vicidial.org

by **hellsing357** » Wed Oct 16, 2013 7:42 am

System Information

Code: Select all: Red Hat 4.1.2-54 goautodial-32bit-ce-3.0-final.iso Linux version 2.6.18-348.12.1.el5.go Asterisk 1.4.39.1-vici.go Single Server No Digium/Sangoma Hardware No Extra Software After Installation Fresh Install

Story :

I downloaded and installed goautodial-32bit-ce-3.0-final.iso.
I passionately followed the installation process and got it working. I heard the "You are the only person in this conference" .
When I tried to manually dial a number I cant make a call BUT when I try to call using X-Lite I manage to get a call .
My Softphone Configuration

Code: Select all: userID: 8001 Domain : 192.168.2.11 password : goautodialnow

*fresh install

Problem :

Call Rejected: BUSY
Cause: 17 - User busy.

Account Entry

Code: Select all: [123] disallow=all allow=gsm allow=ulaw type=friend dtmfmode=rfc2833 context=trunkinbound qualify=yes insecure=very nat=yes host=sip21.voiceflex. com username=84420539 secret=TotoyBrownIsMyIdol allow=g729 allow=alaw externip=121.54.54.143

Dial Plan Entry

Code: Select all: exten => _944.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _944.,2,Dial(SIP/${EXTEN:1}@123,,tTo) exten => _944.,3,Hangup exten => _3789662494.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _3789662494.,2,Dial(SIP/${EXTEN:10}@123,,tTo) exten => _3789662494.,3,Hangup

sip show peers

Code: Select all: Name/username Host Dyn Nat ACL Port Status 8020/8020 (Unspecified) D N 0 UNKNOWN 8019/8019 (Unspecified) D N 0 UNKNOWN 8018/8018 (Unspecified) D N 0 UNKNOWN 8017/8017 (Unspecified) D N 0 UNKNOWN 8016/8016 (Unspecified) D N 0 UNKNOWN 8015/8015 (Unspecified) D N 0 UNKNOWN 8014/8014 (Unspecified) D N 0 UNKNOWN 8013/8013 (Unspecified) D N 0 UNKNOWN 8012/8012 (Unspecified) D N 0 UNKNOWN 8011/8011 (Unspecified) D N 0 UNKNOWN 8010/8010 (Unspecified) D N 0 UNKNOWN 8009/8009 (Unspecified) D N 0 UNKNOWN 8008/8008 (Unspecified) D N 0 UNKNOWN 8007/8007 (Unspecified) D N 0 UNKNOWN 8006/8006 (Unspecified) D N 0 UNKNOWN 8005/8005 (Unspecified) D N 0 UNKNOWN 8004/8004 (Unspecified) D N 0 UNKNOWN 8003/8003 (Unspecified) D N 0 UNKNOWN 8002/8002 (Unspecified) D N 0 UNKNOWN 8001/8001 192.168.2.5 D N 12974 OK (6 ms) 123/84420539 93.95.124.21 N 5060 OK (753 ms) 21 sip peers [Monitored: 2 online, 19 offline Unmonitored: 0 online, 0 offline]

sip show registry

Code: Select all: Host Username Refresh State Reg.Time sip21.voiceflex. com:5060 84420539 165 Registered Wed, 16 Oct 2013 20:14:57

