01:48:06] VERBOSE[20456] logger.c: [Apr 23 01:48:06] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:48:06] VERBOSE[20456] logger.c: [Apr 23 01:48:06] Found
[Apr 23 01:48:06] VERBOSE[20456] logger.c: [Apr 23 01:48:06] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:48:07] ERROR[20456] utils.c: write() returned error: Connection reset by peer
[Apr 23 01:48:07] ERROR[20456] utils.c: write() returned error: Broken pipe
[Apr 23 01:48:07] VERBOSE[20456] logger.c: [Apr 23 01:48:07] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:48:07] VERBOSE[20436] logger.c: [Apr 23 01:48:07] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:48:07] VERBOSE[4261] logger.c: [Apr 23 01:48:07] Reliably Transmitting (no NAT) to 62.80.200.53:5060:
OPTIONS sip:sip.mysecretary.net;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK622408d5;rport
From: "asterisk" <sip:asterisk@10.27.98.3>;tag=as11df3fb7
To: <sip:sip.mysecretary.net;cpd=on>
Contact: <sip:asterisk@10.27.98.3>
Call-ID: 2f1e04216b5c7143507fff3c668310ea@10.27.98.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Apr 2010 23:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Apr 23 01:48:07] VERBOSE[4261] logger.c: [Apr 23 01:48:07]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK622408d5;rport=5060
From: asterisk <sip:asterisk@10.27.98.3>;tag=as11df3fb7
To: <sip:sip.mysecretary.net;cpd=on>
Call-ID: 2f1e04216b5c7143507fff3c668310ea@10.27.98.3
CSeq: 102 OPTIONS
Server: Sippy


<------------->
[Apr 23 01:48:07] VERBOSE[4261] logger.c: [Apr 23 01:48:07] --- (7 headers 0 lines) ---
[Apr 23 01:48:07] VERBOSE[4261] logger.c: [Apr 23 01:48:07] Reliably Transmitting (NAT) to 62.80.200.53:5060:
OPTIONS sip:sip.mysecretary.net;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK53172506;rport
From: "asterisk" <sip:asterisk@10.27.98.3>;tag=as30f6e02b
To: <sip:sip.mysecretary.net;cpd=on>
Contact: <sip:asterisk@10.27.98.3>
Call-ID: 2d7b088419d26f33425637cf701c3102@10.27.98.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Apr 2010 23:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Apr 23 01:48:07] VERBOSE[4261] logger.c: [Apr 23 01:48:07]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK53172506;rport=5060
From: asterisk <sip:asterisk@10.27.98.3>;tag=as30f6e02b
To: <sip:sip.mysecretary.net;cpd=on>
Call-ID: 2d7b088419d26f33425637cf701c3102@10.27.98.3
CSeq: 102 OPTIONS
Server: Sippy


<------------->
[Apr 23 01:48:07] VERBOSE[4261] logger.c: [Apr 23 01:48:07] --- (7 headers 0 lines) ---
[Apr 23 01:48:08] VERBOSE[4261] logger.c: [Apr 23 01:48:08] Really destroying SIP dialog '2d7b088419d26f33425637cf701c3102@10.27.98.3' Method: OPTIONS
[Apr 23 01:48:08] VERBOSE[4261] logger.c: [Apr 23 01:48:08] Really destroying SIP dialog '2f1e04216b5c7143507fff3c668310ea@10.27.98.3' Method: OPTIONS
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK78d0e691;rport=5060
Record-Route: <sip:62.80.200.53;ftag=as3c8c579a;lr>
From: V0423014804000022533 <sip:46855118000@10.27.98.3>;tag=as3c8c579a
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=b5f11cf086fe5b0b69d9925f888e988fi
Call-ID: 3b2b62a0627e203c488ec39924361cef@10.27.98.3
CSeq: 102 INVITE
Server: Sippy
Contact: Anonymous <sip:62.80.200.53:5061>
H323-credit-time: 10800
Content-Length: 265
Content-Type: application/sdp

