Vicidial not able to dial out

General and Support topics relating to ViciDialNow and GoAutoDial ISO installers

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Vicidial not able to dial out

Postby vjonvicidial » Sat May 02, 2015 8:40 am

We have been using the Digium's Switchvox with Vicidial from quite sometime and everything was working fine. Today we had to restart the Switchvox server post which vicidial has stopped working.
The problem seemed to be with SIP, here is the Debug

Code: Select all
SIP Debugging re-enabled
[May  2 19:00:14] Really destroying SIP dialog '31014d9b3be160bc64a3ae06294084fc@192.168.1.10' Method: OPTIONS
go*CLI> SIP DEBUG
SIP Debugging re-enabled
[May  2 19:00:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK59c6bc22;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as04e70eab
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 43c182d134b778733f0af6ce53059809@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:30:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:00:42] --- (13 headers 0 lines) ---
[May  2 19:00:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:00:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK59c6bc22;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as04e70eab
To: <sip:192.168.1.105>;tag=as176c2d4a
Call-ID: 43c182d134b778733f0af6ce53059809@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:00:42] Scheduling destruction of SIP dialog '43c182d134b778733f0af6ce53059809@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:00:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK2acdb522;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as30a4e609
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 1c7c7a7513997dd37ec8d3b2578388ce@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:30:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:00:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK2acdb522;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as30a4e609
To: <sip:192.168.1.10;cpd=on>;tag=as140910a6
Call-ID: 1c7c7a7513997dd37ec8d3b2578388ce@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
[May  2 19:00:45] --- (11 headers 0 lines) ---
[May  2 19:00:45] Really destroying SIP dialog '1c7c7a7513997dd37ec8d3b2578388ce@192.168.1.105' Method: OPTIONS
[May  2 19:01:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:01:01] Found
[May  2 19:01:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:01:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:01:01] Found
[May  2 19:01:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:01:02]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:01:02]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:01:07]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:01:07] Found
[May  2 19:01:07]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:01:07]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:01:14] Really destroying SIP dialog '43c182d134b778733f0af6ce53059809@192.168.1.10' Method: OPTIONS
[May  2 19:01:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2d4269f6;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as677446b5
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 7523e97b310d184f6fc550e80e34aaec@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:31:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:01:42] --- (13 headers 0 lines) ---
[May  2 19:01:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:01:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2d4269f6;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as677446b5
To: <sip:192.168.1.105>;tag=as3dafafa6
Call-ID: 7523e97b310d184f6fc550e80e34aaec@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:01:42] Scheduling destruction of SIP dialog '7523e97b310d184f6fc550e80e34aaec@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:01:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK4270d3fe;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as6aec23ef
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 0d5c770a14083cc7557405c043cc971b@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:31:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:01:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK4270d3fe;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as6aec23ef
To: <sip:192.168.1.10;cpd=on>;tag=as30e11aa0
Call-ID: 0d5c770a14083cc7557405c043cc971b@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
[May  2 19:01:45] --- (11 headers 0 lines) ---
[May  2 19:01:45] Really destroying SIP dialog '0d5c770a14083cc7557405c043cc971b@192.168.1.105' Method: OPTIONS
[May  2 19:01:45]   == Refreshing DNS lookups.
[May  2 19:02:02]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:02:02] Found
[May  2 19:02:02]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:02:02]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:02:02] Found
[May  2 19:02:02]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:02:02]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:02:02]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:02:07]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:02:07] Found
[May  2 19:02:07]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:02:07]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:02:14] Really destroying SIP dialog '7523e97b310d184f6fc550e80e34aaec@192.168.1.10' Method: OPTIONS
[May  2 19:02:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1ab87730;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as027d4874
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 63f8346e7425fdb83852302d2a02a855@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:32:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:02:42] --- (13 headers 0 lines) ---
[May  2 19:02:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:02:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1ab87730;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as027d4874
To: <sip:192.168.1.105>;tag=as3b816bef