Logs
*After Call

Code: Select all: Asterisk 1.4.39.1-vici.go RPM by demian@goautodial. com, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer <markster@digium. com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf Connected to Asterisk 1.4.39.1-vici.go RPM by demian@goautodial. com currently running on go (pid = 2725) go*CLI> [0KVerbosity is at least 21 Core debug is at least 4 [Kgo*CLI> [0K[Oct 16 20:32:42] Really destroying SIP dialog 'MzY3ZDdiY2IzZjJiZGEyYjYwNGI3ZWJiMjkxNGMxODc' Method: REGISTER [Kgo*CLI> [0K[Oct 16 20:32:50] Reliably Transmitting (NAT) to 93.95.124.21:5060: OPTIONS sip:sip21.voiceflex. com;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK36ca6d9e;rport From: "asterisk" <sip:asterisk@192.168.2.11>;tag=as2c0e3b95 To: <sip:sip21.voiceflex. com;cpd=on> Contact: <sip:asterisk@192.168.2.11> Call-ID: 7433e87d4c6872d47ca879de5864bea9@192.168.2.11 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 16 Oct 2013 12:32:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Kgo*CLI> [0K[Oct 16 20:32:51] Retransmitting #1 (NAT) to 93.95.124.21:5060: OPTIONS sip:sip21.voiceflex. com;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK36ca6d9e;rport From: "asterisk" <sip:asterisk@192.168.2.11>;tag=as2c0e3b95 To: <sip:sip21.voiceflex. com;cpd=on> Contact: <sip:asterisk@192.168.2.11> Call-ID: 7433e87d4c6872d47ca879de5864bea9@192.168.2.11 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 16 Oct 2013 12:32:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Kgo*CLI> [0K[Oct 16 20:32:51] <--- SIP read from 93.95.124.21:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK36ca6d9e;received=58.69.77.2;rport=63382 From: "asterisk" <sip:asterisk@192.168.2.11>;tag=as2c0e3b95 To: <sip:sip21.voiceflex. com;cpd=on>;tag=as44c989b3 Call-ID: 7433e87d4c6872d47ca879de5864bea9@192.168.2.11 CSeq: 102 OPTIONS Server: Voiceflex Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [Oct 16 20:32:51] --- (11 headers 0 lines) --- [Kgo*CLI> [0K[Oct 16 20:32:51] Really destroying SIP dialog '7433e87d4c6872d47ca879de5864bea9@192.168.2.11' Method: OPTIONS [Kgo*CLI> [0K[Oct 16 20:32:53] <--- SIP read from 93.95.124.21:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK36ca6d9e;received=58.69.77.2;rport=63382 From: "asterisk" <sip:asterisk@192.168.2.11>;tag=as2c0e3b95 To: <sip:sip21.voiceflex. com;cpd=on>;tag=as44c989b3 Call-ID: 7433e87d4c6872d47ca879de5864bea9@192.168.2.11 CSeq: 102 OPTIONS Server: Voiceflex Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Accept: application/sdp Content-Length: 0 <-------------> [Oct 16 20:32:53] --- (11 headers 0 lines) --- [Kgo*CLI> [0K[Oct 16 20:32:56] == Parsing '/etc/asterisk/manager.conf': [Oct 16 20:32:56] Found [Oct 16 20:32:56] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 16 20:32:56] -- Executing [8600051@default:1] [1;36;40mMeetMe[0;37;40m("[1;35;40mLocal/8600051@default-af79,2[0;37;40m", "[1;35;40m8600051|F[0;37;40m") in new stack [Kgo*CLI> [0K[Oct 16 20:32:56] > Channel Local/8600051@default-af79,1 was answered. [Oct 16 20:32:56] -- Executing [944441244752299@default:1] [1;36;40mAGI[0;37;40m("[1;35;40mLocal/8600051@default-af79,1[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log[0;37;40m") in new stack [Oct 16 20:32:56] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Oct 16 20:32:56] -- Executing [944441244752299@default:2] [1;36;40mDial[0;37;40m("[1;35;40mLocal/8600051@default-af79,1[0;37;40m", "[1;35;40mSIP/44441244752299@123||tTo[0;37;40m") in new stack [Oct 16 20:32:56] Audio is at 192.168.2.11 port 15384 [Oct 16 20:32:56] Adding