v=0
o=Sippy 181776812 1 IN IP4 62.80.200.53
s=session
t=0 0
m=audio 36832 RTP/AVP 8 0 101
c=IN IP4 62.80.200.53
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] --- (12 headers 13 lines) ---
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Found RTP audio format 8
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Found RTP audio format 0
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Found RTP audio format 101
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Found audio description format PCMA for ID 8
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Found audio description format PCMU for ID 0
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Found audio description format telephone-event for ID 101
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Peer audio RTP is at port 62.80.200.53:36832
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] list_route: hop: <sip:62.80.200.53;ftag=as3c8c579a;lr>
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] set_destination: Parsing <sip:62.80.200.53;ftag=as3c8c579a;lr> for address/port to send to
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] set_destination: set destination to 62.80.200.53, port 5060
[Apr 23 01:48:11] VERBOSE[4261] logger.c: [Apr 23 01:48:11] Transmitting (NAT) to 62.80.200.53:5060:
ACK sip:62.80.200.53:5061 SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK31c3cc79;rport
Route: <sip:62.80.200.53;ftag=as3c8c579a;lr>
From: "V0423014804000022533" <sip:46855118000@10.27.98.3>;tag=as3c8c579a
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=b5f11cf086fe5b0b69d9925f888e988fi
Contact: <sip:46855118000@10.27.98.3>
Call-ID: 3b2b62a0627e203c488ec39924361cef@10.27.98.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V0423014804000022533" <sip:0000000000@10.27.98.3>;privacy=off;screen=no
Content-Length: 0


---
[Apr 23 01:48:11] VERBOSE[20452] logger.c: [Apr 23 01:48:11] -- SIP/cellip-0000008d answered Local/0418332828@default-6dfc,2
[Apr 23 01:48:11] VERBOSE[20451] logger.c: [Apr 23 01:48:11] > Channel Local/0418332828@default-6dfc,1 was answered.
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- Executing [8368@default:1] Playback("Local/0418332828@default-6dfc,1", "sip-silence") in new stack
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- <Local/0418332828@default-6dfc,1> Playing 'sip-silence' (language 'en')
[Apr 23 01:48:11] WARNING[20467] file.c: Unexpected control subclass '-1'
[Apr 23 01:48:11] VERBOSE[20451] logger.c: [Apr 23 01:48:11] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:48:11] VERBOSE[20452] logger.c: [Apr 23 01:48:11] -- Executing [h@default:1] DeadAGI("Local/0418332828@default-6dfc,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0") in new stack
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- Executing [8368@default:2] AGI("SIP/cellip-0000008d", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 23 01:48:11] VERBOSE[20452] logger.c: [Apr 23 01:48:11] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---7-----0 completed, returning 0
[Apr 23 01:48:11] VERBOSE[20452] logger.c: [Apr 23 01:48:11] == Spawn extension (default, 0418332828, 2) exited non-zero on 'Local/0418332828@default-6dfc,2'
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- Executing [8368@default:3] AGI("SIP/cellip-0000008d", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- Executing [010*027*098*003*8600051@default:1] Goto("SIP/cellip-0000008d", "default|8600051|1") in new stack
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- Goto (default,8600051,1)
[Apr 23 01:48:11] VERBOSE[20467] logger.c: [Apr 23 01:48:11] -- Executing [8600051@default:1] MeetMe("SIP/cellip-0000008d", "8600051|F") in new stack
[Apr 23 01:48:11] DEBUG[20467] app_meetme.c: Placed channel SIP/cellip-0000008d in DAHDI conf 1023
[Apr 23 01:48:17] VERBOSE[20486] logger.c: [Apr 23 01:48:17] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:48:17] VERBOSE[20486] logger.c: [Apr 23 01:48:17] Found
[Apr 23 01:48:17] VERBOSE[20486] logger.c: [Apr 23 01:48:17] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:48:17] VERBOSE[20467] logger.c: [Apr 23 01:48:17] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cellip-0000008d'
[Apr 23 01:48:17] VERBOSE[20467] logger.c: [Apr 23 01:48:17] -- Executing [h@default:1] DeadAGI("SIP/cellip-0000008d", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 23 01:48:17] VERBOSE[20467] logger.c: [Apr 23 01:48:17] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 23 01:48:17] VERBOSE[20467] logger.c: [Apr 23 01:48:17] Scheduling destruction of SIP dialog '3b2b62a0627e203c488ec39924361cef@10.27.98.3' in 6400 ms (Method: INVITE)
[Apr 23 01:48:17] VERBOSE[20467] logger.c: [Apr 23 01:48:17] set_destination: Parsing <sip:62.80.200.53;ftag=as3c8c579a;lr> for address/port to send to
[Apr 23 01:48:17] VERBOSE[20467] logger.c: [Apr 23 01:48:17] set_destination: set destination to 62.80.200.53, port 5060
[Apr 23 01:48:17] VERBOSE[20467] logger.c: [Apr 23 01:48:17] Reliably Transmitting (NAT) to 62.80.200.53:5060:
BYE sip:62.80.200.53:5061 SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK21751f80;rport
Route: <sip:62.80.200.53;ftag=as3c8c579a;lr>
From: "V0423014804000022533" <sip:46855118000@10.27.98.3>;tag=as3c8c579a
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=b5f11cf086fe5b0b69d9925f888e988fi
Call-ID: 3b2b62a0627e203c488ec39924361cef@10.27.98.3
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V0423014804000022533" <sip:0000000000@10.27.98.3>;privacy=off;screen=no
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
[Apr 23 01:48:17] VERBOSE[20487] logger.c: [Apr 23 01:48:17] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:48:17] VERBOSE[20487] logger.c: [Apr 23 01:48:17] Found
[Apr 23 01:48:17] VERBOSE[20487] logger.c: [Apr 23 01:48:17] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:48:17] VERBOSE[4261] logger.c: [Apr 23 01:48:17]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK21751f80;rport=5060
From: V0423014804000022533 <sip:46855118000@10.27.98.3>;tag=as3c8c579a
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=b5f11cf086fe5b0b69d9925f888e988fi
Call-ID: 3b2b62a0627e203c488ec39924361cef@10.27.98.3
CSeq: 103 BYE
Server: Sippy


