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  1. FreePBX
  2. FREEPBX-18310

Another RECEIVED INCOMING SIP CONNECTION FROM UNKNOWN PEER Message

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    Details

    • Type: Bug
    • Status: Closed
    • Priority: Minor
    • Resolution: Not an issue
    • Affects Version/s: None
    • Fix Version/s: None
    • Component/s: None
    • Labels:
      None
    • ToDo:
    • Asterisk Version:
      Asterisk 13.19.1 built by mockbuild @ jenkins7 on a x86_64 running Linux
    • Distro Version:
      12.7.4-1804-2.sng7
    • Distro:
      FreePBX Distro

      Description

      Hello,

      I am having issues with a “received incoming sip connection from unknown peer” message when routing a call into my Asterisk server. I just downloaded and installed a FreePBX (latest distro) onto a virtual machine.

      Have researched and can’t seem to figure out the issue – I am a bit new to Asterisk configuration.

      Below is a working configuration:

      GATE ASTERISK – connects to my upstream SIP Providers and connects to internal ASTERISK servers.

      Outgoing Settings

      Trunk Name: MY_TRUNK

      username=GATE2_LINK

      type=friend

      secret=<secret>

      qualify=yes

      insecure=very

      host=voip4.<domain.tld> … this is the domain pointing to the internal Asterisk Server

      dtmfmode=auto

      context=from-internal

      trustrpid=yes

      Incoming Settings

                      Blank

       

      VOIP Server – internal and sits “behind” the Gate Asterisk Server

      Outgoing Settings

      Trunk Name: GATE2_LINK

      username=MY_TRUNK

      type=friend

      secret=<secret>

      qualify=yes

      insecure=very

      host=gate2.<domain.tld> … this is the domain pointing to the gate server (upstream)

      dtmfmode=auto

      context=from-trunk

      Incoming Settings

                      Blank

       

      The Firewall is disabled and both servers sit behind a firewall. Calls flow in and out of VOIP4 as expected. All extensions are CHAN_SIP and connected to VOIP4 via CISCO SPA504g phone.

       

      I setup in a different location a new instance using the latest FreePBX distro and installed using default settings (Asterisk 13). This time I used the included Firewall and setup using the default settings.

      Created CHAN_SIP and PJSIP Extensions and connected to a CISCO SPA504g phone.

       

      PBX Server – connected to Gate2 Asterisk server

      Outgoing Settings

      Trunk Name: GATE2_LINK

      username=MY_TRUNK

      type=friend

      secret=<secret>

      qualify=yes

      insecure=very

      host=gate2.thecomputerzone.ca

      dtmfmode=auto

      context=from-trunk

      Incoming Settings

                      Blank

      On the Extension connected to PBX Server I can make an outbound call. However, when I route a called to the same extension get the message “received incoming sip connection from unknown peer to …”

      I can set “Allow Anonymous Inbound SIP Calls” on the PBX Server to “YES” and the calls come in but obviously this is not a permanent solution – not one I would deploy.

      Any help or insight would be appreciated.

       

       

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              • Assignee:
                Unassigned
                Reporter:
                TheRussian1968 DT
              • Votes:
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                Dates

                • Created:
                  Updated:
                  Resolved:

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