GOautodial Open Source Omni-channel Contact Center Suite

I have configured the carrier. It is showing registered. I can see calls being ringing but I am not getting anything from my end. I have assigned fixed IP. After agent login, I am getting confirmation call. After 20 seconds, agent is automatically logged out.

Please help me.

Host dnsmgr Username Refresh State Reg.Time
xx.xxx.xxx.xx:5060 N xxxxxxx 585 Registered Wed, 17 Dec 2014 06:34:38
1 SIP registrations.

[Dec 17 06:43:38] Scheduling destruction of SIP dialog 'YTNhYzgyMDY5ZGFlYzYxM2FjNWYwYmQ4YmRhYjQwYTI' in 6400 ms (Method: SUBSCRIBE)
[Dec 17 06:43:38] Reliably Transmitting (NAT) to 192.168.0.100:46924:
OPTIONS sip:8001@192.168.0.100:46924;rinstance=53e58f0876896619 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.102:5060;branch=z9hG4bK2c9bfa88;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.102>;tag=as3e34b582
To: <sip:8001@192.168.0.100:46924;rinstance=53e58f0876896619>
Contact: <sip:asterisk@192.168.0.102:5060>
Call-ID: 3b5f976f2b70790c271335ac53bc4f90@192.168.0.102:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Date: Wed, 17 Dec 2014 11:43:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Dec 17 06:43:22] Using SIP RTP CoS mark 5
[Dec 17 06:43:22] ERROR[2044]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("SIPtrunk", "(null)", ...): Temporary failure in name resolution
[Dec 17 06:43:22] WARNING[2044]: chan_sip.c:5865 create_addr: No such host: SIPtrunk
[Dec 17 06:43:22] Really destroying SIP dialog '7a327e1133bc20a33f9f570368caa558@127.0.0.1:5060' Method: INVITE
[Dec 17 06:43:22] WARNING[2044]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Subscriber absent)
[Dec 17 06:43:22] Everyone is busy/congested at this time (1:0/0/1)
[Dec 17 06:43:22] -- Executing [13305355952@default:3] Hangup("Local/13305355952@default-00000030;2", "") in new stack
[Dec 17 06:43:22] Spawn extension (default, 13305355952, 3) exited non-zero on 'Local/13305355952@default-00000030;2'
[Dec 17 06:43:22] -- Executing [h@default:1] AGI("Local/13305355952@default-00000030;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Dec 17 06:43:22] Manager 'sendcron' logged on from 127.0.0.1
[Dec 17 06:43:22] -- Executing [19378906710@default:1] AGI in new stack
[Dec 17 06:43:22] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=CAMP001))
[Dec 17 06:43:22] -- <Local/19378906710@default-00000031;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 17 06:43:22] -- Executing [19378906710@default:2] Dial("Local/19378906710@default-00000031;2", "sip/19378906710@SIPtrunk,55,tTo") in new stack
[Dec 17 06:43:22] Using SIP RTP CoS mark 5
[Dec 17 06:43:22] ERROR[2052]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("SIPtrunk", "(null)", ...): Temporary failure in name resolution
[Dec 17 06:43:22] WARNING[2052]: chan_sip.c:5865 create_addr: No such host: SIPtrunk
[Dec 17 06:43:22] Really destroying SIP dialog '299683f415ba06543ea417f7538761a6@127.0.0.1:5060' Method: INVITE
[Dec 17 06:43:22] WARNING[2052]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Subscriber absent)
[Dec 17 06:43:22] Everyone is busy/congested at this time (1:0/0/1)
[Dec 17 06:43:22] -- Executing [19378906710@default:3] Hangup("Local/19378906710@default-00000031;2", "") in new stack
[Dec 17 06:43:22] Spawn extension (default, 19378906710, 3) exited non-zero on 'Local/19378906710@default-00000031;2'
[Dec 17 06:43:22] -- Executing [h@default:1] AGI("Local/19378906710@default-00000031;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Dec 17 06:43:22] Manager 'sendcron' logged on from 127.0.0.1
[Dec 17 06:43:22] -- Executing [15139412772@default:1] AGI in new stack

Dial Plan ENTRY
exten => _1XXXXXXXXXX,1,AGI
exten => _1XXXXXXXXXX,2,Dial(sip/${EXTEN}@SIPtrunk,55,tTo)
exten => _1XXXXXXXXXX,3,Hangup

SIP Details
[voip]
disallow=all
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes
host=xxxxxxx
username=xxxxxxxxxx
secret=xxxxxxxx
allow=alaw