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Feature #1765

closed

Calls ringing but not connecting.

Added by sarath matcha over 9 years ago. Updated almost 8 years ago.

Status:
Closed
Priority:
Urgent
Assignee:
sarath matcha
Category:
Features
Target version:
-
Start date:
12/17/2014
Due date:
% Done:

100%

Estimated time:

Description

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::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:>;tag=as3e34b582
To: <sip::46924;rinstance=53e58f0876896619>
Contact: <sip::5060>
Call-ID: :5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5.go RPM by
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 ':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 ':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

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 #1

Updated by kuldeep patel over 9 years ago

sip.conf

[trunk]
type=friend
username=XXXXXXXXXX
secret=XXXXXXXX
host=XXXXXXXXX
dfmfmode=rfc2833
disallow=all
allow=all
allow=g729
allow=alaw
allow=ulaw

exten => _9X.,1,AGI
exten => _9X.,2,NoOp(CALLERID ::::: ${CALLERID})
exten => _9X.,3,Dial(SIP/trunk/${EXTEN:1},,To)
exten => _9X.,4,Hangup

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 #2

Updated by Levy Ryan Nolasco almost 8 years ago

  • Status changed from New to Closed
  • Assignee set to sarath matcha
  • % Done changed from 0 to 100

Hi,

Your carrier context is voip not SIPTrunk. Try to change your dial plan context to voip or either. This issue/bug tracker is meant for bugs, feature request and issues related to the GOautodial CE ISO, system installation and the GOautodial applications (GOadmin, GOreports and GOagent). To get help from the community, please post your concerns in our forum board at http://goautodial.org/projects/goautodialce/boards

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