codec 0x2 (gsm) to SDP [Oct 16 20:32:56] Adding codec 0x4 (ulaw) to SDP [Oct 16 20:32:56] Adding codec 0x8 (alaw) to SDP [Oct 16 20:32:56] Adding non-codec 0x1 (telephone-event) to SDP [Oct 16 20:32:56] Reliably Transmitting (NAT) to 93.95.124.21:5060: INVITE sip:44441244752299@sip21.voiceflex. com;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK03bb3812;rport From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b To: <sip:44441244752299@sip21.voiceflex. com;cpd=on> Contact: <sip:0000000000@192.168.2.11> Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "M0162032560000000006" <sip:0000000000@192.168.2.11>;privacy=off;screen=no Date: Wed, 16 Oct 2013 12:32:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 258 v=0 o=root 2725 2725 IN IP4 192.168.2.11 s=session c=IN IP4 192.168.2.11 t=0 0 m=audio 15384 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Oct 16 20:32:56] -- Called 44441244752299@123 [Oct 16 20:32:56] == Manager 'sendcron' logged off from 127.0.0.1 [Kgo*CLI> [0K[Oct 16 20:32:56] == Parsing '/etc/asterisk/manager.conf': [Oct 16 20:32:56] Found [Oct 16 20:32:56] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 16 20:32:56] -- Executing [58600051@default:1] [1;36;40mMeetMe[0;37;40m("[1;35;40mLocal/58600051@default-323d,2[0;37;40m", "[1;35;40m8600051|Fmq[0;37;40m") in new stack [Oct 16 20:32:56] > Channel Local/58600051@default-323d,1 was answered. [Oct 16 20:32:56] -- Executing [8309@default:1] [1;36;40mAnswer[0;37;40m("[1;35;40mLocal/58600051@default-323d,1[0;37;40m", "[1;35;40m[0;37;40m") in new stack [Oct 16 20:32:56] -- Executing [8309@default:2] [1;36;40mMonitor[0;37;40m("[1;35;40mLocal/58600051@default-323d,1[0;37;40m", "[1;35;40mwav|20131016-203255_1244752299[0;37;40m") in new stack [Oct 16 20:32:56] -- Executing [8309@default:3] [1;36;40mWait[0;37;40m("[1;35;40mLocal/58600051@default-323d,1[0;37;40m", "[1;35;40m3600[0;37;40m") in new stack [Oct 16 20:32:56] == Manager 'sendcron' logged off from 127.0.0.1 [Kgo*CLI> [0K[Oct 16 20:32:57] <--- SIP read from 93.95.124.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK03bb3812;received=58.69.77.2;rport=63382 From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b To: <sip:44441244752299@sip21.voiceflex. com;cpd=on>;tag=as71b7c427 Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 102 INVITE Server: Voiceflex Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="voiceflex", nonce="40d2d2ec" Content-Length: 0 <-------------> [Oct 16 20:32:57] --- (11 headers 0 lines) --- [Oct 16 20:32:57] Transmitting (NAT) to 93.95.124.21:5060: ACK sip:44441244752299@sip21.voiceflex. com;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK03bb3812;rport From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b To: <sip:44441244752299@sip21.voiceflex. com;cpd=on>;tag=as71b7c427 Contact: <sip:0000000000@192.168.2.11> Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "M0162032560000000006" <sip:0000000000@192.168.2.11>;privacy=off;screen=no Content-Length: 0 --- [Oct 16 20:32:57] Audio is at 192.168.2.11 port 15384 [Oct 16 20:32:57] Adding codec 0x2 (gsm) to SDP [Oct 16 20:32:57] Adding codec 0x4 (ulaw) to SDP [Oct 16 20:32:57] Adding codec 0x8 (alaw) to SDP [Oct 16 20:32:57] Adding non-codec 0x1 (telephone-event) to SDP [Oct 16 20:32:57] Reliably Transmitting (NAT) to 93.95.124.21:5060: INVITE sip:44441244752299@sip21.voiceflex. com;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK2da0822d;rport From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b To: <sip:44441244752299@sip21.voiceflex. com;cpd=on> Contact: <sip:0000000000@192.168.2.11> Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "M0162032560000000006" <sip:0000000000@192.168.2.11>;privacy=off;screen=no Authorization: Digest username="84420539", realm="voiceflex", algorithm=MD5, uri="sip:44441244752299@sip21.voiceflex. com;cpd=on", nonce="40d2d2ec", response="185b7bc635719f32de850f59220925b0" Date: Wed, 16 Oct 2013 12:32:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 258 v=0 o=root 2725 2726 IN IP4 192.168.2.11 s=session c=IN IP4 192.168.2.11 t=0 0 m=audio 15384 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Kgo*CLI> [0K[Oct 16 20:32:57] <--- SIP read from 93.95.124.21:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK03bb3812;received=58.69.77.2;rport=63382 From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b To: <sip:44441244752299@sip21.voiceflex. com;cpd=on>;tag=as71b7c427 Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 102 INVITE Server: Voiceflex Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="voiceflex", nonce="40d2d2ec" Content-Length: 0 <-------------> [Oct 16 20:32:57] --- (11 headers 0 lines) --- [Kgo*CLI> [0K[Oct 16 20:32:57] <--- SIP read from 93.95.124.21:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK2da0822d;received=58.69.77.2;rport=63382 From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b To: <sip:44441244752299@sip21.voiceflex. com;cpd=on> Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 103 INVITE Server: Voiceflex Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: <sip:44441244752299@93.95.124.21:5060> Content-Length: 0 <-------------> [Oct 16 20:32:57] --- (11 headers 0 lines) --- [Kgo*CLI> [0K[Oct 16 20:32:57] <--- SIP read from 93.95.124.21:5060 ---> SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK2da0822d;received=58.69.77.2;rport=63382 From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b To: <sip:44441244752299@sip21.voiceflex. com;cpd=on>;tag=as7a2197a7 Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 103 INVITE Server: Voiceflex Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Call Rejected X-Asterisk-HangupCauseCode: 21 Content-Length: 0 <-------------> [Oct 16 20:32:57] --- (12 headers 0 lines) --- [Oct 16 20:32:57] -- Got SIP response 486 "Busy Here" back from 93.95.124.21 [Oct 16 20:32:57] Transmitting (NAT) to 93.95.124.21:5060: ACK sip:44441244752299@sip21.voiceflex. com;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK2da0822d;rport From: "M0162032560000000006" <sip:0000000000@192.168.2.11>;tag=as3672259b T [Kgo*CLI> [0Ko: <sip:44441244752299@sip21.voiceflex. com;cpd=on>;tag=as7a2197a7 Contact: <sip:0000000000@192.168.2.11> Call-ID: 4e21f8e164ec06571cb5cde90f30554c@192.168.2.11 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "M0162032560000000006" <sip:0000000000@192.168.2.11>;privacy=off;screen=no Content-Length: 0 --- [Kgo*CLI> [0K[Oct 16 20:32:57] -- SIP/123-00000006 is busy [Oct 16 20:32:57] == Everyone is busy/congested at this time (1:1/0/0) [Oct 16 20:32:57] -- Executing [944441244752299@default:3] [1;36;40mHangup[0;37;40m("[1;35;40mLocal/8600051@default-af79,1[0;37;40m", "[1;35;40m[0;37;40m") in new stack [Oct 16 20:32:57] == Spawn extension (default, 944441244752299, 3) exited non-zero on 'Local/8600051@default-af79,1' [Oct 16 20:32:57] -- Executing [h@default:1] [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/8600051@default-af79,1[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY----------[0;37;40m") in new stack [Kgo*CLI> [0K[Oct 16 20:32:57] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY---------- completed, returning 