Vicibox 6.0.4 from .iso | Vicidial VERSION: 2.12-516a BUILD: 151022-1404| Asterisk 1.8.32.3| Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel(R) Xeon(R) CPU E31220 @ 3.10GHz Quad Core Ram 8gb HDD 250gb
carrier settings
register =>@x.x.x.x
[contabo]
disallow=all
allow=g729
port=5060
insecure=invite
type=friend
host=x.x.x.x
dtmfmode=rfc2833
trustrpid=yes
sendrpid=yes
canreinvite=no
qualify=no
context=custom-a2billing
global string: SIPTRUNK= SIP/contabo
exten => _39XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _39XXX.,2,Dial(${SIPTRUNK}/${EXTEN},60,tTo)
exten => _39XXX.,3,Hangup
server1*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
server1*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
101/101 192.168.5.121 D N 36914 OK (100 ms)
contabo x.x.x.x N 5060 Unmonitored
gs102/gs102 (Unspecified) D N 0 UNKNOWN
3 sip peers [Monitored: 1 online, 1 offline Unmonitored: 1 online, 0 offline]
what happen when i try to make a call
- server1:/usr/src # asterisk -r
Asterisk 1.8.32.3-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer
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.
=========================================================================
Connected to Asterisk 1.8.32.3-vici currently running on server1 (pid = 5917)
Verbosity is at least 21
[Nov 2 16:02:13] == Using SIP RTP CoS mark 5
[Nov 2 16:02:13] -- Executing [39043746647@default:1] AGI("SIP/101-00000018", "agi://127.0.0.1:4577/call_log") in new stack
[Nov 2 16:02:14] -- <SIP/101-00000018>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Nov 2 16:02:14] -- Executing [39043746647@default:2] Dial("SIP/101-00000018", "SIP/contabo/39043746647,60,tTo") in new stack
[Nov 2 16:02:14] == Using SIP RTP CoS mark 5
[Nov 2 16:02:14] -- Called SIP/contabo/39043746647
[Nov 2 16:02:46] WARNING[13752]: chan_sip.c:3822 retrans_pkt: Retransmission timeout reached on transmission 5a38b1813258519c722ef93a5294c856@X.Y.Z.W:5060 for seqno 102 (Critical Request) -- See wiki.asterisk /wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Nov 2 16:02:46] WARNING[13752]: chan_sip.c:3851 retrans_pkt: Hanging up call 5a38b1813258519c722ef93a5294c856@X.Y.Z.W:5060:5060 - no reply to our critical packet (see wiki.asterisk /wiki/display/AST/SIP+Retransmissions).
[Nov 2 16:02:46] -- SIP/contabo-00000019 is circuit-busy
[Nov 2 16:02:46] == Everyone is busy/congested at this time (1:0/1/0)
[Nov 2 16:02:46] -- Executing [39043746647@default:3] Hangup("SIP/101-00000018", "") in new stack
[Nov 2 16:02:46] == Spawn extension (default, 39043746647, 3) exited non-zero on 'SIP/101-00000018'
[Nov 2 16:02:46] -- Executing [h@default:1] AGI("SIP/101-00000018", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----CONGESTION----------") in new stack
[Nov 2 16:02:46] -- <SIP/101-00000018>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0