no outgoing calls possible

[miixx]

Hello,

i have a fresh installed Go Autodial 3.0 Server for tests here.

my conf Files:

[PLACETEL]
type = peer
host = fpbx.de
outboundproxy = fpbx.de
port = 5060
username = XXXXXXXXX
fromuser = XXXXXXXXX
fromdomain = fpbx.de
secret = XXXXXXXXXXX
dtmfmode = rfc2833
insecure = port,invite
canreinvite = no
registertimeout = 300
disallow = all
allow = alaw
allow = ulaw
context = trunkinbound

and

exten => _9249.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9249.,2,Dial(SIP/${EXTEN:4}@PLACETEL,,tTo)
exten => _9249.,3,Hangup

these are the logs from Asterisk:

[Feb 18 12:38:03] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-0000001b;2", "8600051,F") in new stack
[Feb 18 12:38:03] > Channel Local/8600051@default-0000001b;1 was answered.
[Feb 18 12:38:03] -- Executing [92491726550972@default:1] AGI("Local/8600051@default-0000001b;1", "agi://127.0.0.1:4577/call_log") in new stack
[Feb 18 12:38:03] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 18 12:38:03] -- <Local/8600051@default-0000001b;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Feb 18 12:38:03] -- Executing [92491726550972@default:2] SIPAddHeader("Local/8600051@default-0000001b;1", "P-Preferred-Identity:<sip:08003002500@fpbx.de>") in new stack
[Feb 18 12:38:03] -- Executing [92491726550972@default:3] SIPAddHeader("Local/8600051@default-0000001b;1", "P-Preferred-Identity:<sip:038766019010@fpbx.de>") in new stack
[Feb 18 12:38:03] -- Executing [92491726550972@default:4] Dial("Local/8600051@default-0000001b;1", "SIP/01726550972@placetel.de,90,tTor") in new stack
[Feb 18 12:38:03] == Using SIP RTP CoS mark 5
[Feb 18 12:38:03] -- Called SIP/01726550972@placetel.de
[Feb 18 12:38:06] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 18 12:38:06] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 18 12:38:12] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 18 12:38:35] -- SIP/placetel.de-0000001d is circuit-busy
[Feb 18 12:38:35] == Everyone is busy/congested at this time (1:0/1/0)
[Feb 18 12:38:35] -- Executing [92491726550972@default:5] Hangup("Local/8600051@default-0000001b;1", "") in new stack
[Feb 18 12:38:35] == Spawn extension (default, 92491726550972, 5) exited non-zero on 'Local/8600051@default-0000001b;1'
[Feb 18 12:38:35] -- Executing [h@default:1] AGI("Local/8600051@default-0000001b;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------") in new stack
[Feb 18 12:38:35] WARNING[6244]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission 034af9aa4bca15c33166d6de7ceeb42a@10.132.1.220:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 32000ms with no response
[Feb 18 12:38:35] -- <Local/8600051@default-0000001b;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Feb 18 12:38:35] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-0000001b;2'
[Feb 18 12:38:35] -- Executing [h@default:1] AGI("Local/8600051@default-0000001b;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Feb 18 12:38:35] -- <Local/8600051@default-0000001b;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


I hear the message you are the first Person etc....

But no Call goes outside...

Can you help me??

Greetz from Germany...

[geoff3dmg]

Code: Select all

[Feb 18 12:38:35] WARNING[6244]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission 034af9aa4bca15c33166d6de7ceeb42a@10.132.1.220:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions

You have a firewall/NAT issue by the looks of things.

[williamconley]

Possibilities:

1) You are on a private network (192.168.x.x or 10.x.x.x) and have not set your "externip" inside "sip.conf" to be the public IP of your router. This can cause the carrier to misroute return packets so you'll never get them.
2) Your carrier will not allow your IP address to contact them. In which case this must be solved with your carrier.
3) You have the wrong IP/domain for your carrier
4) You have NOT listed the actual "dialplan entry" that applies to this call: Your exten => entries show three lines, agi/dial/hangup ... but your asterisk log shows two sipaddheader lines which indicates that you have not provided matching information or there are entries in your system of which you are unaware.
5) Consider "qualify=500" in your "Account Entry" and then use "sip show peers" to see if you can at least make some sort of contact. NOTE: if sip show peers shows this connection as unreachable, vicidial will NOT call the carrier! No call has occurred! It never left the box! If the carrier does not support qualify, you'll need to set qualify=no. If the carrier does support qualify, then the packets are not making the entire trip so you can begin tracing packets to see why (see #1 as a possibility).
6) You have a firewall that is dropping the return packets (not normal, but does happen). This could be a firewall between you and the Net or iptables running on your Vicidial server.