Call-ID: 63f8346e7425fdb83852302d2a02a855@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:02:42] Scheduling destruction of SIP dialog '63f8346e7425fdb83852302d2a02a855@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:02:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK386b8983;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as6438789c
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 5a5cd07721a79fe82569d8bf41e65335@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:32:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:02:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK386b8983;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as6438789c
To: <sip:192.168.1.10;cpd=on>;tag=as6bdc2aa8
Call-ID: 5a5cd07721a79fe82569d8bf41e65335@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
[May  2 19:02:45] --- (11 headers 0 lines) ---
[May  2 19:02:45] Really destroying SIP dialog '5a5cd07721a79fe82569d8bf41e65335@192.168.1.105' Method: OPTIONS
[May  2 19:03:02]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:03:02] Found
[May  2 19:03:02]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:03:02]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:03:02] Found
[May  2 19:03:02]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:03:02]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:03:02]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:03:07]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:03:07] Found
[May  2 19:03:07]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:03:07]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:03:14] Really destroying SIP dialog '63f8346e7425fdb83852302d2a02a855@192.168.1.10' Method: OPTIONS
[May  2 19:03:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK79a83247;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as6204ed31
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 434377782eb99c942a1d9b8329b42337@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:33:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:03:42] --- (13 headers 0 lines) ---
[May  2 19:03:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:03:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK79a83247;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as6204ed31
To: <sip:192.168.1.105>;tag=as2e679174
Call-ID: 434377782eb99c942a1d9b8329b42337@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:03:42] Scheduling destruction of SIP dialog '434377782eb99c942a1d9b8329b42337@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:03:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK0540334f;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as61626931
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 2eb221a93c0e98b35016d57a258adb3a@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:33:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:03:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK0540334f;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as61626931
To: <sip:192.168.1.10;cpd=on>;tag=as259847d6
Call-ID: 2eb221a93c0e98b35016d57a258adb3a@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
[May  2 19:03:45] --- (11 headers 0 lines) ---
[May  2 19:03:45] Really destroying SIP dialog '2eb221a93c0e98b35016d57a258adb3a@192.168.1.105' Method: OPTIONS
[May  2 19:04:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:04:01] Found
[May  2 19:04:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:04:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:04:01] Found
[May  2 19:04:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:04:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:04:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:04:06]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:04:06] Found
[May  2 19:04:06]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:04:06]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:04:14] Really destroying SIP dialog '434377782eb99c942a1d9b8329b42337@192.168.1.10' Method: OPTIONS
[May  2 19:04:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3a1c13e4;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as77b424f4
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 6e45033c178516f81a012aa51c8ccb2b@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:34:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:04:42] --- (13 headers 0 lines) ---
[May  2 19:04:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:04:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK3a1c13e4;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as77b424f4
To: <sip:192.168.1.105>;tag=as3bb5d048
Call-ID: 6e45033c178516f81a012aa51c8ccb2b@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:04:42] Scheduling destruction of SIP dialog '6e45033c178516f81a012aa51c8ccb2b@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:04:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK67022557;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as76a18faa
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 138709600f2baf476c8a171a7a8d64f6@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:34:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:04:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK67022557;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as76a18faa
To: <sip:192.168.1.10;cpd=on>;tag=as692010ea
Call-ID: 138709600f2baf476c8a171a7a8d64f6@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
[May  2 19:04:45] --- (11 headers 0 lines) ---
[May  2 19:04:45] Really destroying SIP dialog '138709600f2baf476c8a171a7a8d64f6@192.168.1.105' Method: OPTIONS
[May  2 19:05:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:05:01] Found
[May  2 19:05:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:05:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:05:01] Found
[May  2 19:05:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:05:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:05:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:05:06]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:05:06] Found
