[asterisk-bugs] [JIRA] (ASTERISK-22579) peer is not matched to an IP address

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22579?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-22579:
------------------------------------

    Assignee: Private Name
      Status: Waiting for Feedback  (was: Triage)

There is not enough information here to verify the issue that you describe.


https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

* Please follow the instructions above and provide a full DEBUG and VERBOSE log, with verbose and debug turned up to 5 (see asterisk.conf to set this where they will affect log output from logger.conf)

* Be sure to use "sip set debug on" so that we have SIP packets in the debug log.

* We need to see the log with the complete call path from beginning to end.

* Please attach your sanitized sip.conf.
                
> peer is not matched to an IP address
> ------------------------------------
>
>                 Key: ASTERISK-22579
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22579
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 11.5.1
>         Environment: debian 64
>            Reporter: Private Name
>            Assignee: Private Name
>
> My peer is behind a NAT. 
> The peer definition is
> {noformat}
> [XXXX.YYY.ZZ.PPP]
> type=peer
> host=XXXX.YYY.ZZ.PPP
> insecure=port,invite
> directrtpsetup=no
> directmedia=no
> nat=comedia,force_rport
> {noformat}
> In spite if that, when I place a call, the bridging always shows "remotely bridging". This is wrong, so I started to investigate and found that when I have the call connected, and type
> {noformat}
>  Using SIP RTP CoS mark 5
>     -- Called SIP/19544447408 at 67.xx.237.xx
>     -- SIP/67.xx.237.xx-00009fc6 is making progress passing it to SIP/8.xx.245.xx-00009fc5
>     -- SIP/67.xx.237.xx-00009fc6 answered SIP/8.xx.245.xx-00009fc5
>     -- Remotely bridging SIP/8.xx.245.xx-00009fc5 and SIP/67.xx.237.xx-00009fc6
> {noformat}
> {noformat}
> sys254*CLI> sip show channels
> Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry     Peer
> 67.xx.237.xx    19544447408      507f0a921c46ce6  (ulaw)           No       Tx: ACK                    <guest>
> 8.xx.245.xx      Asterisk         NGY0NzE5ZjBlNDI  (ulaw)           No       Rx: ACK                    <guest>
> 2 active SIP dialogs
> {noformat}
> The call drops after a few seconds because my peer never gets a packet.
> If you look closely, "8.xx.245.xx      Asterisk         NGY0NzE5ZjBlNDI " this is the issue. The IP is that of my Asterisk server, not of my peer. It is not matching my peer to the IP address, thus, the NAT and media configuration are not being used.
> If I set the server as a whole, in the [general] section to my nat and media restrictions, then it works fine. 
> This means that something is broken with Asterisk 11, something that worked fine before.
> I can give a developer access 24x7 to my server. He/she may send a call from behind a NAT. I will create a peer with the IP address, and the issue will be evident.

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