Cant reach my Asterisk server from outside my network

Hi!

My name is Marcos,
I create a AsteriskNow Server with freepbx, and configure it almost like the tutorials i read.
Right now im able to connect to my server with two extensions i create on the web admin using linux and windows clients from the local network.

But cant reach it over internet… I have forwarded ports 5060, 5061, 4569, and a range from 10.000 to 11.000 as i configure in the Settings > SIP Settings (for the RTP Port Ranges).

Using the Asterisk console, i see that my attempt of connection is reaching the server…this is what i get:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f18d000faf0 – Strict RTP learning after remote address set to: 158.69.245.102:5073
– Executing [381046920479342@from-sip-external:1] NoOp(“SIP/190.52.34.174-0000000a”, “Received incoming SIP connection from unknown peer to 381046920479342”) in new stack
– Executing [381046920479342@from-sip-external:2] Set(“SIP/190.52.34.174-0000000a”, “DID=381046920479342”) in new stack
– Executing [381046920479342@from-sip-external:3] Goto(“SIP/190.52.34.174-0000000a”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/190.52.34.174-0000000a”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“SIP/190.52.34.174-0000000a”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“SIP/190.52.34.174-0000000a”, “0?noanonymous”) in new stack
– Executing [s@from-sip-external:4] Goto(“SIP/190.52.34.174-0000000a”, “from-trunk,381046920479342,1”) in new stack
– Goto (from-trunk,381046920479342,1)
– Executing [381046920479342@from-trunk:1] Set(“SIP/190.52.34.174-0000000a”, “__FROM_DID=381046920479342”) in new stack
– Executing [381046920479342@from-trunk:2] NoOp(“SIP/190.52.34.174-0000000a”, “Received an unknown call with DID set to 381046920479342”) in new stack
– Executing [381046920479342@from-trunk:3] Goto(“SIP/190.52.34.174-0000000a”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/190.52.34.174-0000000a”, “”) in new stack
– Executing [s@from-trunk:3] Log(“SIP/190.52.34.174-0000000a”, “WARNING,Friendly Scanner from 158.69.245.102”) in new stack
[2017-11-21 16:51:07] WARNING[56115][C-0000000a]: Ext. s:3 @ from-trunk: Friendly Scanner from 158.69.245.102
– Executing [s@from-trunk:4] Wait(“SIP/190.52.34.174-0000000a”, “2”) in new stack
– Executing [s@from-trunk:5] Playback(“SIP/190.52.34.174-0000000a”, “ss-noservice”) in new stack
– <SIP/190.52.34.174-0000000a> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-trunk:6] SayAlpha(“SIP/190.52.34.174-0000000a”, “381046920479342”) in new stack
– <SIP/190.52.34.174-0000000a> Playing ‘digits/3.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/8.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/0.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/4.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/6.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/9.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/2.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/0.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/4.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/7.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/9.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/3.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/4.ulaw’ (language ‘en’)
– <SIP/190.52.34.174-0000000a> Playing ‘digits/2.ulaw’ (language ‘en’)
– Executing [s@from-trunk:7] Hangup(“SIP/190.52.34.174-0000000a”, “”) in new stack
== Spawn extension (from-trunk, s, 7) exited non-zero on ‘SIP/190.52.34.174-0000000a’
– Executing [h@from-trunk:1] Macro(“SIP/190.52.34.174-0000000a”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/190.52.34.174-0000000a”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“SIP/190.52.34.174-0000000a”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] Hangup(“SIP/190.52.34.174-0000000a”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/190.52.34.174-0000000a’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/190.52.34.174-0000000a’
[2017-11-21 16:51:38] WARNING[51710]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 4b408e72e2fc0754fa43af148a1b4aac for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response

Can anyone help me to understand my problem, where i start looking for a solution o whatever bone you can throw to me xD

The FreePBX folks have a forum at https://community.freepbx.org

I do see a Retransmission warning, Usually that’s an indication of network problems, I’d make sure your external IP is set correctly and that your PBX knows it’s behind nat.

Actually i dont have a public ip, i have a DNS working:

voip.jujuy.gob.ar

wich is running against a noip over a dinamic ip… I have 8 diferentes servers running on 80 and 443, other ssh, ftp, rdp and “staff” working without problems in the same Connection.

It seems to have arrived at and been rejected by FreePBX (not supported on this forum) before the retransmission error.