cause 20 - unable to create channel type

[john_usc]

I am using go-auto dial and I am getting this error in sip. What is going on. these setting were working last night when we closed up the shop. Can you help. My production is down. thanks

[Aug 9 22:18:19] -- Executing [917201234567@default:1] AGI("SIP/104-00000016", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 9 22:18:19] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 9 22:18:19] -- Executing [917201234567@default:2] Dial("SIP/104-00000016", "SIP/sipTrucks/7203334762||tTor") in new stack
[Aug 9 22:18:19] Really destroying SIP dialog '74c5eeb7212aa4b16e34039e0d2435de@127.0.0.1' Method: INVITE
[Aug 9 22:18:19] WARNING[23549]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Aug 9 22:18:19] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 9 22:18:19] -- Executing [917201234567@default:3] Hangup("SIP/104-00000016", "") in new stack
[Aug 9 22:18:19] == Spawn extension (default, 917201234567, 3) exited non-zero on 'SIP/104-00000016'
[Aug 9 22:18:19] -- Executing [h@default:1] DeadAGI("SIP/104-00000016", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Aug 9 22:18:19] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 9 22:18:19] Scheduling destruction of SIP dialog 'ZTA3OWMwYmRlNDZiZjNmYzg5N2EwYWRiYzI2OTk4MDE.' in 32000 ms (Method: INVITE)
[Aug 9 22:18:19]
<--- Reliably Transmitting (NAT) to 192.168.1.11:5060 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK-d8754z-8dc4a34a52120bb9-1---d8754z-;received=192.168.1.11;rport=5060
From: "104"<sip:104@192.168.1.2>;tag=2c57aa9c
To: <sip:917201234567@192.168.1.2>;tag=as61ae6226
Call-ID: ZTA3OWMwYmRlNDZiZjNmYzg5N2EwYWRiYzI2OTk4MDE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Aug 9 22:18:19]
<--- SIP read from 192.168.1.11:5060 --->
ACK sip:917201234567@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK-d8754z-8dc4a34a52120bb9-1---d8754z-;rport
Max-Forwards: 70
To: <sip:917201234567@192.168.1.2>;tag=as61ae6226
From: "104"<sip:104@192.168.1.2>;tag=2c57aa9c
Call-ID: ZTA3OWMwYmRlNDZiZjNmYzg5N2EwYWRiYzI2OTk4MDE.
CSeq: 2 ACK
Content-Length: 0


<------------->
[Aug 9 22:18:19] --- (8 headers 0 lines) ---
[Aug 9 22:18:22] Reliably Transmitting (NAT) to 75.71.46.229:5060:
OPTIONS sip:106@192.168.10.46:5060;rinstance=7a8e79c97127415c;cpd=on SIP/2.0
Via: SIP/2.0/UDP 67.50.195.190:5060;branch=z9hG4bK3c84f7bd;rport
From: "asterisk" <sip:asterisk@67.50.195.190>;tag=as77ef9ab7
To: <sip:106@192.168.10.46:5060;rinstance=7a8e79c97127415c;cpd=on>
Contact: <sip:asterisk@67.50.195.190>
Call-ID: 70e474f62c782c3911b11ae86640bc55@67.50.195.190
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 10 Aug 2012 02:18:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Aug 9 22:18:22]
<--- SIP read from 75.71.46.229:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.50.195.190:5060;branch=z9hG4bK3c84f7bd;rport=5060;received=50.132.55.38
Contact: <sip:192.168.10.46:5060>
To: <sip:106@192.168.10.46:5060;rinstance=7a8e79c97127415c;cpd=on>;tag=1990b9c0
From: "asterisk"<sip:asterisk@67.50.195.190>;tag=as77ef9ab7
Call-ID: 70e474f62c782c3911b11ae86640bc55@67.50.195.190
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 0


<------------->
[Aug 9 22:18:22] --- (13 headers 0 lines) ---
[Aug 9 22:18:22] Really destroying SIP dialog '70e474f62c782c3911b11ae86640bc55@67.50.195.190' Method: OPTIONS
[Aug 9 22:18:22] Reliably Transmitting (NAT) to 192.168.1.11:5060:
OPTIONS sip:104@192.168.1.11:5060;rinstance=c490cc4cd4226c70;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2b026f7a;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as0f79b46c
To: <sip:104@192.168.1.11:5060;rinstance=c490cc4cd4226c70;cpd=on>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 0a46798c736379a32c9c88702a2e6045@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 10 Aug 2012 02:18:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Aug 9 22:18:22]
<--- SIP read from 192.168.1.11:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2b026f7a;rport=5060
Contact: <sip:192.168.1.11:5060>
To: <sip:104@192.168.1.11:5060;rinstance=c490cc4cd4226c70;cpd=on>;tag=98f363e7
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as0f79b46c
Call-ID: 0a46798c736379a32c9c88702a2e6045@192.168.1.2
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 0


<------------->
[Aug 9 22:18:22] --- (13 headers 0 lines) ---
[Aug 9 22:18:22] Really destroying SIP dialog '0a46798c736379a32c9c88702a2e6045@192.168.1.2' Method: OPTIONS
[Aug 9 22:18:24]
<--- SIP read from 173.164.46.6:5108 --->




VERSION: 2.4-309a
BUILD: 110430-1642

[williamconley]

show your carrier "account entry" section. specifically:

SIP/sipTrucks/7203334762

there must be a "[sipTrucks]" and the carrier must be of type "SIP".