[May  2 19:05:06]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:05:06]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:05:14] Really destroying SIP dialog '6e45033c178516f81a012aa51c8ccb2b@192.168.1.10' Method: OPTIONS
[May  2 19:05:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK353501a9;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as3fc72921
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 06853a9262f6444b48dece697ce7a7d1@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:35:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:05:42] --- (13 headers 0 lines) ---
[May  2 19:05:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:05:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK353501a9;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as3fc72921
To: <sip:192.168.1.105>;tag=as23107830
Call-ID: 06853a9262f6444b48dece697ce7a7d1@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:05:42] Scheduling destruction of SIP dialog '06853a9262f6444b48dece697ce7a7d1@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:05:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK35504ed9;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as5bf187fc
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 303f748032882fbb60efc12f4d070249@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:35:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:05:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK35504ed9;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as5bf187fc
To: <sip:192.168.1.10;cpd=on>;tag=as12a3a7d5
Call-ID: 303f748032882fbb60efc12f4d070249@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
[May  2 19:05:45] --- (11 headers 0 lines) ---
[May  2 19:05:45] Really destroying SIP dialog '303f748032882fbb60efc12f4d070249@192.168.1.105' Method: OPTIONS
[May  2 19:06:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:06:01] Found
[May  2 19:06:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:06:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:06:01] Found
[May  2 19:06:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:06:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:06:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:06:06]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:06:06] Found
[May  2 19:06:06]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:06:06]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:06:14] Really destroying SIP dialog '06853a9262f6444b48dece697ce7a7d1@192.168.1.10' Method: OPTIONS
[May  2 19:06:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2a6ece0c;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as6713b019
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 7945596450144ce31ccdaf2c76509525@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:36:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:06:42] --- (13 headers 0 lines) ---
[May  2 19:06:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:06:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK2a6ece0c;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as6713b019
To: <sip:192.168.1.105>;tag=as064e1ca7
Call-ID: 7945596450144ce31ccdaf2c76509525@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:06:42] Scheduling destruction of SIP dialog '7945596450144ce31ccdaf2c76509525@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:06:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK2ce492df;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as392b21cf
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 716748ab1332674c4ee8379403824c4d@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:36:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:06:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK2ce492df;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as392b21cf
To: <sip:192.168.1.10;cpd=on>;tag=as220fae2e
Call-ID: 716748ab1332674c4ee8379403824c4d@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------->
[May  2 19:06:45] --- (11 headers 0 lines) ---
[May  2 19:06:45] Really destroying SIP dialog '716748ab1332674c4ee8379403824c4d@192.168.1.105' Method: OPTIONS
[May  2 19:06:45]   == Refreshing DNS lookups.
[May  2 19:07:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:07:01] Found
[May  2 19:07:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:07:01]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:07:01] Found
[May  2 19:07:01]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:07:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:07:01]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:07:06]   == Parsing '/etc/asterisk/manager.conf': [May  2 19:07:06] Found
[May  2 19:07:06]   == Manager 'sendcron' logged on from 127.0.0.1
[May  2 19:07:06]   == Manager 'sendcron' logged off from 127.0.0.1
[May  2 19:07:14] Really destroying SIP dialog '7945596450144ce31ccdaf2c76509525@192.168.1.10' Method: OPTIONS
[May  2 19:07:42]
<--- SIP read from 192.168.1.10:5060 --->
OPTIONS sip:192.168.1.105 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1d136a6a;rport
From: "unknown" <sip:unknown@192.168.1.10>;tag=as32f82ddc
To: <sip:192.168.1.105>
Contact: <sip:unknown@192.168.1.10:5060>
Call-ID: 1be5cff13910a7735634a3394c16cdfd@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Max-Forwards: 70
Date: Sat, 02 May 2015 13:37:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------->
[May  2 19:07:42] --- (13 headers 0 lines) ---
[May  2 19:07:42] Looking for s in trunkinbound (domain 192.168.1.105)
[May  2 19:07:42]
<--- Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1d136a6a;received=192.168.1.10;rport=5060
From: "unknown" <sip:unknown@192.168.1.10>;tag=as32f82ddc
To: <sip:192.168.1.105>;tag=as4085e028
Call-ID: 1be5cff13910a7735634a3394c16cdfd@192.168.1.10
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


