Incoming calls are failing when Anonymous Inbound SIP calls set to no

[faisalkhan]

HI all,

I am into a strange issue. I have freepbx 14 distro running and firewall enabled.

I have allowed various trusted ips in the firewall even carrier media ip’s.

I have incoming with Flow Route and what happens when I disallow anonymous calls the Pbx rejects the calls and considering their whitelisted ip’s in the firewall as unknown ip. but when I allow anonymous calls incoming works fine.

This is totally strange.

please check the rejected call logs:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [18015987341@from-sip-external:1] NoOp(“SIP/fl.gg-000002b1”, “Received incoming SIP connection from unknown peer to 18015987341”) in new stack
– Executing [18015987341@from-sip-external:2] Set(“SIP/fl.gg-000002b1”, “DID=18015987341”) in new stack
– Executing [18015987341@from-sip-external:3] Goto(“SIP/fl.gg-000002b1”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/fl.gg-000002b1”, “1?setlanguage:checkanon”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] Set(“SIP/fl.gg-000002b1”, “CHANNEL(language)=en”) in new stack
– Executing [s@from-sip-external:3] GotoIf(“SIP/fl.gg-000002b1”, “1?noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“SIP/fl.gg-000002b1”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2018-08-28 11:45:16.501 MDT.
– Executing [s@from-sip-external:6] Log(“SIP/fl.gg-000002b1”, "WARNING,“Rejecting unknown SIP connection from 34.210.91.114"”) in new stack
[2018-08-28 11:45:01] WARNING[26160][C-0000069b]: Ext. s:6 @ from-sip-external: “Rejecting unknown SIP connection from 34.210.91.114”
– Executing [s@from-sip-external:7] Answer(“SIP/fl.gg-000002b1”, “”) in new stack
– Executing [s@from-sip-external:8] Wait(“SIP/fl.gg-000002b1”, “2”) in new stack
– Executing [s@from-sip-external:9] Playback(“SIP/fl.gg-000002b1”, “ss-noservice”) in new stack
– <SIP/fl.gg-000002b1> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-sip-external:10] PlayTones(“SIP/fl.gg-000002b1”, “congestion”) in new stack
– Executing [s@from-sip-external:11] Congestion(“SIP/fl.gg-000002b1”, “5”) in new stack
[2018-08-28 11:45:11] WARNING[2488]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission 956deb93167e5f92c60aad8eb671cf74 for seqno 1 (Critical Response) – See Home - Asterisk Documentation
Packet timed out after 31999ms with no response
== Spawn extension (from-sip-external, s, 11) exited non-zero on ‘SIP/fl.gg-000002b1’
– Executing [h@from-sip-external:1] Hangup(“SIP/fl.gg-000002b1”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/fl.gg-000002b1’

[matthewljensen (Matthew Jensen)]

I had a similar problem a while back. I was never able to solve it and I’ve just been too busy to go back and figure it out. It happened when I tried to use Flowroute’s PoP servers. I contacted Flowroute and they didn’t know what the problem was. I didn’t have time to go back and forth on the problem with them, so I just switched back to not using a preferred PoP.

[faisalkhan]

so what do you suggest me to do?

[dicko]

https://support.flowroute.com/SIP_Trunking_and_Voice/Getting_Started/Interconnection_with_the_New_PoPs

Both allow these ips in your firewall and construct trunks for them

https://support.flowroute.com/SIP_Trunking_and_Voice/PBX__Configuration_Guides/Asterisk/Configure_Asterisk_13

[faisalkhan]

so we have to create new trunks for every single of their IP.

Is there any way that we can allow multiple ips in one single trunk

[dicko]

Probably if you use pjsip and add them to your “Match (permit)” list. Never tried it though, you only need chan_sip trunks for your main and backup servers.