OK, thanks. DNS is running on the sipXecs server.

I re-installed and used the fully qualified domain during the setup. The auto generated DNS zone file is correct and all the tests pass.

However, I still have the initial problem of not being able to transfer to an incoming call to an ext from the Auto Attendant.

Below is the contents of sipXproxy.log for the call. Some things that I noticed are

 * repeated warnings of "missing 'signature' param"
 * DNS query for name '_sip._tls.voip.datatek-net.com', type = 33
   (SRV): returned error ----->This SRV record is not in the zone file,
   is this something needed by polycom phones?
 * KERNEL:ERR ... doSendCore message send failed for queue
   'SipTcpServer-3' - no room, ret = 9

sipXproxy.log:

"2012-03-21T14:59:18.826072Z":756:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode '<sip:[email protected]>' missing 'signature' param" "2012-03-21T14:59:18.827981Z":757:SIP:WARNING:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipXauthIdentity::decode '<sip:[email protected]>' missing 'signature' param" "2012-03-21T14:59:18.829672Z":758:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send outgoing call 1" "2012-03-21T14:59:18.857719Z":759:SIP:WARNING:voip.datatek-net.com:SipSrvLookupThread-14:B6A74B90:SipXProxy:"DNS query for name '_sip._tls.voip.datatek-net.com', type = 33 (SRV): returned error" "2012-03-21T14:59:18.867593Z":760:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode '<sip:[email protected]>' missing 'signature' param" "2012-03-21T14:59:18.872363Z":761:SIP:WARNING:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send INVITE request matches existing transaction" "2012-03-21T14:59:18.873147Z":762:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send outgoing call 1" "2012-03-21T14:59:18.937284Z":763:SIP:ERR:voip.datatek-net.com:SipUserAgent-2:B6E78B90:SipXProxy:"SipUserAgent::handleMessage SIP message timeout expired with no matching transaction" "2012-03-21T14:59:18.947827Z":764:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode '\"IVR\" <sip:[email protected]>' missing 'signature' param" "2012-03-21T14:59:18.954528Z":765:SIP:WARNING:voip.datatek-net.com:SipXProxyCseObserver-10:B77FAB90:SipXProxy:"SipXauthIdentity::decode '\"IVR\" <sip:[email protected]>' missing 'signature' param" "2012-03-21T14:59:19.140082Z":766:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send outgoing call 1" "2012-03-21T14:59:30.718114Z":767:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send outgoing call 1" "2012-03-21T14:59:30.729993Z":768:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send outgoing call 1" "2012-03-21T14:59:46.579296Z":769:SIP:ERR:voip.datatek-net.com:SipRouter-11:B6D77B90:SipXProxy:"SipUserAgent::send outgoing call 1" "2012-03-21T15:01:41.158371Z":770:KERNEL:ERR:voip.datatek-net.com:SipClientTcp-377:FFFFFFFF:SipXProxy:"OsMsgQShared::doSendCore message send failed for queue 'SipTcpServer-3' - no room, ret = 9"

Any ideas?

Stiles

On 03/19/2012 04:40 PM, Douglas Hubler wrote:

Nettica.com should work.

If you only have one server, you don't need srv records, but dns should be running on sipx server and it should be configured automatically

On Mar 19, 2012 3:53 PM, "Stiles Watson" <[email protected] <mailto:[email protected]>> wrote:

    Are there instructions for setting up SRV records with Godaddy? I
    do not plan to ever ad servers, but if this is the preferred way.
    If no one like Godaddy for this, are there DNS providers which
    work well for this setup?

    I tried a couple of months ago to get SRV recs set up, but I had a
    problem with the _sip._tcp.rr record I asked about in my original
    email. Godaddy is not setup to except it.

    Stiles

    On 03/19/2012 02:38 PM, Michael Picher wrote:
    Changing the SIP domain is a bad thing to do...

    If you want to use A-Records, you should start that way.

    I'd suggest another reinstall and do it with A-Records....

    Be aware if you use A-Records you can not just add servers to the
    cluster.  This assumes SRV records.

    Mike

    On Mon, Mar 19, 2012 at 2:31 PM, Stiles Watson
    <[email protected] <mailto:[email protected]>> wrote:

        This is a follow up question to a problem I submitted about 2
        months
        ago. Long story short, I reinstalled sipxecs 4.2 and made the
        sipxecs
        server the DHCP and DNS servers as well. This solved some of
        my issues,
        but not all. I still am unable to transfer incoming calls to any
        extension. I can dial out from any ext, I can dial ext to
        ext, I can
        dial into the system and the Auto Attendant answers, but when
        I dial an
        ext I get "Please hold while I transfer your call" but the
        ext never
        rings. The CDR says the call was transfered and there is no
        error reported.

        All calls in and out of sipxecs are through a SIP Trunk via
        Flowroute.
        I'm using Polycom SoundPoint IP 335 phones. My firewall is a
        Sonicwall
        NSA, with SIP Transformations turned off and H.323 turned
        off, and
        consistent NAT enabled. RTP ports 30000-31000 UDP, SIP port
        5060 UDP &
        TCP, SIP port 5080 UDP all open and NAT rules set up to send
        traffic on
        these ports to the sipxecs server. I have to support remote
        phones.

        My external DNS records are through godaddy and the only
        record for
        sipxecs is an A record. I've followed the instructions at
        http://wiki.sipfoundry.org/display/sipXecs/DNS+Concepts+for+sipXecs
        for
        setting up A records and have changed the domain name to the
        fully
        qualified domain under System-->Domain. I clicked Apply,
        restarted the
        services and then rebooted. I also sent new profiles to the
        phones. This
        also required me to change the internal DNS zone file to use
        the fully
        qualified domain. After doing this I restarted the named service.

        DNS advisor runs successfully, but the DNS test under
        Diagnostics-->Configuration Tests returns a warning that SRV
        records for
        the unqualified domain can not be found:

        Starting DNS servers test.
        DNS testing of name server: voip.datatek-net.com
        <http://voip.datatek-net.com>
          NAPTR Lookup for datatek-net.com <http://datatek-net.com>
        domain:
            No NAPTR records found.

          SRV Lookup for Target: _sip._udp.datatek-net.com
        <http://udp.datatek-net.com>
            No SRV records found.

          SRV Lookup for Target: _sip._tcp.datatek-net.com
        <http://tcp.datatek-net.com>
            No SRV records found.

          SRV Lookup for Target: _sips._tcp.datatek-net.com
        <http://tcp.datatek-net.com>
            No SRV records found.

        Why is the test looking for SRV records for datatek-net.com
        <http://datatek-net.com> and not
        voip.datatek-net.com <http://voip.datatek-net.com> (which
        exist and which I had to add to make the DNS
        advisor happy)?

        There is one record in the internal DNS config that I do not
        understand:

        ; SRV record for service SIP TCP rr.voip.datatek-net.com
        <http://rr.voip.datatek-net.com>
        ;     priority: 1  weight: 0  port: 5070  server:
        voip.datatek-net.com <http://voip.datatek-net.com>
        ;
        _sip._tcp.rr.voip.datatek-net.com
        <http://tcp.rr.voip.datatek-net.com>. IN      SRV     1   0 5070
        voip.datatek-net.com <http://voip.datatek-net.com>.

        Why is port 5070 being used and for what?

        Thanks for your help...

        Stiles

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