<------------>
[May  2 19:07:42] Scheduling destruction of SIP dialog '1be5cff13910a7735634a3394c16cdfd@192.168.1.10' in 32000 ms (Method: OPTIONS)
[May  2 19:07:45] Reliably Transmitting (NAT) to 192.168.1.10:5060:
OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK29e08cd1;rport
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as668b5a77
To: <sip:192.168.1.10;cpd=on>
Contact: <sip:asterisk@192.168.1.105>
Call-ID: 6b90fc5128c1bd6540cef78d74949946@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 02 May 2015 13:37:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[May  2 19:07:45]
<--- SIP read from 192.168.1.10:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK29e08cd1;received=192.168.1.105;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as668b5a77
To: <sip:192.168.1.10;cpd=on>;tag=as12348bfb
Call-ID: 6b90fc5128c1bd6540cef78d74949946@192.168.1.105
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX (digium)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0


Otherwise the Switchvox is running fine with both outbounds and inbounds done manually.
Goautodial 2.0 Vicidial VERSION: 2.4-308a BUILD: 110428-1108
Switchvox SMB 5.5
vjonvicidial
 
Posts: 12
Joined: Sat May 12, 2012 4:56 am
Location: India

Re: Vicidial not able to dial out

Postby vjonvicidial » Sat May 02, 2015 8:50 am

Also here is the part of debug code from Switchvox
Code: Select all
1430572425]<------------->
[1430572425]
[1430572425]<--- SIP read from 192.168.1.105:5060 --->
[1430572425]OPTIONS sip:192.168.1.10;cpd=on SIP/2.0
[1430572425]Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK157d0fde;rport
[1430572425]From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as14581e1b
[1430572425]To: <sip:192.168.1.10;cpd=on>
[1430572425]Contact: <sip:asterisk@192.168.1.105>
[1430572425]Call-ID: 1d539685007e5629354f8f0e03e7391f@192.168.1.105
[1430572425]CSeq: 102 OPTIONS
[1430572425]User-Agent: Asterisk PBX
[1430572425]Max-Forwards: 70
[1430572425]Date: Sat, 02 May 2015 13:13:45 GMT
[1430572425]Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[1430572425]Supported: replaces
[1430572425]Content-Length: 0
[1430572425]
[1430572425]
[1430572425]<------------->
[1430572425]--- (13 headers 0 lines) ---
[1430572425]Looking for s in unknown_provider (domain 192.168.1.10)
[1430572425]
[1430572425]<--- Transmitting (no NAT) to 192.168.1.105:5060 --->
[1430572425]SIP/2.0 404 Not Found
[1430572425]Via: SIP/2.0/UDP 192.168.1.105:5060;branch=z9hG4bK157d0fde;received=192.168.1.105;rport=5060
[1430572425]From: "asterisk" <sip:asterisk@192.168.1.105>;tag=as14581e1b
[1430572425]To: <sip:192.168.1.10;cpd=on>;tag=as4021bb02
[1430572425]Call-ID: 1d539685007e5629354f8f0e03e7391f@192.168.1.105
[1430572425]CSeq: 102 OPTIONS
[1430572425]User-Agent: Asterisk PBX (digium)
[1430572425]Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
[1430572425]Supported: replaces
[1430572425]Accept: application/sdp
[1430572425]Content-Length: 0
[1430572425]
[1430572425]
[1430572425]<------------>
[1430572425]Scheduling destruction of SIP dialog '1d539685007e5629354f8f0e03e7391f@192.168.1.105' in 32000 ms (Method: OPTIONS)
[1430572425]
[1430572425]<--- SIP read from 192.168.1.26:8520 --->
[1430572425]
[1430572425]
[1430572425]<------------->
[1430572426]
[1430572426]<--- SIP read from 192.168.1.21:6198 --->
[1430572426]
[1430572426]
[1430572426]<------------->
[1430572428]
[1430572428]<--- SIP read from 192.168.1.20:8466 --->
[
Goautodial 2.0 Vicidial VERSION: 2.4-308a BUILD: 110428-1108
Switchvox SMB 5.5
vjonvicidial
 
Posts: 12
Joined: Sat May 12, 2012 4:56 am
Location: India


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