<------------->
[Apr 23 01:48:17] VERBOSE[4261] logger.c: [Apr 23 01:48:17] --- (7 headers 0 lines) ---
[Apr 23 01:48:17] VERBOSE[4261] logger.c: [Apr 23 01:48:17] Really destroying SIP dialog '3b2b62a0627e203c488ec39924361cef@10.27.98.3' Method: INVITE
[Apr 23 01:48:21] VERBOSE[20486] logger.c: [Apr 23 01:48:21] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:48:21] VERBOSE[20487] logger.c: [Apr 23 01:48:21] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:48:57] VERBOSE[20564] logger.c: [Apr 23 01:48:57] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:48:57] VERBOSE[20564] logger.c: [Apr 23 01:48:57] Found
[Apr 23 01:48:57] VERBOSE[20564] logger.c: [Apr 23 01:48:57] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] -- Executing [0418332828@default:1] AGI("Local/0418332828@default-918b,2", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] -- Executing [0418332828@default:2] Dial("Local/0418332828@default-918b,2", "SIP/cellip/0418332828|55|tTo") in new stack
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] Audio is at 10.27.98.3 port 16118
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] Adding codec 0x8 (alaw) to SDP
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] Adding codec 0x4 (ulaw) to SDP
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] Adding non-codec 0x1 (telephone-event) to SDP
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] Reliably Transmitting (NAT) to 62.80.200.53:5060:
INVITE sip:0418332828@sip.mysecretary.net;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK71d853a3;rport
From: "V0423014856000022534" <sip:46855118000@10.27.98.3>;tag=as5a22d06c
To: <sip:0418332828@sip.mysecretary.net;cpd=on>
Contact: <sip:46855118000@10.27.98.3>
Call-ID: 4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V0423014856000022534" <sip:0000000000@10.27.98.3>;privacy=off;screen=no
Date: Thu, 22 Apr 2010 23:48:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 258

v=0
o=root 4252 4252 IN IP4 10.27.98.3
s=session
c=IN IP4 10.27.98.3
t=0 0
m=audio 16118 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Apr 23 01:48:57] VERBOSE[20565] logger.c: [Apr 23 01:48:57] -- Called cellip/0418332828
[Apr 23 01:48:57] VERBOSE[4261] logger.c: [Apr 23 01:48:57]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK71d853a3;rport=5060;received=94.254.112.33
From: "V0423014856000022534" <sip:46855118000@10.27.98.3>;tag=as5a22d06c
To: <sip:0418332828@sip.mysecretary.net;cpd=on>
Call-ID: 4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