0 [Kgo*CLI> [0K[Oct 16 20:32:57] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-af79,2' [Oct 16 20:32:57] -- Executing [h@default:1] [1;36;40mDeadAGI[0;37;40m("[1;35;40mLocal/8600051@default-af79,2[0;37;40m", "[1;35;40magi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------[0;37;40m") in new stack [Kgo*CLI> [0K[Oct 16 20:32:57] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Kgo*CLI> [0K[Oct 16 20:32:57] Really destroying SIP dialog '4e21f8e164ec06571cb5cde90f30554c@192.168.2.11' Method: INVITE [Kgo*CLI> [0K[Oct 16 20:33:02] == Parsing '/etc/asterisk/manager.conf': [Oct 16 20:33:02] Found [Kgo*CLI> [0K[Oct 16 20:33:02] == Manager 'sendcron' logged on from 127.0.0.1 [Kgo*CLI> [0K[Oct 16 20:33:02] == Manager 'sendcron' logged off from 127.0.0.1 [Kgo*CLI> [0K[Oct 16 20:33:02] == Parsing '/etc/asterisk/manager.conf': [Oct 16 20:33:02] Found [Kgo*CLI> [0K[Oct 16 20:33:02] == Manager 'sendcron' logged on from 127.0.0.1 [Kgo*CLI> [0K[Oct 16 20:33:03] == Manager 'sendcron' logged off from 127.0.0.1 [Kgo*CLI> [0K[Oct 16 20:33:07] == Parsing '/etc/asterisk/manager.conf': [Oct 16 20:33:07] Found [Oct 16 20:33:07] == Manager 'sendcron' logged on from 127.0.0.1 [Oct 16 20:33:07] == Manager 'sendcron' logged off from 127.0.0.1 [Kgo*CLI> [0K[Oct 16 20:33:10] Reliably Transmitting (NAT) to 192.168.2.5:12974: OPTIONS sip:8001@192.168.2.5:12974;rinstance=37a2c7ee3ae8843f;cpd=on SIP/2.0 Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK4a624baf;rport From: "asterisk" <sip:asterisk@192.168.2.11>;tag=as3e85a3cc To: <sip:8001@192.168.2.5:12974;rinstance=37a2c7ee3ae8843f;cpd=on> Contact: <sip:asterisk@192.168.2.11> Call-ID: 5626cb00578b2986076f780f2655fe60@192.168.2.11 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 16 Oct 2013 12:33:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 --- [Kgo*CLI> [0K[Oct 16 20:33:10] <--- SIP read from 192.168.2.5:12974 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.11:5060;branch=z9hG4bK4a624baf;rport=5060 Contact: <sip:192.168.2.5:12974> To: <sip:8001@192.168.2.5:12974;rinstance=37a2c7ee3ae8843f;cpd=on>;tag=94863227 From: "asterisk"<sip:asterisk@192.168.2.11>;tag=as3e85a3cc Call-ID: 5626cb00578b2986076f780f2655fe60@192.168.2.11 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: replaces, eventlist User-Agent: X-Lite release 4.5.5 stamp 71236 Allow-Events: hold, talk Content-Length: 0 <-------------> [Oct 16 20:33:10] --- (14 headers 0 lines) --- [Kgo*CLI> [0K[Oct 16 20:33:10] Really destroying SIP dialog '5626cb00578b2986076f780f2655fe60@192.168.2.11' Method: OPTIONS [Kgo*CLI>

by **hellsing357** » Thu Oct 17, 2013 7:50 am

I manage to solve my problem

*by checking the dial code in the campaign. made sure that it is 9.

* update my dial plan entry to

Code: Select all: exten => _944.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _944.,2,Dial(SIP/${EXTEN:1}@voiceflexConf,,tTo) exten => _944.,3,Hangup

because client will only dial/call UK phone

*this lines also helps

Code: Select all: exten => _XXX.,1,AGI(agi://127.0.0.1:4577/call_log) exten => _XXX.,2,Dial(${VOIPTRUNK}/${EXTEN:1},,tToR) exten => _XXX,3,Hangup

The fact that I can make a call on my x-lite means I got it working , the only problem is the campaign and other configuration (dial plan entry)

and ohh I can also make calls using the web interface.

:oops:

vicidial.org

Call rejected Busy with goautodial-32bit-ce-3.0

Call rejected Busy with goautodial-32bit-ce-3.0

Re: Call rejected Busy with goautodial-32bit-ce-3.0

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