Unknown peer - without anonymous on

FreePBX Providers
1

Hi All,

I’m sure this has been asked a lot, but I am trying to get incoming calls working without turning on ‘anonymous on’. I’ve searched around the forums but cant find a definitive answer.

I have an incoming SIP Trunk from my telco, all incoming calls come from the same IP. Works fine if it turn anonymous calling on. But there must me some setting I can use on the Incoming trunk that will allow the calls without turning this on?

Many Thanks.

freepbx*CLI>
  == Setting global variable 'SIPDOMAIN' to '10.50.102.11'
    -- Executing [0203XXX44XX@from-sip-external:1] NoOp("PJSIP/anonymous-00000000", "Received incoming SIP connection from unknown peer to 0203XXX44XX") in new stack
    -- Executing [0203XXX44XX@from-sip-external:2] Set("PJSIP/anonymous-00000000", "DID=0203XXX44XX") in new stack
    -- Executing [0203XXX44XX@from-sip-external:3] Goto("PJSIP/anonymous-00000000", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("PJSIP/anonymous-00000000", "1?setlanguage:checkanon") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [s@from-sip-external:2] Set("PJSIP/anonymous-00000000", "CHANNEL(language)=en") in new stack
    -- Executing [s@from-sip-external:3] GotoIf("PJSIP/anonymous-00000000", "1?noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("PJSIP/anonymous-00000000", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2020-05-01 18:46:17.199 UTC.
    -- Executing [s@from-sip-external:6] Set("PJSIP/anonymous-00000000", "receveip=pjsip,remote_addr") in new stack
    -- Executing [s@from-sip-external:7] Log("PJSIP/anonymous-00000000", "WARNING,"Rejecting unknown SIP connection from 164.39.XX.XX:5060"") in new stack
[2020-05-01 18:46:02] WARNING[13310][C-00000001]: Ext. s:7 @ from-sip-external: "Rejecting unknown SIP connection from 164.39.XX.XX:5060"
    -- Executing [s@from-sip-external:8] Answer("PJSIP/anonymous-00000000", "") in new stack
       > 0x7fb414013220 -- Strict RTP learning after remote address set to: 164.39.XX.XX:18288
    -- Executing [s@from-sip-external:9] Wait("PJSIP/anonymous-00000000", "2") in new stack
       > 0x7fb414013220 -- Strict RTP switching to RTP target address 164.39.XX.XX:18288 as source
    -- Executing [s@from-sip-external:10] Playback("PJSIP/anonymous-00000000", "ss-noservice") in new stack
    -- <PJSIP/anonymous-00000000> Playing 'ss-noservice.alaw' (language 'en')
       > 0x7fb414013220 -- Strict RTP learning complete - Locking on source address 164.39.XX.XX:18288
    -- Executing [h@from-sip-external:1] Hangup("PJSIP/anonymous-00000000", "") in new stack
  == Spawn extension (from-sip-external, h, 1) exited non-zero on 'PJSIP/anonymous-00000000'
freepbx*CLI>
2

Managed to figure it out… if I use chan_pjsip trunk instead of chan_sip, seems to work fine.

3

You need to allow for the incoming IP addresses of the service to connect to the Trunk. Are you using a PJ-SIP trunk, or Chan-SIP? If your provider can send calls from more than one server (not uncommon even with the smallest providers) you need to identify all of the server IPs that can communicate with your inbound trunk. In Chan-SIP, this means one trunk per server. If PJ-SIP, you can identify multiple blocks of addresses to allow inbound calls.

Not to call you out, but we’ve covered this at least once a week for the past year.

Unless your trunk correctly identifies the remote server, the call will be treated as anonymous. If anonymous calls are disallowed, these unidentified server sessions will be terminated.

4

This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.