<------------->
[Apr 23 01:48:57] VERBOSE[4261] logger.c: [Apr 23 01:48:57] --- (8 headers 0 lines) ---
[Apr 23 01:48:58] VERBOSE[4261] logger.c: [Apr 23 01:48:58]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK71d853a3;rport=5060
Record-Route: <sip:62.80.200.53;ftag=as5a22d06c;lr>
From: V0423014856000022534 <sip:46855118000@10.27.98.3>;tag=as5a22d06c
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=83959f7a99c9dd7ff089808d52870c5fi
Call-ID: 4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3
CSeq: 102 INVITE
Server: Sippy


<------------->
[Apr 23 01:48:58] VERBOSE[4261] logger.c: [Apr 23 01:48:58] --- (8 headers 0 lines) ---
[Apr 23 01:48:58] VERBOSE[20565] logger.c: [Apr 23 01:48:58] -- SIP/cellip-0000008e is ringing
[Apr 23 01:49:01] VERBOSE[20591] logger.c: [Apr 23 01:49:01] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:49:01] VERBOSE[20591] logger.c: [Apr 23 01:49:01] Found
[Apr 23 01:49:01] VERBOSE[20591] logger.c: [Apr 23 01:49:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:49:01] VERBOSE[20592] logger.c: [Apr 23 01:49:01] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:49:01] VERBOSE[20592] logger.c: [Apr 23 01:49:01] Found
[Apr 23 01:49:01] VERBOSE[20592] logger.c: [Apr 23 01:49:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:49:01] ERROR[20591] utils.c: write() returned error: Connection reset by peer
[Apr 23 01:49:01] ERROR[20591] utils.c: write() returned error: Broken pipe
[Apr 23 01:49:01] VERBOSE[20591] logger.c: [Apr 23 01:49:01] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK71d853a3;rport=5060
Record-Route: <sip:62.80.200.53;ftag=as5a22d06c;lr>
From: V0423014856000022534 <sip:46855118000@10.27.98.3>;tag=as5a22d06c
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=83959f7a99c9dd7ff089808d52870c5fi
Call-ID: 4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3
CSeq: 102 INVITE
Server: Sippy
Contact: Anonymous <sip:62.80.200.53:5061>
H323-credit-time: 10800
Content-Length: 265
Content-Type: application/sdp

v=0
o=Sippy 145836236 1 IN IP4 62.80.200.53
s=session
t=0 0
m=audio 36836 RTP/AVP 8 0 101
c=IN IP4 62.80.200.53
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] --- (12 headers 13 lines) ---
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Found RTP audio format 8
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Found RTP audio format 0
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Found RTP audio format 101
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Found audio description format PCMA for ID 8
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Found audio description format PCMU for ID 0
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Found audio description format telephone-event for ID 101
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Peer audio RTP is at port 62.80.200.53:36836
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] list_route: hop: <sip:62.80.200.53;ftag=as5a22d06c;lr>
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] set_destination: Parsing <sip:62.80.200.53;ftag=as5a22d06c;lr> for address/port to send to
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] set_destination: set destination to 62.80.200.53, port 5060
[Apr 23 01:49:02] VERBOSE[4261] logger.c: [Apr 23 01:49:02] Transmitting (NAT) to 62.80.200.53:5060:
ACK sip:62.80.200.53:5061 SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK709df820;rport
Route: <sip:62.80.200.53;ftag=as5a22d06c;lr>
From: "V0423014856000022534" <sip:46855118000@10.27.98.3>;tag=as5a22d06c
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=83959f7a99c9dd7ff089808d52870c5fi
Contact: <sip:46855118000@10.27.98.3>
Call-ID: 4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V0423014856000022534" <sip:0000000000@10.27.98.3>;privacy=off;screen=no
Content-Length: 0


---
[Apr 23 01:49:02] VERBOSE[20565] logger.c: [Apr 23 01:49:02] -- SIP/cellip-0000008e answered Local/0418332828@default-918b,2
[Apr 23 01:49:02] VERBOSE[20564] logger.c: [Apr 23 01:49:02] > Channel Local/0418332828@default-918b,1 was answered.
[Apr 23 01:49:02] VERBOSE[20594] logger.c: [Apr 23 01:49:02] -- Executing [8368@default:1] Playback("Local/0418332828@default-918b,1", "sip-silence") in new stack
[Apr 23 01:49:02] VERBOSE[20564] logger.c: [Apr 23 01:49:02] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:49:02] VERBOSE[20594] logger.c: [Apr 23 01:49:02] -- <Local/0418332828@default-918b,1> Playing 'sip-silence' (language 'en')
[Apr 23 01:49:02] WARNING[20594] file.c: Unexpected control subclass '-1'
[Apr 23 01:49:02] VERBOSE[20594] logger.c: [Apr 23 01:49:02] -- Executing [8368@default:2] AGI("Local/0418332828@default-918b,1", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 23 01:49:02] VERBOSE[20594] logger.c: [Apr 23 01:49:02] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 23 01:49:02] VERBOSE[20594] logger.c: [Apr 23 01:49:02] -- Executing [8368@default:3] AGI("Local/0418332828@default-918b,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Apr 23 01:49:02] VERBOSE[20594] logger.c: [Apr 23 01:49:02] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Apr 23 01:49:03] VERBOSE[20594] logger.c: [Apr 23 01:49:03] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Apr 23 01:49:03] VERBOSE[20594] logger.c: [Apr 23 01:49:03] -- Executing [8368@default:4] AGI("Local/0418332828@default-918b,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Apr 23 01:49:03] VERBOSE[20594] logger.c: [Apr 23 01:49:03] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Apr 23 01:49:04] VERBOSE[20594] logger.c: [Apr 23 01:49:04] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Apr 23 01:49:04] VERBOSE[20594] logger.c: [Apr 23 01:49:04] -- Executing [8368@default:5] Hangup("Local/0418332828@default-918b,1", "") in new stack
[Apr 23 01:49:04] VERBOSE[20594] logger.c: [Apr 23 01:49:04] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/0418332828@default-918b,1'
[Apr 23 01:49:04] VERBOSE[20594] logger.c: [Apr 23 01:49:04] -- Executing [h@default:1] DeadAGI("Local/0418332828@default-918b,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Apr 23 01:49:04] VERBOSE[20594] logger.c: [Apr 23 01:49:04] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 23 01:49:04] VERBOSE[20565] logger.c: [Apr 23 01:49:04] -- Executing [h@default:1] DeadAGI("Local/0418332828@default-918b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----2") in new stack
[Apr 23 01:49:04] VERBOSE[20565] logger.c: [Apr 23 01:49:04] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---7-----2 completed, returning 0
[Apr 23 01:49:04] VERBOSE[20565] logger.c: [Apr 23 01:49:04] Scheduling destruction of SIP dialog '4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3' in 6400 ms (Method: INVITE)
[Apr 23 01:49:04] VERBOSE[20565] logger.c: [Apr 23 01:49:04] set_destination: Parsing <sip:62.80.200.53;ftag=as5a22d06c;lr> for address/port to send to
[Apr 23 01:49:04] VERBOSE[20565] logger.c: [Apr 23 01:49:04] set_destination: set destination to 62.80.200.53, port 5060
[Apr 23 01:49:04] VERBOSE[20565] logger.c: [Apr 23 01:49:04] Reliably Transmitting (NAT) to 62.80.200.53:5060:
BYE sip:62.80.200.53:5061 SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK40cf5b1f;rport
Route: <sip:62.80.200.53;ftag=as5a22d06c;lr>
From: "V0423014856000022534" <sip:46855118000@10.27.98.3>;tag=as5a22d06c
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=83959f7a99c9dd7ff089808d52870c5fi
Call-ID: 4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "V0423014856000022534" <sip:0000000000@10.27.98.3>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Apr 23 01:49:04] VERBOSE[20565] logger.c: [Apr 23 01:49:04] == Spawn extension (default, 0418332828, 2) exited non-zero on 'Local/0418332828@default-918b,2'
[Apr 23 01:49:04] VERBOSE[4261] logger.c: [Apr 23 01:49:04]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK40cf5b1f;rport=5060
From: V0423014856000022534 <sip:46855118000@10.27.98.3>;tag=as5a22d06c
To: <sip:0418332828@sip.mysecretary.net;cpd=on>;tag=83959f7a99c9dd7ff089808d52870c5fi
Call-ID: 4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3
CSeq: 103 BYE
Server: Sippy


<------------->
[Apr 23 01:49:04] VERBOSE[4261] logger.c: [Apr 23 01:49:04] --- (7 headers 0 lines) ---
[Apr 23 01:49:04] VERBOSE[4261] logger.c: [Apr 23 01:49:04] Really destroying SIP dialog '4d82c7a91af62e8b2cd56e424109bb28@10.27.98.3' Method: INVITE
[Apr 23 01:49:06] VERBOSE[20610] logger.c: [Apr 23 01:49:06] == Parsing '/etc/asterisk/manager.conf': [Apr 23 01:49:07] VERBOSE[20610] logger.c: [Apr 23 01:49:07] Found
[Apr 23 01:49:07] VERBOSE[20610] logger.c: [Apr 23 01:49:07] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 23 01:49:07] VERBOSE[20592] logger.c: [Apr 23 01:49:07] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:49:07] ERROR[20610] utils.c: write() returned error: Connection reset by peer
[Apr 23 01:49:07] ERROR[20610] utils.c: write() returned error: Broken pipe
[Apr 23 01:49:07] VERBOSE[20610] logger.c: [Apr 23 01:49:07] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 23 01:49:07] VERBOSE[4261] logger.c: [Apr 23 01:49:07] Reliably Transmitting (no NAT) to 62.80.200.53:5060:
OPTIONS sip:sip.mysecretary.net;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK603ad482;rport
From: "asterisk" <sip:asterisk@10.27.98.3>;tag=as7420bf82
To: <sip:sip.mysecretary.net;cpd=on>
Contact: <sip:asterisk@10.27.98.3>
Call-ID: 4ef83799114e6f69016aa06b4b8ffe06@10.27.98.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Apr 2010 23:49:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Apr 23 01:49:07] VERBOSE[4261] logger.c: [Apr 23 01:49:07]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK603ad482;rport=5060
From: asterisk <sip:asterisk@10.27.98.3>;tag=as7420bf82
To: <sip:sip.mysecretary.net;cpd=on>
Call-ID: 4ef83799114e6f69016aa06b4b8ffe06@10.27.98.3
CSeq: 102 OPTIONS
Server: Sippy


<------------->
[Apr 23 01:49:07] VERBOSE[4261] logger.c: [Apr 23 01:49:07] --- (7 headers 0 lines) ---
[Apr 23 01:49:07] VERBOSE[4261] logger.c: [Apr 23 01:49:07] Reliably Transmitting (NAT) to 62.80.200.53:5060:
OPTIONS sip:sip.mysecretary.net;cpd=on SIP/2.0
Via: SIP/2.0/UDP 10.27.98.3:5060;branch=z9hG4bK409cece8;rport
From: "asterisk" <sip:asterisk@10.27.98.3>;tag=as51683df1
To: <sip:sip.mysecretary.net;cpd=on>
Contact: <sip:asterisk@10.27.98.3>
Call-ID: 56a567d755e76e5c607db0ef319e94f1@10.27.98.3
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Apr 2010 23:49:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Apr 23 01:49:07] VERBOSE[4261] logger.c: [Apr 23 01:49:07]
<--- SIP read from 62.80.200.53:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.27.98.3:5060;received=94.254.112.33;branch=z9hG4bK409cece8;rport=5060
From: asterisk <sip:asterisk@10.27.98.3>;tag=as51683df1
To: <sip:sip.mysecretary.net;cpd=on>
Call-ID: 56a567d755e76e5c607db0ef319e94f1@10.27.98.3
CSeq: 102 OPTIONS
Server: Sippy


<------------->
[Apr 23 01:49:07] VERBOSE[4261] logger.c: [Apr 23 01:49:07] --- (7 headers 0 lines) ---
[Apr 23 01:49:08] VERBOSE[4261] logger.c: [Apr 23 01:49:08] Really destroying SIP dialog '56a567d755e76e5c607db0ef319e94f1@10.27.98.3' Method: OPTIONS
[Apr 23 01:49:08] VERBOSE[4261] logger.c: [Apr 23 01:49:08] Really destroying SIP dialog '4ef83799114e6f69016aa06b4b8ffe06@10.27.98.3' Method: OPTIONS
Michael_N
 
Posts: 687
Joined: Wed Jul 05, 2006 3:13 pm
Location: sweden
Top

Postby mflorell » Fri Apr 23, 2010 6:18 am

That output doesn't help at all, show the Real Asterisk CLI(no debug please) and the agiout logfile output for the same call.
mflorell
Site Admin
 
Posts: 18387
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida
Top
Post a reply

Return to ViciBox Server Install and Demo

Who is online

Users browsing this forum: No registered users and 42 guests

Powered by phpBB® Forum Software © phpBB Group